00:00.09 | *** part/#asterisk atil (i=hugo@212-41-80-186.adsl.solnet.ch) |
00:01.04 | [hC] | Anyone have access to a CCM server? This 7970 is requesting a distinctiveringlist.xml file that is -supposed- to be the same format as ringlist.xml (as described on cisco's site) yet it doesnt seem to work, a copy of that file would be handy |
00:01.11 | kippi | when I do *CLI> iax2 show |
00:01.12 | kippi | No such command 'iax2 show' (type 'help' for help) |
00:01.24 | robin_sz | kippi: |
00:01.35 | robin_sz | the reason is that there is no such command iax2 show |
00:01.41 | benjk | you need to tell it *what* to show |
00:01.48 | benjk | iax2 show peers |
00:01.51 | benjk | for example |
00:02.03 | robin_sz | try help iax2 to see what you can type |
00:02.23 | kippi | ah ok |
00:02.47 | kippi | any ideas why i am getting this error? |
00:02.48 | kippi | Feb 22 00:01:50 NOTICE[9082]: chan_iax2.c:3918 register_verify: No registration for peer '1001' (from 10.6.10.149) |
00:03.14 | robin_sz | could be any number of reasons |
00:03.19 | *** join/#asterisk sergeus|w (n=sergeus@ippe-245.ippe.ru) |
00:03.43 | robin_sz | paster yer iax2.conf some place (less the passwords) .. maybe someone can see |
00:05.58 | kippi | http://pastebin.ca/42712 |
00:07.33 | clyrrad | is it possible to define custom variables for a SIP user in sip.conf, that can be referenced from extensions.conf? |
00:09.28 | clyrrad | I am not looking to make global variable, but just a variable that will exist in the dial plan wherever this sip account is used |
00:12.22 | *** join/#asterisk p0g0__ (n=pogo@madwifi/support/p0g0) |
00:12.26 | *** join/#asterisk Dorphalsig (n=Me@200.71.58.39) |
00:12.28 | Dorphalsig | Hello |
00:13.08 | Dorphalsig | I have an outbound campaing I want to run, its basically to dial a number in a db, wait for somebody to answer, play a recording and hangup |
00:13.25 | kippi | robin_sz: any ideas? |
00:13.40 | Dorphalsig | can anybody give me a hand? |
00:14.28 | *** join/#asterisk dlublink (n=dlublink@modemcable114.38-201-24.mc.videotron.ca) |
00:14.50 | dlublink | how do I disable "Native Bridge"s in asterisk |
00:14.56 | dlublink | everytime that message appears I lose all audio |
00:15.23 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
00:15.28 | Dorphalsig | native bridge is a "flash" where users redirect |
00:15.40 | Dorphalsig | the call to another extension |
00:15.54 | file[laptop] | no, it isn't |
00:16.01 | delmar | hey guys.. i just did a subversion update of the latest Asterisk, on a little debian box.. compiled/installed it all no problem... used make samples... slapped an entry in sip.conf for a grandstream phone (the entry works fine on another box)... for context "default".. and when dialing the demo, it shows it working on the CLI but there is no audio. |
00:16.04 | delmar | any ideas? |
00:16.18 | file[laptop] | dlublink: what's the two protocols? SIP? |
00:17.45 | MoutaPT | delmar NAT? |
00:17.58 | delmar | nevermind. ill copy the entire working config dir from the working box. if its still broken ill blow away the debian install. |
00:18.06 | delmar | MoutaPT, nope. local LAN. |
00:18.40 | delmar | MoutaPT, what got me started on this was.. like.. ok the intention was to setup a second Asterisk box to test some hardware... specifically im having some quirks with the TDM400 on my main box.... |
00:19.10 | delmar | So I installed the second box, and set it up with an IAX between the main one.. and tested demo/echo test etc... and i got silence... |
00:19.20 | MoutaPT | delmar i'm on the phone brb |
00:19.39 | delmar | so i wanted to prove the second box was working.. so dumped that idea.. and hooked up a SIP phone to the second box.. sure enuf.. same problem. |
00:19.40 | delmar | ok |
00:19.50 | delmar | im gonna go mess with stuff anyway. bbl |
00:19.58 | Dorphalsig | Hi! |
00:20.11 | Dorphalsig | I need to have a PHP script trigger a dialing process |
00:20.17 | Dorphalsig | is there any API I can use? |
00:20.20 | Dorphalsig | I tried PHPAGI |
00:20.39 | Dorphalsig | but it seems its only to be used as an AGI in the dialplan :$ |
00:24.31 | glm2k | Dorphalsig: i think what you are looking for is a .call file |
00:25.05 | glm2k | Dorphalsig: mine for example is a simple bash script i plugged into nagios |
00:25.30 | glm2k | Dorphalsig: so a PHP routine should be easy to do |
00:29.58 | brookshire | Dorphalsig: /var/spool/asterisk/outgoing |
00:31.01 | mzo | anyone familiar with fwd, does it really take 24 hours for your registration for iax to work? |
00:32.16 | *** join/#asterisk websae (i=icechat5@CPE-24-167-204-30.wi.res.rr.com) |
00:32.50 | glm2k | mzo: i don't remember it being that long. |
00:33.45 | *** join/#asterisk yoll (n=BoraScrp@85.108.204.58) |
00:34.38 | mzo | i did it two days ago, and it still says rejected? =( i want to call people! |
00:38.23 | *** part/#asterisk themikester60 (n=mikey@209-83-240-50-static.dsl.oplink.net) |
00:39.26 | glm2k | mzo: i just checked my archives. it was connected on the same day |
00:39.39 | mzo | got any ideas what I'm doing wrong? |
00:40.33 | Abydos313 | is your firewall blocking? |
00:40.59 | mzo | um, don't think so, there's a POS linksys betwen it and the outside world and 43whatever 4358? i think it is open to the outside and forwarding back in |
00:41.00 | glm2k | iax? last i read it punches through firewalls. |
00:41.16 | mzo | iax, the firewall slayer (+5 to demons) |
00:41.19 | *** part/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
00:41.22 | glm2k | lol |
00:41.38 | glm2k | you must be a pally in your free time |
00:41.44 | mzo | angband addict :p |
00:41.54 | mzo | now if i could find a way to asterisk to play angband while I'm there :P |
00:41.55 | mzo | ... |
00:42.05 | glm2k | hehe |
00:43.08 | glm2k | iirc, fwd will allow you to register using SIP or IAX2 (or both? *shrugs*) |
00:43.22 | glm2k | can you register thru SIP? |
00:43.55 | mzo | i didn't try sip, they said to do iax2 because it was better? |
00:44.03 | glm2k | i agree |
00:44.11 | [hC] | man ive got this 7970 doing almost everything now |
00:44.12 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
00:44.19 | [hC] | custom ringers, background images, etc. |
00:44.25 | Abydos313 | Nice! |
00:44.42 | [hC] | Considering the little to no documentation on XML file formats it wants em all in, heh. |
00:44.45 | Abydos313 | how much did that phone set you back? |
00:44.47 | [hC] | I should write up an SCCP howto for all of this |
00:44.47 | *** join/#asterisk brockj49464_ (n=brockj49@63.87.56.235) |
00:44.52 | glm2k | in my case, i registered with SIP and only added IAX2 later when i couldn't get it to work. |
00:44.55 | [hC] | I think we pay 6-700 canadian for em |
00:45.06 | [hC] | so a fair penny |
00:45.11 | [hC] | but clients like them for their execs |
00:45.12 | Abydos313 | how much is that in USD |
00:45.29 | [hC] | 450-600 range |
00:45.37 | [hC] | probably low to mid 5's |
00:45.39 | Abydos313 | big bucks for one phone |
00:45.43 | [hC] | yep |
00:45.47 | Abydos313 | it must do alot |
00:45.47 | mzo | cheaper than a DHD |
00:45.47 | [hC] | color touch screen with camera support |
00:45.49 | glm2k | mzo: i just don't know why fwd had to hide the IAX2 option in the advanced settings page. |
00:45.50 | [hC] | gets em every time. |
00:45.59 | mzo | glm2k, so what else to do? :P |
00:46.06 | mzo | glm2k i did that with the iax2 option. and that was a day ago |
00:46.09 | Abydos313 | sweet, what kind of camera support? only cisco or any |
00:46.14 | [hC] | Cisco only i believe. |
00:46.20 | Abydos313 | how much |
00:46.20 | [hC] | Ive not found/tried/looked for any |
00:46.31 | [hC] | I know nothing about hte camera accessory at all, sorry.. |
00:46.41 | glm2k | mzo: run a debug? or ethereal? |
00:46.44 | Abydos313 | worth asking to see what they are going for |
00:47.05 | Abydos313 | so are they worth the money? |
00:47.34 | mzo | debug jussays the same thing it's sadi |
00:47.41 | Abydos313 | that's same price and new dell with flatscreen :)) |
00:48.45 | glm2k | mzo: rejected due to what? |
00:48.52 | glm2k | mzo: it could be a codec issue. |
00:50.07 | mzo | http://pastebin.com/566022 that's what i get. |
00:54.37 | mzo | any ideas? :P |
00:55.03 | mzo | http://pastebin.com/566033 that's the complete dump of a reject |
00:55.24 | *** join/#asterisk BrianUT (n=sniffer@c-67-166-96-54.hsd1.ut.comcast.net) |
00:56.48 | glm2k | i can't find what cause code 29 stands for :( |
00:57.33 | websae | any suggestions why my sip phone or my sip softphone won't connect to my asterisk server that i just compiled, i did sip debug, nothing comes up, as if no packets are hitting the server and coming up---my phones register fine with my other asterisk server.....any suggestions, anyone? |
00:57.41 | *** join/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
00:57.41 | mzo | bleh, is that bad? :P |
00:58.06 | *** join/#asterisk Gir19 (i=Gir@67.189.110.174) |
00:58.36 | glm2k | mzo: nah, just have to find what it is trying to say. |
00:59.25 | MoutaPT | G723 is available free for ASterisk? |
00:59.45 | websae | any suggestions anyone? |
00:59.56 | glm2k | did you double check your password? |
01:00.00 | mzo | yes |
01:00.07 | websae | anyone know why my asterisk server can't see my sip phones trying to connect? |
01:00.09 | mzo | if i change it on the site it should be right? |
01:00.10 | websae | any suggestions? |
01:00.15 | websae | i just compiled asterisk--- |
01:00.24 | glm2k | webmind: that on a lan? |
01:00.29 | glm2k | er, sorry. |
01:00.33 | glm2k | websae: that on a lan? |
01:00.43 | websae | at my colo |
01:01.25 | *** part/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
01:01.38 | glm2k | mzo: no reason for you to change it. it could be that your number has not been provisioned yet. fwd was down yesterday i think. |
01:01.55 | mzo | I did http://asteriskathome.sourceforge.net/handbook/index.html and the instructions for fwd, and no joy |
01:01.59 | mzo | it's been like this since sunday or so |
01:02.37 | websae | i don't know what's up with it |
01:02.41 | websae | why nothing will connect |
01:02.58 | websae | i can't see anything in the sip debug for registration packets or anything |
01:03.31 | glm2k | websae: firewall perhaps? or did your previous * version work fine? |
01:03.48 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
01:04.06 | websae | shouldn't be an issue, registers with my other colo fine |
01:04.24 | websae | it's odd because i see nothing at all |
01:05.44 | glm2k | websae: "nothing at all" sounds like a firewall. |
01:05.52 | glm2k | websae: specialyl with sip |
01:06.30 | Gir19 | websae: have you also checked to make sure the ip address hasn't been changed as well. |
01:06.56 | websae | yep |
01:07.06 | websae | the ip addresses are static |
01:07.19 | websae | i can't see anything, this quite odd |
01:07.22 | websae | *this is |
01:07.31 | brockj49464 | Asterisk does NOT do TOS on SIP correct? I have to use iptables to tag the packets correct? |
01:07.57 | justinu | it doesn't do TOS on RTP packets |
01:08.00 | justinu | but does on SIP |
01:08.04 | websae | any other suggestions |
01:08.05 | justinu | iirc |
01:08.13 | websae | it's not a firewall issue |
01:08.36 | websae | cause i should be able to at least see the packets hitting the server |
01:08.37 | websae | but nothing is |
01:08.59 | Gir19 | sounds to me more of a routing issue than a firewall issue. |
01:09.16 | websae | routing issue |
01:09.21 | websae | on the server end? |
01:09.54 | brockj49464 | what is the setting? Can it be global? Does it work if * is running as asterisk and not root? |
01:10.13 | *** join/#asterisk GoRK (i=GoRK@ip68-111-119-231.lu.dl.cox.net) |
01:10.18 | Gir19 | no, I would believe it would be on the phone side. I wasn't here for everything you said so I do not know exactly what your setup is or how I can really help. |
01:10.26 | websae | what kind of routing issue could it be |
01:10.59 | GoRK | anyone know of a softphone SIP or IAX that has a push-to-talk capability -- ie you can join a conference call or something but you have to hold a button for it to transmit audio? |
01:11.10 | websae | i have an asterisk server at a COLO that i just compiled asterisk on |
01:11.24 | GoRK | not "real PTT" like cell phones or the like |
01:11.28 | websae | i tried setting up a atest account and setting up my sip phone to connect |
01:11.44 | websae | but it doesn't register |
01:11.52 | websae | i don't see anything trying to register in the CLI |
01:12.44 | websae | i then tried with my softphone---still same thing |
01:12.54 | Gir19 | websae: how is the phone trying to connect? IE: phone > router > internet > router > * |
01:12.55 | websae | i don't see anything trying to register at all |
01:13.15 | websae | well my phones work fine---i can connect to my other asterisk server that is at a different colo just fine |
01:13.22 | *** join/#asterisk peanuter (n=saasdf@216.176.177.138) |
01:13.31 | GoRK | websae: i assume you have enabled debug with 'sip debug' ? |
01:16.10 | websae | correct |
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01:16.16 | websae | and set verbose to 10000000000000000000000000 |
01:16.16 | websae | lol and nothing |
01:16.17 | Gir19 | mmm, netsplit |
01:16.17 | delmar | great |
01:16.17 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) [NETSPLIT VICTIM] |
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01:16.19 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
01:16.19 | delmar | [7012] |
01:16.19 | delmar | type=friend ; either "friend" (peer+user), "peer" or "user" |
01:16.19 | delmar | regexten=7012 |
01:16.19 | delmar | context=testing |
01:16.19 | delmar | secret=7012 |
01:16.20 | delmar | callerid="Testphone" <7012> |
01:16.20 | delmar | host=dynamic ; we have a static but private IP address |
01:16.20 | GoRK | websae: the only thing i know to tell you is that if you dont see any debug messages coming through it's possible that the colo is blocking 5060/udp or something.. the only other thing is that your sip UA is misconfigured.. you should at least see something there |
01:16.20 | delmar | nat=no ; there is not NAT between phone and Asterisk |
01:16.20 | glm2k | holy macaroni, that's some split |
01:16.20 | delmar | canreinvite=no ; allow RTP voice traffic to bypass Asterisk |
01:16.20 | delmar | dtmfmode=info ; either RFC2833 or INFO for the BudgeTone |
01:16.20 | delmar | ;mailbox=7012@local ; mailbox 1234 in voicemail context "default" |
01:16.20 | delmar | disallow=all ; need to disallow=all before we can use allow= |
01:16.20 | delmar | allow=ulaw |
01:16.20 | delmar | allow=g729 ; Pass-thru only unless g729 license obtained |
01:16.20 | justinu | ~pb |
01:16.20 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
01:16.20 | delmar | qualify=50 |
01:17.08 | delmar | defaultip=10.20.1.248 |
01:17.08 | delmar | grr |
01:17.08 | delmar | sorry. hit the wrong key guys |
01:17.09 | GoRK | argh ever heard of a pastebot? |
01:17.09 | websae | UA----what's that? |
01:17.09 | justinu | supposedly there's some net backbone problem |
01:17.09 | GoRK | user agent -- your softphone or hardphone or whatever is trying to register |
01:17.09 | websae | they are all setup right |
01:17.09 | websae | just is frustrating i guess, not knowing what's blocking this |
01:17.09 | Gir19 | websae: what is the physical configuration? |
01:17.09 | websae | how can i tell if that port is open? |
01:17.09 | GoRK | well you can tell for sure with netcat |
01:17.09 | websae | how does that work? |
01:17.09 | GoRK | stop asterisk on the box, set netcat as a listener on port 5060/udp and then use netcat somewhere else to send it some data |
01:17.09 | delmar | i dunno. i give up. same asterisk, zaptel, and libpri source AND same configs.... running on a second box... setup a sip phone.. and nothing but silence.. no audio. |
01:17.09 | websae | netcat is that on fedora core |
01:17.09 | websae | can you give me an example GoRK |
01:17.10 | websae | of what to do? |
01:17.10 | GoRK | netcat is available for everything; whether or not it's installed by default .. can't help you there |
01:17.10 | justinu | yum install netcat |
01:17.50 | *** join/#asterisk slePP (n=slepp@S0106000f663692da.ed.shawcable.net) |
01:17.55 | GoRK | websae: on the server: nc -l -p 5060 -u |
01:18.21 | GoRK | websae: on the client echo test | nc -u -p 5060 your.server.name |
01:18.41 | mitcheloc | does anyone here know if the sipura devices have an API to them? |
01:19.05 | GoRK | websae: make sure asterisk isnt running on the server and nothing else is listening on UDP/5060 .. netstat -l -p will tell you if something else is on udp/5060 |
01:19.21 | GoRK | websae: if traffic flows you will see "test" echoed to the remote terminal |
01:19.27 | websae | ok |
01:19.29 | websae | let me try this |
01:20.02 | Gir19 | mitcheloc: from what I can tell about sipura's is that you can only access them via http. |
01:20.33 | mitcheloc | Gir19: i guess what i'm asking is, can i initiate acall on them by sending a command, via http or anything? |
01:21.45 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
01:21.55 | GoRK | websae: the only other problem may be if you have one or more sip UAs behind the same nat device and trying to speak with multiple servers outside the firewall.. you could have problems depending on how your nat device handles UDP traffic |
01:22.25 | websae | GorK: that didn't work on the client side |
01:22.29 | websae | that command you told me to do |
01:22.43 | GoRK | sorry what was the error |
01:22.54 | websae | just brought up the nc commands |
01:22.57 | GoRK | im not on a machine with netcat right now |
01:22.57 | GoRK | oh |
01:23.10 | GoRK | hang on |
01:23.13 | websae | ok |
01:23.24 | Gir19 | mitcheloc: so you mean you are trying to initiate a call by a command instead of just picking up a handset and dialing? |
01:23.38 | mitcheloc | Gir19: yes, i tend to get very lazy ;) |
01:23.47 | MoutaPT | is it G723 available free to Asterisk? |
01:24.07 | GoRK | websae: ok sorry the client syntax should be: |
01:24.14 | websae | ... |
01:24.14 | websae | yep |
01:24.15 | websae | ? |
01:24.20 | GoRK | echo test | nc -u your.server.here 5060 |
01:24.38 | GoRK | the -p argument is only the port on the listener |
01:24.54 | *** join/#asterisk _deg_ (n=deg@200-233-51-145.corp.ajato.com.br) |
01:25.15 | websae | hrm nothing on the server side |
01:25.24 | Gir19 | mitcheloc: I am trying to understand this a little better, so you want to start a call via command that will do what exactly? call the number you want and call your handset at the same time, based on a timer or something? |
01:25.53 | GoRK | websae: the client will go ahead an exit; since udp is stateless it just sends the packets out.. but you should have seen something on the server screen if traffic was getting through on the port |
01:26.01 | mitcheloc | Gir19: yes, i just tell it to call another extension, so first it calls my handset, i pickup, then it rings another extension |
01:26.04 | websae | hrm |
01:26.06 | websae | nothing at all |
01:27.03 | websae | GoRK: could try sending a packet |
01:27.06 | GoRK | websae: do this on the server: netstat -l -p -n | grep 5060 |
01:27.07 | mzo | does anyone have a clue why I can't register to FWD? It's been like this since Sunday? |
01:27.14 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
01:27.21 | Gir19 | mitcheloc: the sipura does not have any way of doing that, it could be setup in asterisk to do something like that, but the sipura does not have any type of programable area for such a task. |
01:27.25 | GoRK | websae: that will tell you if anything else is listening on udp/5060 |
01:27.58 | GoRK | if you run it while netcat is listening you should see nc in the list |
01:27.58 | websae | went to the next line |
01:28.20 | GoRK | ok so it's not another app on the machine... i'd ask your telco if port 5060/udp is open |
01:28.20 | mitcheloc | Gir19: aww, well i know how to set up asterisk to do that, do you know, maybe a fake udp packet sent to the spa? thinking it came from the asterisk box to initiate the call...? |
01:28.28 | GoRK | err isp/colo whatever |
01:28.32 | GoRK | not the telco |
01:28.37 | GoRK | unless maybe they are one in the same |
01:28.47 | websae | so it's their side |
01:28.54 | websae | darnit |
01:29.06 | websae | hey well thanks for your help GoRK :)! |
01:31.33 | *** join/#asterisk Chai_Sangeen (n=Chai@c-24-61-4-191.hsd1.ma.comcast.net) |
01:31.40 | Chai_Sangeen | hello everyone |
01:31.42 | Gir19 | mitcheloc: the only way I know of that you could do that type of thing would be to have asterisk call your extension, then wait for a pickup then initiate the next line to call the other extension. so theoreticaly it would create the circut for the 2 extensions to talk to eachother. |
01:32.03 | mitcheloc | Gir19: i was hoping to bypass asterisk if possible |
01:33.35 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
01:33.44 | Gir19 | mitcheloc: unfortunatly the sipura will only start or stop a call when it's extension is dialed, but if you send a fake udp packet it would answer, but then just hangup. |
01:34.30 | mitcheloc | ahh good point |
01:34.55 | mitcheloc | perhaps I should e-mail and ask them to expand their firmware ;) |
01:34.57 | Gir19 | mitcheloc: it has to have a pre existing circut to continue the call otherwise it sees it as junk traffic on the network. |
01:36.37 | *** join/#asterisk riddlebox (n=victoria@24-171-11-166.dhcp.stls.mo.charter.com) |
01:36.40 | mitcheloc | how would someone go about contacting a developer or project manager (i guess whoever makes decisions at sipura?)? |
01:36.56 | robin_sz | oh, bucking follocks :( |
01:37.05 | file[laptop] | mitcheloc: sacrifice a goat, or pig |
01:37.10 | Gir19 | lol |
01:37.15 | riddlebox | is it possible to start an agi script without calling into asterisk first? |
01:37.27 | mitcheloc | file: i would seriously do that because i'm actually serious about this though |
01:37.30 | robin_sz | 2 hours spent messing around setting up presentation numbers etc, and my provider has sett it all to "withheld number" .. grr |
01:37.35 | GoRK | file: better than getting in touch with someone like that at cisco. you have to sacrifice a human baby |
01:38.02 | mitcheloc | well what are the chances someone in here can put me in contact with one of those people? ;) |
01:38.05 | websae | file: how much to port an 800 number? |
01:38.13 | file[laptop] | websae: $25 |
01:38.22 | file[laptop] | plus $1.95/mth standard fee |
01:38.28 | file[laptop] | and another $25 if you want it NOW NOW NOW FAST FAST FAST |
01:38.33 | robin_sz | and your first-born |
01:38.34 | Gir19 | GoRK: Sipura is an affiliate of Cisco. lol |
01:38.42 | websae | what's the incoming rate on that? |
01:38.43 | mishehu | anybody running asterisk on mini-itx? I want to know if it works well, and am trying to make sure that a casetronic chassis that I'm looking at gives out the correct power. |
01:38.51 | file[laptop] | 2 cents per minute, 6/6 |
01:39.02 | *** part/#asterisk freat (n=freat@h-72-244-84-43.chcgilgm.covad.net) |
01:39.04 | file[laptop] | for US48 |
01:39.07 | file[laptop] | inbound... |
01:39.21 | websae | alright, thanks much |
01:39.23 | websae | what's the link |
01:39.27 | mitcheloc | Do you think if I flew out to their offices they would speak with me? |
01:39.36 | Gir19 | mitcheloc: the only way I know is to use there webcomments / questions email addy. sipura-webmaster@cisco.com |
01:39.37 | file[laptop] | for porting? there is no link really, you just email me.... |
01:39.42 | file[laptop] | it's a manual procedure thing |
01:39.46 | websae | ah |
01:39.51 | file[laptop] | I talk nice to certain people and they get it done for me quickly |
01:39.56 | file[laptop] | because they know otherwise I'll hurt them |
01:40.12 | *** join/#asterisk livinded (n=livinded@cpe-24-24-190-252.socal.res.rr.com) |
01:40.22 | mitcheloc | sipura's offices are in san jose mm...thats not too far from orange county ;) |
01:40.32 | mitcheloc | lol ya'll probably tired of me rambling about this |
01:40.38 | tuxinator_linux | mitcheloc: you in OC? |
01:40.45 | mitcheloc | yes sir |
01:41.00 | tuxinator_linux | a bunch of us are |
01:41.01 | livinded | if i use the regular source can i use "make update" to update to newer versions or dos thta only work with the cvs builds? |
01:41.41 | mitcheloc | tuxinator, are there any groups or anything for asterisk around here? |
01:42.10 | tuxinator_linux | Let's start on |
01:42.15 | tuxinator_linux | one |
01:42.26 | *** join/#asterisk SwK (n=Silik0nJ@12-219-147-107.client.mchsi.com) |
01:42.33 | Gir19 | I am still being annoyed by my fax and poping noise on my asterisk server. I am starting to run out of ideas. |
01:42.40 | mitcheloc | that sounds good to me, but i only know you, and one or two other people that i just slowly started introducing to asterisk |
01:42.56 | SwK | who's doing that asterikast thing? |
01:43.13 | *** join/#asterisk asteriskmonkey (n=phil@69.158.149.183) |
01:43.20 | asteriskmonkey | hey |
01:43.21 | file[laptop] | SwK: I dunno but Russell and I were making fun of it earlier today :D |
01:43.21 | Gir19 | I've got a whole team I direct for asterisk here in Portland, OR |
01:43.36 | mitcheloc | Gir19: mind if i pm you? |
01:43.37 | SwK | file[laptop] someone needs to explain to him but TDM means |
01:43.38 | tuxinator_linux | asterikast? |
01:43.41 | asteriskmonkey | anyone know much about ztmonitor ? and where volume levels should be for good echo cancellation? |
01:43.51 | Gir19 | mitcheloc: go for it. |
01:43.53 | file[laptop] | SwK: too technical! |
01:44.23 | Gir19 | asteriskmonkey: they should be as close to center as possible. |
01:44.23 | SwK | asteriskmonkey there is no specific gain level ... it all depends on the switch you are connected to |
01:44.23 | file[laptop] | unfortunately the server is overloaded so I can't download the episode |
01:44.48 | SwK | file[laptop] they have a pretty fast torrent |
01:45.01 | file[laptop] | I'm too lazy though |
01:45.02 | tuxinator_linux | SwK: post it |
01:45.10 | tuxinator_linux | SwK: the torret link |
01:45.30 | file[laptop] | SwK: thankies |
01:45.34 | SwK | the torrent link is on their website |
01:45.36 | asteriskmonkey | SwK Gir19 : its a pri connect right at a colocation been fighting with echo for 8months now .. have a te110p and a te406p both have echo issues... the graphical crap shouldnt be trusted i read its a numberic value right like 14555 is the limit? |
01:45.55 | SwK | asteriskmonkey: have you talked to your carrier? |
01:46.19 | SwK | asteriskmonkey: you could be over-driving |
01:46.36 | SwK | echo is one of those PITA things to get rid of |
01:46.46 | asteriskmonkey | SwK: carried says its not them .. over driving? mmm i dont think so my rx and tx are in the - range |
01:47.05 | file[laptop] | I have a can of Echobgone(tm)... I'll sell it to you! |
01:47.21 | SwK | asteriskmonkey: I have 1 box with a T1 from qwest and -9 of pad on the RX and -6 on the TX side |
01:47.28 | xachen | what you guys making fun of? |
01:47.41 | GoRK | i hate echo; i always seem to have it and everyone else always seems to not have it. . it's like i cant get rid of it |
01:47.43 | X-Rob | file[laptop], geez, I've been looking for EchoBgone(tm) for ages! |
01:47.48 | file[laptop] | I shouldn't download this here... |
01:47.51 | X-Rob | Is that the aerosol or the roll-on? |
01:47.51 | GoRK | i have an ata186 that echos no matter what i do |
01:47.54 | file[laptop] | I should download it on the Mini |
01:48.06 | SwK | xachen people doing howto videos and not knowing proper terminology |
01:48.10 | file[laptop] | although it's working... |
01:48.11 | xachen | oh |
01:48.14 | xachen | Asterikast or whatever? |
01:48.17 | file[laptop] | yeah |
01:48.18 | SwK | teah |
01:48.25 | xachen | I heard something bout that |
01:48.27 | asteriskmonkey | swk : im at -4rx and -8tx |
01:48.28 | xachen | What did he confuse? |
01:48.44 | justinu | asteriskmonkey: what kind of phones? |
01:48.59 | SwK | i forget what he said but when he defined TDM is wasnt time domain mux'ind |
01:49.03 | asteriskmonkey | ive gone throught, iaxys, polycoms, atcoms.. everyting :P |
01:49.21 | file[laptop] | SwK: did he? |
01:49.21 | justinu | asteriskmonkey: :( |
01:49.21 | SwK | it was some king of time derived multiplication ro something |
01:49.22 | file[laptop] | O.o |
01:49.30 | SwK | yeah |
01:49.37 | xachen | haha |
01:49.38 | file[laptop] | time division multiplexing foreva! |
01:49.38 | asteriskmonkey | its not all the time so its 100% of the time though so its anoying |
01:49.50 | asteriskmonkey | ive request a 1004hz test so i can fine tune |
01:50.25 | justinu | please clarify: (17:49:47) asteriskmonkey: its not all the time so its 100% of the time though so its anoying |
01:50.35 | asteriskmonkey | echo is not all the time |
01:50.42 | SwK | asteriskmonkey: have you checked all the typical hardware stuffs like missing interupts etc and so forth, made sure its just not the user over driving his handset or having 2 loud phones right next to each other etc etc |
01:50.43 | asteriskmonkey | and it is usually only on 1 end at a time |
01:50.48 | justinu | ok |
01:51.04 | justinu | equally on far end vs. near end? |
01:51.05 | asteriskmonkey | SwK: yep checked all that :P |
01:51.11 | Chai_Sangeen | I have my spa3k behind a nat in a remote location and it working nicely in/out calls, I've been strugling trying to get the spa3k IP updated in asterisk. Here is a link that of a person having the same problem on the mailing list http://lists.digium.com/pipermail/asterisk-users/2005-September/126386.html |
01:51.31 | SwK | well turn down the tx 3 more DB |
01:51.50 | asteriskmonkey | justinu: yes , only happens usually on orginator .. so if i call i get echo if someone calls me i dont but they do |
01:52.01 | SwK | asteriskmonekey: is this a local PRI or a dedicated LD circuit? |
01:52.20 | asteriskmonkey | its a local pri in a backbone facitily NI2 flavour |
01:52.22 | *** join/#asterisk ManxPower (n=ewieling@24-179-48-91.static.slid.la.charter.com) |
01:52.47 | lunaphyte | tonight is dumb question night, right? |
01:52.58 | justinu | k |
01:53.45 | lunaphyte | why would i want more than 1 context? |
01:53.50 | asteriskmonkey | its been 8months and spent a shit load on the te406 in hopes of echo cancelling love with no luck :( |
01:54.00 | asteriskmonkey | and digium guys around? |
01:54.01 | Gir19 | nothing is a dumb question unless you have not Read the FM |
01:54.01 | justinu | wow |
01:54.07 | SwK | so then get an outboard echo can |
01:54.08 | justinu | asteriskmonkey: sounds like a nightmare |
01:54.33 | *** join/#asterisk froguz (n=froguz@224-139-222-201.adsl.terra.cl) |
01:54.35 | lunaphyte | i guess it's less of a 'how do i do this' and more of a 'what are some applications' question... |
01:54.35 | Corydon76-home | asteriskmonkey: what kind of circuit are you trying to echo cancel? |
01:54.39 | mitcheloc | file: may i pm you a quick question? |
01:54.40 | asteriskmonkey | justinu: yep :P 20k in bills and still an echo syste |
01:54.41 | justinu | you guys know that rx/txgain parameters are not dB, right? |
01:54.44 | Chai_Sangeen | The only problem is that I have to manully change the spa3k IP if the spa3k ip changes, to place an outgoing call |
01:54.51 | asteriskmonkey | Corydon76-home: PRI NI2 |
01:55.01 | justinu | asteriskmonkey: i'm worried I might be in a similar situation :( |
01:55.05 | Corydon76-home | lunaphyte: do you want people who call into your system to be able to make long distance calls out? |
01:55.27 | Corydon76-home | asteriskmonkey: odd, I rarely have echo problems with a PRI |
01:55.49 | Corydon76-home | asteriskmonkey: Are you hearing echo or is the other side hearing echo? |
01:55.53 | Gir19 | lunaphyte: you would want to have more than one context to help seperate networks or help keep track of things and better documentation built in. |
01:55.57 | asteriskmonkey | both |
01:56.13 | Corydon76-home | asteriskmonkey: are you using echotraining? |
01:56.14 | lunaphyte | Corydon-w: i think yes, but not just anyone. |
01:56.15 | SwK | I think he's probably oer driving since he's in a colo facility probably with little more then a corss connect from the carrier |
01:56.30 | asteriskmonkey | yes |
01:56.35 | Corydon76-home | lunaphyte: Now you see why there's a need for multiple contexts |
01:56.38 | asteriskmonkey | well my bars arnt high at all in ztmonitor |
01:56.46 | SwK | and by his statement he's tried 'everything else' |
01:56.54 | lunaphyte | Gir19: ah. i see. |
01:57.12 | lunaphyte | Corydon76-home: thanks for the example. |
01:57.18 | justinu | what are typical rx/txgain parameters for PRIs from a ILEC CO? |
01:57.21 | justinu | not inside a colo facility |
01:57.46 | SwK | justinu: it varies from switch to switch and connection to connection |
01:57.59 | justinu | is there a typical range at all? |
01:58.08 | SwK | i've set gains everywhere from -12 to +6 |
01:58.19 | justinu | what do you use to tweak it? milliwatt generator? |
01:58.28 | SwK | my years |
01:58.30 | Corydon76-home | justinu: trial and error |
01:58.37 | froguz | somebody know were can i buy an Ambient MD3200 modem? |
01:58.37 | SwK | err ears |
01:58.41 | Corydon76-home | justinu: you adjust each until it sounds right |
01:58.42 | justinu | trial and error sounds awfully primitve |
01:58.49 | SwK | froguz ebay |
01:58.53 | asteriskmonkey | what is the numberic value suppost to be on call im using zt monitor and everything looks fine |
01:58.58 | Corydon76-home | justinu: There really isn't a better way |
01:59.00 | justinu | how do i know what sounds right? compare to a POTS call? |
01:59.01 | asteriskmonkey | the bars dont go to the end |
01:59.18 | Corydon76-home | asteriskmonkey: you adjust it until it "sounds right" |
01:59.39 | justinu | how do you know when it "sounds right"? |
01:59.41 | SwK | ztmonitor its not a very goot tool for adjusting gains... it just shows you that you are getting something |
01:59.49 | asteriskmonkey | thanks ive been doing that for 8 months without success |
01:59.51 | Corydon76-home | Somebody can do all the engineering in the world, but in the end, if it doesn't sound right, it's not done. |
01:59.52 | Gir19 | justinu: I usual try to use ztmonitor and set the rx/tx gains to make it about midway in the monitor, this is because I have run into some locations that it will have issue detecting DTMF tones or hangups on cellphones. |
01:59.54 | asteriskmonkey | any logical method |
01:59.57 | SwK | justinu: when you can hear it loud enuff but dont get echo |
02:00.02 | froguz | SwK, ebay search: 0 items found for |
02:00.02 | froguz | ambient md3200 |
02:00.07 | justinu | k, so start low and go up |
02:00.13 | Corydon76-home | asteriskmonkey: Set your gains to -10 each |
02:00.34 | asteriskmonkey | gah ok did that a while ago.. will try again |
02:00.35 | Gir19 | justinu: I normally go in incraments of +/- 4 |
02:00.37 | SwK | froguz: then search for x101 or x100 clone since thats what you are really asking for |
02:00.42 | justinu | increments of 4, ok |
02:00.43 | Corydon76-home | asteriskmonkey: if you still have echo problems at that point, your telco needs to run a line test |
02:00.57 | justinu | what type of line test? BERT? |
02:01.18 | Corydon76-home | justinu: the telco doesn't usually give us that information |
02:01.35 | justinu | it's a t1 so, BERT should be appropriate |
02:01.43 | Gir19 | justinu: I would try more than one type of line test just to get slightly better readings. |
02:01.45 | SwK | the scale on the gains is db... every 3 db you change it you double or half the power level |
02:01.53 | justinu | it's /not/ db tho |
02:02.12 | justinu | the scale is percentage of whatever the amplifier algorithm can do |
02:02.17 | justinu | -100 to +100 |
02:02.19 | lunaphyte | if a phone is in more than one context, what dictates which context is applied? |
02:02.36 | SwK | i guess i should rfsc closer |
02:02.36 | Corydon76-home | lunaphyte: depends upon the channel type |
02:02.44 | asteriskmonkey | ok -10 on each and im getting echo |
02:02.48 | asteriskmonkey | damn it |
02:02.50 | lunaphyte | let's say a sip channel |
02:03.02 | Corydon76-home | lunaphyte: for SIP, it's whichever one finds a match first |
02:03.04 | Gir19 | lunaphyte: and the channels configuration |
02:03.19 | lunaphyte | so - first in the list kind of thing? |
02:03.21 | Corydon76-home | lunaphyte: for IAX2, you specify the context you want to search |
02:03.25 | _Sam-- | im messing around trying to learn how to make 2 *'s talk to eachother.. Server B registers to Server A fine. But when server A tries to call server B, server B issues CAUSE : No authority found and refuses the call...where should i look? |
02:03.31 | lunaphyte | ok |
02:04.05 | Gir19 | lunaphyte: usualy asterisk will go in priority of which context was listed first for that extension then continue down the line. |
02:04.17 | lunaphyte | can an extension in 1 context be configured to point to a different context? |
02:04.32 | *** join/#asterisk themikester60 (n=mikey@209-83-240-50-static.dsl.oplink.net) |
02:05.00 | Corydon76-home | lunaphyte: That's a Goto |
02:05.05 | lunaphyte | ah |
02:05.09 | Corydon76-home | lunaphyte: or an include |
02:05.14 | justinu | SwK: unfortunatly it's not even clear from the source |
02:05.45 | Corydon76-home | asteriskmonkey: like I said, schedule a line test with the telco |
02:06.00 | themikester60 | is there a way in asterisk to set a variable from a file on the hard drive? |
02:06.04 | Corydon76-home | For such a noisy line, they're likely to find something |
02:06.32 | justinu | the only thing I could see affecting echo on a typical HDSL PRI circuit is latency |
02:06.45 | Gir19 | Corydon76: that depends on the phone company. |
02:06.47 | justinu | slips/frame errors typically will cause clicks or pops |
02:07.03 | asteriskmonkey | no issue using sip provider |
02:07.09 | Corydon76-home | Gir19: true |
02:07.39 | Corydon76-home | Gir19: we have a few CLEC's that we've found to be grossly incompetant |
02:07.47 | Corydon76-home | (but that could be just the local office) |
02:08.05 | Corydon76-home | *cough*Birch*cough* |
02:08.32 | Corydon76-home | *cough*Xpedius*cough* |
02:08.43 | Corydon76-home | Pardon me... had some phlegm in my throat |
02:09.44 | *** join/#asterisk outtolunc (n=me@adsl-69-110-25-46.dsl.pltn13.pacbell.net) |
02:09.45 | *** join/#asterisk Kizmet (i=Kizmet@freematrix/sponsor/kizmet) |
02:10.57 | Kizmet | Does anyone know which headers the Grandstream GXP-2000 accepts if any to alter the Missed Calls display. Im running 1.0.2.8 Firmware and Asterisk 1.2.4 |
02:11.26 | Kizmet | And |
02:11.41 | _Sam-- | the only you can do with the missed calls display is turn it on or off |
02:11.46 | Kizmet | Does anyone know of how to add a HINT context in AEL ? |
02:11.49 | *** join/#asterisk firestrm (n=no@S01060013105c1c69.gv.shawcable.net) |
02:11.51 | asteriskmonkey | Corydon76-home: didnt know it was possible to get echo on pri :P |
02:11.57 | Kizmet | _Sam--, i just want to turn it off heh. |
02:11.59 | _Sam-- | you cant modify any sip headers for the gxp |
02:12.03 | _Sam-- | you can turn it off |
02:12.06 | _Sam-- | its on a per account basis |
02:12.11 | _Sam-- | go to account 1 |
02:12.31 | _Sam-- | from the web ui |
02:12.42 | Corydon76-home | asteriskmonkey: I didn't know it was possible for CLECs to suck so badly and stay in business |
02:12.43 | *** join/#asterisk cyonics (i=cyon@tx-71-52-77-114.dhcp.sprint-hsd.net) |
02:12.46 | _Sam-- | Disable Missed-Call: No Yes (Missed calls NOT recorded) |
02:12.52 | Corydon76-home | asteriskmonkey: and yet... |
02:12.55 | Kizmet | _Sam--, cheers. |
02:12.56 | *** join/#asterisk doushanes (n=dwright@c-71-194-48-241.hsd1.il.comcast.net) |
02:12.58 | asteriskmonkey | So they do ? |
02:13.10 | Corydon76-home | asteriskmonkey: and not just CLECs, either |
02:13.12 | asteriskmonkey | like youve had a bad pri beofre? ive had dead d channels but thats it |
02:13.23 | _Sam-- | good luck with hints and BLF on your gxp |
02:13.26 | justinu | i've had PRIs with echo |
02:13.30 | justinu | common problem for me |
02:13.32 | Corydon76-home | The ILEC has the same issues, but everybody expects that out of them |
02:13.36 | ManxPower | Corydon-w, our CLEC was bought about 6 months ago. They started to REALLY SUCK about a week ago. |
02:13.45 | justinu | sounds like broadwing |
02:13.54 | asteriskmonkey | yes im with mci |
02:13.58 | asteriskmonkey | aka worldcom |
02:14.07 | Corydon76-home | aka Verizon |
02:14.13 | outtolunc | oh no, aka hell |
02:14.15 | _Sam-- | aka verizon business |
02:14.17 | ManxPower | Their new support group don't even know who we are. We used to be their 4th largest customer. |
02:14.19 | Kizmet | _Sam--, I was told that you could add a header to make the phone answer automagically for paging ? |
02:14.24 | [av]bani | \o> |
02:14.25 | [av]bani | <o/ |
02:14.51 | _Sam-- | Kizmet: ask [av]bani |
02:14.57 | asteriskmonkey | Kizmet: if you want blf and pagine get allworx stuff |
02:15.02 | [av]bani | o.o |
02:15.07 | ManxPower | Kizmet, your extensive search of the mailing list archives and Wiki did not turn anything up? |
02:15.28 | [av]bani | SIPAddHeader(Call-Info: answer-after=0) |
02:15.30 | Kizmet | ManxPower, I had a look on VoIP-Info.... |
02:15.38 | ManxPower | ~mailinglist |
02:15.38 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
02:15.41 | ManxPower | ~docs |
02:15.42 | jbot | docs is probably probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
02:16.43 | _Sam-- | [av]bani: how do you use that command? the wiki page for SIPAddHeader is pretty bleak: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SIPAddHeader |
02:17.31 | [av]bani | exten => s,1,SIPAddHeader(Call-Info: answer-after=0) |
02:17.34 | [av]bani | exten => s,2,Dial(${ARG2},20) |
02:17.35 | [av]bani | or so |
02:18.03 | [av]bani | gxp's will add the header if you punch page on them, though that only works for direct phone to phone |
02:18.15 | [av]bani | you can pick it up from the gxp and then set it for outgoing though |
02:25.29 | asteriskmonkey | Here is my zapata.conf |
02:25.31 | asteriskmonkey | http://pastebin.ca/42725 |
02:25.36 | asteriskmonkey | looks minty dont it |
02:25.48 | asteriskmonkey | think ill crap my foot up mcis ass shortly |
02:30.00 | asteriskmonkey | just as long as i can say its not my digium cards ill lay into them full tilt :D |
02:30.29 | justinu | i'm having the issue with a sangoma card |
02:30.34 | outtolunc | i must have missed the first part |
02:30.52 | asteriskmonkey | audiocodes gateway for the win |
02:31.24 | outtolunc | first thing i seen was |
02:31.27 | outtolunc | [18:11] <asteriskmonkey> Corydon76-home: didnt know it was possible to get echo on pri :P |
02:31.48 | asteriskmonkey | well am i right on that? or is that out to lunch |
02:31.59 | outtolunc | too which i replied something to the effect.. 'wonders why telco's have the term 'echo can's |
02:32.06 | justinu | the pri doesn't have anything to do with the echo, it's the endpoints |
02:32.15 | justinu | and how much energy you put into the network |
02:32.20 | asteriskmonkey | ok |
02:32.31 | asteriskmonkey | so i cant cram it up there ass... |
02:32.35 | mzo | WORK dammit |
02:32.48 | justinu | the EC built into some PRI cards helps alleviate echo |
02:32.57 | asteriskmonkey | well i was at -30 and i could hear stuff nor could anyone else |
02:32.57 | justinu | yeah, there's a tail and longhaul side |
02:33.08 | justinu | tail is your pbx to phone loop |
02:33.14 | *** part/#asterisk doushanes (n=dwright@c-71-194-48-241.hsd1.il.comcast.net) |
02:33.15 | justinu | long haul is pbx to co loop |
02:33.25 | asteriskmonkey | so its the tail loop then maybe |
02:33.26 | Corydon76-home | outtolunc: actually, I'm aware of echo on PRIs |
02:33.35 | asteriskmonkey | really good |
02:33.40 | Corydon76-home | outtolunc: I just find it a much bigger problem on analog lines |
02:33.48 | outtolunc | usually 'long haul' is used to talk about 'end points that are 500+ miles apart' |
02:33.53 | asteriskmonkey | i though i was goign insance i spent uber loot on a te406 to have 60ms of echo can on each channel |
02:33.57 | outtolunc | not sure what you are talkin about |
02:34.12 | justinu | 60ms isn't all that much |
02:34.26 | justinu | sometimes the echo tail length can exceed 128ms which is the itu standard echo can |
02:34.40 | *** join/#asterisk katakefalos (i=katakefa@194.214.77.65.in-addr.arpa.ethernext.com) |
02:34.48 | asteriskmonkey | so what change my echo can to 256 for the taps? |
02:34.55 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
02:35.31 | justinu | no idea |
02:36.03 | outtolunc | Corydon76-home, i wasn't worried about your side of the convo <G> |
02:36.04 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
02:36.10 | asteriskmonkey | local numbers it tends not to happen on anytime a cell or ld (as in 1 city away) calls happens its very apperent |
02:36.25 | lunaphyte | what does s@default mean in "IAX2/guest@misery.digium.com/s@default" |
02:36.25 | justinu | cell is notorious for increasing latency |
02:36.44 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
02:36.46 | Corydon76-home | outtolunc: that's okay, I'm sure my sarcasm is about to get flamed on the -dev list. |
02:36.58 | outtolunc | most people with pri's never have an issue, but there are some that have 'all the issues' |
02:37.18 | outtolunc | ut oh <G> |
02:37.20 | justinu | i've had enough PRIs |
02:37.24 | justinu | and i've always had these issues |
02:37.41 | outtolunc | right, and you are in the same place you always been |
02:38.10 | *** join/#asterisk |omni| (i=rob@net98.limelyte.net) |
02:38.16 | asteriskmonkey | lol |
02:38.22 | asteriskmonkey | location location location |
02:38.38 | outtolunc | rocket science 001 |
02:38.40 | Kizmet | I work for a company that has 4 sets of pri's |
02:38.56 | *** join/#asterisk samueltc (n=sam@Toronto-HSE-ppp3880832.sympatico.ca) |
02:38.57 | outtolunc | move away from the BLAST |
02:39.03 | Kizmet | it just happened that the 2 sets that we allocated for internal support lines have shit loads of echo :( |
02:39.19 | samueltc | is iax2-trunking is consuming less bandwidth than SIP? |
02:39.25 | Kizmet | yet the customer lines have none.... |
02:39.54 | asteriskmonkey | depends on the codec |
02:39.57 | outtolunc | internal? |
02:40.06 | Kizmet | as in |
02:40.08 | outtolunc | what cable lengths |
02:40.12 | outtolunc | etc etc etc |
02:40.16 | Kizmet | Call XXX XXX XXX for Support. |
02:40.24 | [hC] | anyone here use the open79xx xml directory for cisco phones? |
02:40.36 | samueltc | asteriskmonkey: using the same codec indeed.. |
02:40.41 | samueltc | g729 |
02:40.41 | Kizmet | We are localted in the BACK of the data centre right near the distrib board. |
02:40.59 | asteriskmonkey | then iax2 has less overhead than sip |
02:41.00 | asteriskmonkey | so there :D |
02:41.03 | samueltc | nice |
02:41.04 | justinu | actually, i've had PRIs from a bunch of different telcos |
02:41.08 | justinu | on a bunch of facilities |
02:41.11 | justinu | HDSL, DS3 |
02:41.14 | justinu | fiber |
02:41.29 | samueltc | justinu: any in europe? hehe |
02:41.31 | justinu | i've got 2 DS3s of PRI |
02:41.33 | justinu | nope |
02:41.41 | justinu | but they have always been in the city of LA |
02:41.47 | justinu | or suburbs of |
02:41.50 | mzo | anyone have an idea why i can't get fwd to connect via IAX? it's been busted for me since sunday and it's never worked and i can't make heads of tails of teh error. http://www.freeworlddialup.com/community/forum/viewtopic.php?t=3611 |
02:42.01 | Kizmet | outtolunc, The Asterisk Servers are up the top of the rack with 3m CAT6 cables going to the patch at the bottom of the rack then about 5 meters of cable going to the distrib board |
02:42.16 | outtolunc | [18:41] <justinu> actually, i've had PRIs from a bunch of different telcos |
02:42.21 | mzo | pics plz |
02:42.24 | *** join/#asterisk websae (i=icechat5@CPE-24-167-204-30.wi.res.rr.com) |
02:42.27 | mzo | hot picz of sexy rack |
02:42.29 | outtolunc | [18:37] <outtolunc> right, and you are in the same place you always been |
02:42.42 | justinu | well, sorta |
02:42.48 | katakefalos | hello room can someone help me? is it possible in extensions.conf if a caller dials a number NXXXXXXXXX to add instantly a 1 prefix and pass it on? somthing like (which is wrong of course): exten => _NXXXXXXXXX,1,Answer |
02:42.48 | katakefalos | exten => 1+_NXXXXXXXXX,2,Wait(1) |
02:42.48 | katakefalos | exten => 1+_NXXXXXXXXX,3,DeadAGI(a2billing.php|1) |
02:42.48 | katakefalos | exten => 1+_NXXXXXXXXX,4,Hangup |
02:42.51 | Kizmet | mzo, no pics working on it tho :) |
02:42.54 | outtolunc | either explain yourself better or stop while you are behind |
02:43.10 | justinu | not worth it to me :P |
02:43.13 | outtolunc | k |
02:43.25 | mzo | i will trade hot rack pics (sexy ciscos!) for help with fwd! |
02:43.44 | lunaphyte | any opinions on ipkall.com ? |
02:44.29 | Kizmet | lunaphyte, There incoming calls seem to be of poor quality sometimes and my number got deleted after 2 months (It was under relitivly high use) |
02:45.00 | mzo | i need to find something to call finland cheap ;) |
02:45.09 | lunaphyte | is it true that you can only talk to them through fwd or similar? |
02:45.24 | Kizmet | mzo, a VoIP prov in Finland perhapse :P |
02:45.43 | Kizmet | lunaphyte, Nope. You may forward the call to any SIP capable host. |
02:45.55 | websae | firestrm: are you there? |
02:46.29 | lunaphyte | ah - so i f i configure asterisk to accept a sip connection from them, i can point them towards my server when i sign up? |
02:46.48 | Kizmet | lunaphyte, yep |
02:47.05 | mzo | Kizmet, im trying to find one, to talk to there |
02:47.34 | Kizmet | mzo, Yes, Get one from there and call inside the country :) Mostly the cheapest solution :) |
02:47.50 | mzo | that's what im looking for, i can't find one iwth good english translation :) |
02:48.16 | Kizmet | mzo, heh |
02:49.00 | *** join/#asterisk iaxy (n=iaxy@modemcable236.55-131-66.mc.videotron.ca) |
02:49.37 | iaxy | hi everyone |
02:49.41 | outtolunc | i'm still wondering why kizmet wasn't including the distance to the interchange |
02:49.53 | iaxy | Has anyone succesfully used Citels handset gateway? |
02:49.53 | outtolunc | (50'-100') |
02:50.02 | Kizmet | outtolunc, dist to the interchange im unsure of. |
02:50.49 | outtolunc | once again... |
02:50.50 | Kizmet | outtolunc, Im one of the 'Asterisk' admins all the admining i do is with the dialplan -_- |
02:50.50 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
02:51.16 | outtolunc | i'm just a guy that thinks a dialplan is a script |
02:51.22 | outtolunc | silly me |
02:51.24 | jeebusroxors | Hey - Is IAX recomended over SIP when registering with a PSTN termination? |
02:51.54 | xachen | erm |
02:51.57 | xachen | I'd take SIP anyday |
02:52.09 | Kizmet | outtolunc, Heh all i do with the dial plan is change the way certain dest's route and update billing stuff. |
02:52.18 | jeebusroxors | xachen; i was reading that iax is better on bandwith and what not |
02:52.23 | xachen | yes |
02:52.26 | xachen | but IAX is also more insecure |
02:52.28 | Qwell | 355 |
02:52.28 | outtolunc | i think i get that, really <G> |
02:52.31 | xachen | and can be real buggy |
02:52.35 | jeebusroxors | hm |
02:52.37 | Telamon | jeebusroxors: If they support it, IAX is supposedly better. If you don't have to deal with NATS and firewalls though, there isn't much difference. |
02:52.50 | xachen | I'm just a SIP nerd though |
02:53.19 | jeebusroxors | oh, i defanetly dig SIP, im just concered with bandwith issues as im using my own machine as a server |
02:53.32 | iaxy | anyone use citels sip handset gateway, to connect nortel digital phones to asterisk? |
02:54.08 | asteriskmonkey | iaxy : yes |
02:54.23 | asteriskmonkey | have to setup 4 of them tommorow :D |
02:54.58 | russellb | xachen: IAX is more insecure? |
02:55.00 | iaxy | AH... I haven't got mine to send a single sip ack.... |
02:55.01 | russellb | did you make that up? |
02:55.13 | xachen | actually no :) |
02:55.18 | Telamon | Anyone else having problems getting GXP-2000's to download their config from the tftp server after upgrading to 1.0.2.x? Do you need to specify the TFTP server via DHCP now or something? |
02:55.20 | xachen | may I reprhase |
02:55.31 | russellb | of course :) |
02:55.33 | outtolunc | tis the season |
02:55.37 | jeebusroxors | heh |
02:55.43 | xachen | IAX is insecure but so is SIP |
02:55.47 | xachen | I'd just take SIP over IAX |
02:55.48 | iaxy | asteriskmonkey; can I msg you |
02:55.58 | xachen | the only thing I disagree about IAX is the whole user/pass bruteforce potentials |
02:56.19 | *** join/#asterisk tainted_ (n=identd@ppp-71-133-241-120.dsl.irvnca.pacbell.net) |
02:56.28 | tainted_ | how many 729 channels could a DSL line hold? |
02:56.32 | tainted_ | assuming 128k up |
02:56.46 | outtolunc | and no ids that looks for 'bruteforce' you agree with? |
02:56.48 | justinu | tainted.... |
02:56.57 | tainted_ | yea |
02:57.02 | outtolunc | meaning.. no nadda |
02:57.04 | justinu | probably about 7 |
02:57.05 | xachen | well you can give no ID and a password and the server will try to find the username for you |
02:57.14 | outtolunc | oh yeah bang the shit out of me.. go for it |
02:57.15 | Telamon | tainted_: Technically, 14. In reality, about 10. |
02:57.16 | justinu | it's pretty light |
02:57.16 | xachen | but thats easily fixd by adding a [guest] entry |
02:57.35 | tainted_ | wow that's a lot more than i expected |
02:57.48 | iaxy | asteriskmonkey;Do you have any docs on using citel GW with Asterisk? |
02:58.06 | outtolunc | sorry :( |
02:58.09 | Telamon | tainted_: g729 is about 8kBytes/sec per call. + UDP/IP overhead. |
02:58.34 | outtolunc | just that i see obvious stuff today as totally silly |
02:59.49 | *** join/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM) |
02:59.52 | *** part/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
03:00.04 | websae | firestrm are you out there? |
03:00.48 | [hC] | urg. |
03:00.49 | [hC] | anyone here use the open79xx xml directory for cisco phones? |
03:03.46 | outtolunc | so, if that seemed 'normal' to you, do NOT ask me a damn thing <G> |
03:03.54 | *** join/#asterisk JCC_ (n=user@207.41.92.131) |
03:06.13 | outtolunc | joy, i just caused world peace! <G> |
03:06.20 | outtolunc | hah |
03:08.05 | asteriskmonkey | iaxy: did you buy them from williams? |
03:08.25 | *** join/#asterisk xbmodder_lappy (i=nobody@unaffiliated/xbmodder) |
03:08.39 | xbmodder_lappy | Who is the most _reliable_ VoIP provider out therE? |
03:09.10 | tuxinator_linux | xbmodder_lappy: when you find out, let us know |
03:09.44 | glm2k | tuxinator_linux: rotfl |
03:09.46 | xbmodder_lappy | lol |
03:09.52 | outtolunc | lappy, 'most' of us know that when it comes to 'reliable' it depends on the 'backhoe' factor |
03:09.53 | asteriskmonkey | xbmodder_lappy: that would be me :) but i got echo issues lol |
03:10.05 | glm2k | outtolunc: aye, i second that |
03:10.59 | *** join/#asterisk wrench (n=signal@68-118-224-178.dhcp.oxfr.ma.charter.com) |
03:11.51 | wrench | hey all, whats a good incoming-only voip provider for small businesses - trying to set up a custom IVR solution |
03:12.05 | outtolunc | seriously, if you plot line issues to power issues, the power issues take the cake |
03:12.33 | outtolunc | so line issues boil down to 'dude with backhoe' |
03:12.48 | evilbuny | wrench: ipkall.com offer free inbound numbers in the US |
03:13.10 | voip470 | "backhoe factor"? |
03:13.24 | outtolunc | prefect example |
03:13.27 | outtolunc | er per |
03:13.43 | wrench | evilbuny > for commercial use? |
03:13.43 | outtolunc | 'reset please?' |
03:14.54 | evilbuny | wrench: they make money per minute on your call |
03:15.04 | evilbuny | fraction of a cent per minute |
03:16.51 | outtolunc | ... punt |
03:17.16 | asteriskmonkey | what the hell |
03:17.16 | asteriskmonkey | i called a number and its not in my zap show channels |
03:17.33 | asteriskmonkey | i got the mci test tone number |
03:17.35 | outtolunc | called = passed tense |
03:17.41 | asteriskmonkey | no im connecte |
03:17.46 | asteriskmonkey | connected currently |
03:17.51 | asteriskmonkey | its a test tone number |
03:17.55 | outtolunc | then say what is really happening |
03:18.07 | outtolunc | i'm not a f'n mind reader |
03:18.11 | asteriskmonkey | sos |
03:18.13 | wrench | hmm |
03:18.50 | outtolunc | which 'type' of channel is it |
03:18.55 | outtolunc | if it is ' |
03:19.00 | outtolunc | of a type' |
03:19.10 | tengulre | Hi,all! I m backing....! good morning every one! |
03:19.13 | outtolunc | then did you do '<the type> show channels' |
03:19.24 | outtolunc | or is that 'out there' |
03:19.54 | outtolunc | sorry, i didn't mean to swear with 'f'n' |
03:20.12 | asteriskmonkey | zap show channels |
03:20.15 | asteriskmonkey | shows nothing |
03:20.23 | outtolunc | is it a zap channel |
03:20.33 | asteriskmonkey | its goign out a zap channel |
03:20.42 | asteriskmonkey | sip=>asterisk=>zap/pri |
03:20.48 | outtolunc | and we would know this HOW |
03:21.20 | outtolunc | and you say 'it IS going out' |
03:21.34 | outtolunc | well if it is, then there isn't an issue |
03:21.47 | asteriskmonkey | except i cant see what channel its on |
03:21.53 | asteriskmonkey | gah |
03:22.05 | asteriskmonkey | im getting sleepy and not explaining myself well anymore |
03:22.05 | outtolunc | did you mean you say that 'you attempted xyz and it failed with THIS error message" |
03:22.17 | asteriskmonkey | no |
03:22.31 | asteriskmonkey | i mean its working currently connected listening to a tone over a zap channel |
03:22.40 | asteriskmonkey | just show zap channels is not showing me what line its using |
03:22.54 | outtolunc | so, you attempted "something", and there is no error message that you want to share with us |
03:23.10 | outtolunc | ok, ummm goodluck |
03:23.55 | outtolunc | it boiled down too |
03:23.57 | outtolunc | [19:17] <asteriskmonkey> no im connecte |
03:23.57 | outtolunc | [19:17] <asteriskmonkey> connected currently |
03:23.58 | outtolunc | [19:17] <asteriskmonkey> its a test tone number |
03:24.05 | asteriskmonkey | sos.. |
03:24.09 | lunaphyte | so when ipkall.com asks me what my sip phone number is, can i just invent that, provided it matches how my config is set up? |
03:24.12 | outtolunc | that is NOT any form of request |
03:24.18 | asteriskmonkey | i called a number using my sip phone which traveled out a zap channel |
03:24.37 | asteriskmonkey | the phone call is currently in progress and a zap show channels did not render which channel it was using |
03:24.42 | asteriskmonkey | there that makes more sense |
03:24.57 | outtolunc | yes it does |
03:25.19 | outtolunc | but it also would make me ask why you didn't do a sip show channels |
03:25.24 | outtolunc | hmmm |
03:25.36 | asteriskmonkey | sip show channels shows stuff but not the zap channel |
03:25.51 | *** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.138) |
03:25.54 | Kernel_core | hi all |
03:25.54 | outtolunc | obviously |
03:26.03 | asteriskmonkey | i need to know that zap channel to use ztmonitor ! |
03:26.04 | outtolunc | sip, zap, sip, zap |
03:26.08 | outtolunc | hmm |
03:26.15 | outtolunc | hello |
03:26.21 | asteriskmonkey | doh i still get * in my ztmonitor .. i guess that means i have to lower it still |
03:26.31 | Kernel_core | anybody successfully compiled H323 on Asterisk ?! |
03:26.32 | outtolunc | how about a 'show channels' |
03:26.37 | outtolunc | hmmmmmmm? |
03:27.27 | outtolunc | obviously he never thought of that one |
03:27.51 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-92-216.cybersurf.com) |
03:28.31 | asteriskmonkey | doh! |
03:28.32 | asteriskmonkey | :P |
03:28.36 | outtolunc | k |
03:28.47 | outtolunc | your welcome |
03:29.20 | asteriskmonkey | thanks |
03:31.30 | Goral | is there a working asterisk module for webmin? |
03:33.24 | outtolunc | google 'asterisk pbx webmin' |
03:34.10 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
03:34.14 | shmaltz | hi every1 |
03:34.36 | Goral | the only thin i have come across is people complaining that it doesn't work... i was just wondering if there was a person here that can confirm that it works.. |
03:35.12 | outtolunc | ah |
03:35.33 | *** join/#asterisk chetan (i=freetibe@cpe-24-193-188-21.nyc.res.rr.com) |
03:35.37 | outtolunc | well since i've never used it i'd be one the 'undecided' |
03:35.56 | *** join/#asterisk camonz (n=camonz@200.8.20.210) |
03:35.59 | camonz | evening :-> |
03:36.10 | outtolunc | which leads to the 'you are in a group of 'those that need/want' it' |
03:36.28 | outtolunc | which mind you, is a 'subset' |
03:36.36 | shmaltz | Goral, what works or doesn't? |
03:37.11 | Goral | webmin module for asterisk |
03:37.25 | outtolunc | i'm sure it 'did' at one rev, but may/maynot now... he just doesn't get that |
03:37.44 | shmaltz | Goral, I looked at it a while back, from all the ones around, I think 3rd lane has done the best job so far |
03:39.32 | shmaltz | http://news.yahoo.com/s/pcworld/20060221/tc_pcworld/124781;_ylt=AloXUQMSFUN4Ygm4SqoGyx_67rEF;_ylu=X3oDMTBjMHVqMTQ4BHNlYwN5bnN1YmNhdA-- |
03:39.44 | Goral | ty shmaltz i'm new to asterisk but i always used webmin to learn a process |
03:40.10 | *** join/#asterisk nurfe (n=rgff@h24-207-10-162.dlt.dccnet.com) |
03:40.10 | shmaltz | Goral, what are you trying to accomplish? |
03:40.22 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
03:40.29 | Goral | well.. |
03:41.26 | shmaltz | outtolunc, back to the ocean |
03:42.00 | outtolunc | oh, so someone that knew there 'were from the ocean' didn't head TOO the ocean |
03:42.23 | Goral | i want to put a box togethere that will allow me to use a sip subscription throught my house |
03:43.03 | shmaltz | Goral, then you should try Asterisk, it's that wonderfull program that can do that for you ;) |
03:43.14 | Goral | eg. i'm using a dvg1120 right now and it has two working outputs.. these allow me to call out from the box at the same time.. |
03:43.17 | outtolunc | he knows this tho |
03:43.25 | outtolunc | but you stepped up |
03:43.32 | outtolunc | you need to make sure his 'home' |
03:43.48 | outtolunc | <PROTECTED> |
03:44.42 | Goral | guess its like splitting my sip account all over the house |
03:44.56 | outtolunc | umm no, it isn't <G> |
03:45.23 | outtolunc | do you pay more than once |
03:45.39 | Goral | what on my sip account? |
03:45.46 | outtolunc | thats a no |
03:46.11 | outtolunc | so it 'IS ONE ACCOUNT' |
03:46.17 | Goral | like i said i'm sort of new |
03:46.19 | Goral | yes |
03:46.24 | Goral | this is one account |
03:46.29 | outtolunc | then why the bs? |
03:46.56 | Kizmet | outtolunc, I have a IAX2 Trunk into my ITSP :) I have a so called 'Unlimited Local-Loop' connection into them :) |
03:47.14 | outtolunc | i think they are high (or you are) |
03:47.25 | outtolunc | take your pick |
03:47.26 | Kizmet | outtolunc, lol |
03:47.29 | *** join/#asterisk websae (i=icechat5@CPE-24-167-204-30.wi.res.rr.com) |
03:47.53 | outtolunc | so, your question 'would have been' what? |
03:47.57 | Kizmet | outtolunc, I work for the ITSP :) |
03:48.27 | Goral | well the way i look at it is that if the dvg1120 can allow me to access my account on 2 seperate phones at one time calling two diferent locations why could i not do it with asterisk |
03:48.36 | Kizmet | Goral, There is no reason why it wont work. |
03:48.44 | outtolunc | i am just wondering what the hell type of person asks an 'open ended, not finished, question' kinda sorta |
03:49.12 | outtolunc | but hey, shit happens |
03:49.23 | outtolunc | kinda, sorta |
03:49.50 | Goral | now i have to learn how to set it all up |
03:49.56 | JCC_ | 'cause he's a little new at this |
03:50.07 | outtolunc | or did i miss some cool ass question that would have just made my day |
03:50.34 | outtolunc | jcc, which is? |
03:50.47 | outtolunc | or do i need to get on you also |
03:51.16 | Goral | outtolunc ty for your perspective its people like you that make me think |
03:51.33 | Qwell | hmm |
03:51.40 | Qwell | my sarcasm detector just exploded |
03:51.53 | outtolunc | goral, i am the type that 'will always help you' i don't ask much |
03:52.12 | outtolunc | but if you ask something silly, you get the same |
03:52.19 | outtolunc | hello |
03:52.19 | Goral | i know but i was serious that was far from sarcasm |
03:52.20 | websae | firestrm: you out there? |
03:52.27 | JCC_ | not sure what you're talking about... |
03:53.17 | outtolunc | JCC: it's ok, it's smooth, gooooo back to sleeeeeeep |
03:53.20 | JCC_ | I'm under the impression he's starting in the wrong place, maybe voip-info.org is his best bet |
03:53.39 | outtolunc | gee |
03:53.57 | outtolunc | that starting point never crossed ANY of our minds |
03:54.06 | outtolunc | ya think |
03:54.19 | JCC_ | it did'nt cross his though, apparently. |
03:54.27 | outtolunc | gee |
03:54.30 | Goral | JCC_ i always jump in head first... no better way that to deal with people that know it the iner workings |
03:54.35 | outtolunc | he gets it, i think |
03:54.54 | JCC_ | he does now |
03:54.57 | outtolunc | now when we place him in the same category, does he |
03:55.29 | outtolunc | you were just under the wire, i retract that last |
03:55.33 | Goral | hell i'm a noob give me a little slack |
03:55.44 | Goral | my linux is rusty too |
03:55.55 | outtolunc | goral, i've done nothing but help |
03:56.04 | outtolunc | i've not even been mean to you |
03:56.32 | JCC_ | check voip-info.org and you'll get 99% of your questions answered there, you'll need to do some reading, lots of it. |
03:57.11 | JCC_ | and get a decent book on linux, too, then. |
03:57.30 | Goral | outtolunc i know and i'm thankful |
03:57.45 | Qwell | Qwell's - Teach Yourself Linux in 24 Weeks, is a good Linux boox |
03:57.46 | Qwell | book |
03:58.20 | JCC_ | couldn't tell you,I haven't read one in 6 or 7 years |
03:58.34 | Qwell | I'm joking, of course |
03:58.54 | JCC_ | :) |
03:59.03 | JCC_ | I figured |
03:59.06 | Goral | i forget how simple it is.. micro$oft is so complex... as in its wording.. |
03:59.32 | Goral | well looks like i need a book |
04:00.04 | JCC_ | OReilly's asterisk book is decent |
04:01.07 | JCC_ | but voip-info is probably your best starting point, not here. |
04:01.33 | JCC_ | right, outtolunc? |
04:01.40 | JCC_ | :) |
04:03.31 | *** join/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net) |
04:04.44 | websae | if anyone needs some quality termination let me know, we have a great network :) and great prices and such, especially for fellow asterisk users :)! you can message me or sales@websae.com to inquire more, hope everyone is having a great night! |
04:05.02 | russellb | Please don't use this channel for commercial purposes. |
04:05.14 | websae | my apologies |
04:05.21 | websae | i dono't mean to advertise |
04:05.46 | websae | just saying dedicated staff out there that keep up a good network for asterisk termination if anyone needs it |
04:05.59 | Mavvie | has anybody here ever tried to contact support@digium.com ? |
04:06.11 | websae | yeah, they are pretty good about getting back to you |
04:06.16 | Mavvie | oh |
04:06.20 | Mavvie | funny :-) |
04:06.24 | websae | at least from my experience |
04:06.38 | Mavvie | did you call them or via email? |
04:06.40 | websae | i had to email them once or twice with a couple hardware issues |
04:06.41 | trixter | I get away with talking about my termination services cause they are free :P |
04:06.46 | trixter | dont even have to have an account |
04:06.52 | websae | that's awesome |
04:07.00 | websae | what type of termination do you have? |
04:07.05 | websae | u.s.--? |
04:08.27 | trixter | us and canada tollfree |
04:08.42 | websae | do you monintor calls? |
04:08.48 | trixter | http://www.trxtel.com/index.php?page=Tollfree_Termination |
04:09.03 | trixter | no that would be illegal to monitor calls without informing people, it is also a violation of the privacy policy |
04:09.26 | trixter | we are trying to work a deal for inbound dids free as well |
04:09.27 | websae | why is the service free? |
04:09.41 | trixter | to ensure that many people make calls |
04:09.51 | outtolunc | the service is not free, send the check too.... |
04:10.30 | mzo | trixter, is t that your service? |
04:10.41 | *** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
04:10.43 | trixter | yes |
04:11.07 | russellb | we used to do free tollfree on iaxtel ... one of these days I'm going to fix that server :) |
04:11.12 | outtolunc | ah |
04:11.21 | trixter | heh |
04:11.28 | outtolunc | scroll back heaven... |
04:11.31 | *** join/#asterisk rustyb (n=rustyb@68-235-135-252.atlsfl.adelphia.net) |
04:11.36 | trixter | well if people push enough minutes I pay them, totally backwards of the way phone service normally works |
04:12.20 | mzo | trixter, is that your company? |
04:12.25 | [hC] | using n(label) in a dial plan, i thought it was only supposed to execute it if it was referenced directly with (label) |
04:12.33 | [hC] | I have this, |
04:12.34 | [hC] | exten => s,n,GotoIf($["${CALLERIDNUM}" = ""]?SetUnknownCID) |
04:12.34 | [hC] | exten => s,n(SetUnknownCID),Set(CALLERID(num)=Unknown) |
04:12.34 | [hC] | exten => s,n,Goto(MainMenu,s,1) |
04:12.44 | *** part/#asterisk chetan (i=freetibe@cpe-24-193-188-21.nyc.res.rr.com) |
04:12.46 | [hC] | and "SetUnknownCID" always gets executed, regardless of the results of the gotoif. |
04:13.00 | russellb | yes, you need to add the rest of the GotoIf |
04:13.11 | russellb | that tells it to skip it if the result is 0 |
04:13.26 | russellb | the label is just a reference, but that doesn't exclude it from being executed ... |
04:13.30 | [hC] | Oh.. okay. |
04:13.45 | [hC] | I didnt know if it executed it anyways... |
04:13.53 | russellb | yup |
04:14.43 | *** join/#asterisk Jizzbug (n=derekm@63-254-64-44.ip.mcleodusa.net) |
04:17.07 | trixter | mzo: yes it is |
04:17.30 | outtolunc | yeah, whatever he said.. just YEAH |
04:17.42 | outtolunc | something |
04:18.11 | outtolunc | otherwise, this channel gets fairly boring |
04:18.44 | outtolunc | or doesn't that matter here?\ |
04:18.47 | Mavvie | -- Launched AGI Script /var/lib/asterisk/agi-bin/check-callerid.pl |
04:18.50 | mzo | trixter, i can't takl atm , but i had to ask you questsion about setting up for it. |
04:18.52 | Mavvie | ls: /var/lib/asterisk/agi-bin/check-callerid.pl: No such file or directory |
04:18.55 | Mavvie | who do I have to believe? |
04:19.19 | outtolunc | so 'check-callerid' |
04:19.33 | outtolunc | is the topic |
04:19.53 | outtolunc | i believe shit happens for a reason |
04:20.06 | russellb | iaxtel is now running the latest code from the 1.2 branch ... maybe that will make it happy |
04:20.13 | Mavvie | it's more that it doesn't throw an error. |
04:21.23 | mzo | trixter, i tried to set that up before, but it wouldn't take my registration, and i think the iax instructions might be a little simple? |
04:21.26 | outtolunc | umm if you can't see that as an error, you just might have issues <G> |
04:21.56 | Mavvie | outtolunc: it wasn't asterisk which did do that ls output |
04:22.10 | outtolunc | regardless |
04:22.23 | outtolunc | there is a spoon (err error) |
04:22.35 | outtolunc | somewhere someplace |
04:22.41 | Mavvie | could be a 1.2.4 vs HEAD issue since it shows up with an error in HEAD. |
04:22.47 | *** join/#asterisk bkw_ (n=bkw_@ip-207-145-170-175.lax.megapath.net) |
04:22.48 | trixter | there is no registration |
04:22.58 | mzo | how do i set it up then :P |
04:23.04 | trixter | cut and paste |
04:23.09 | outtolunc | but all we get is 'life is normal/usual' i have no clue that anything is wrong |
04:23.17 | trixter | you just dial no accounts, no registration, no tracking nothing |
04:23.19 | outtolunc | etc etc etc |
04:23.26 | mzo | oh, hmm, it bombed, i think i did it wrong then ;) |
04:23.56 | trixter | in whatever context you have for outbound dialing, make sure that none of the patterns match asterisk has issues that way, but put those dialing instructions in and you should be ready instantly |
04:24.14 | outtolunc | good, then we can add you to the 'he agrees that shit happens party' |
04:25.02 | outtolunc | personally, i could give a rats ass, but since others here care.. i should |
04:25.10 | *** join/#asterisk xtrvd (n=j@d209-121-36-44.bchsia.telus.net) |
04:25.21 | *** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar) |
04:26.06 | outtolunc | obviously not that much |
04:26.47 | rustyb | russellb Woot! I'm registered w/ IAXTEL again |
04:26.55 | russellb | nice! |
04:27.05 | outtolunc | gee wally, thats special |
04:27.09 | rustyb | its been like a year |
04:27.55 | outtolunc | i once again see why it's fairly hard to take anyone serious here |
04:28.17 | rustyb | russellB are you aware of any zap <=> zap 1 way audio problems w/ ver 1.2.4 ? |
04:28.29 | russellb | no, i have not heard anything |
04:28.52 | mzo | what is iaxtel? |
04:29.00 | outtolunc | gee why would there be any zap to zap audio issues |
04:29.33 | niZon | has anyone here ordered from voxilla? |
04:29.50 | rustyb | i dont know sip <=> zap and iax <=> zap are ok. |
04:30.34 | rustyb | software or driver/ hardware issues likely.. i guess it's just my install |
04:31.04 | niZon | hmm their canadian toll free is down |
04:31.38 | outtolunc | strange, i'm sure there were some slinear issues that floated into my email |
04:32.01 | outtolunc | but gee how can they be the same |
04:32.30 | Mavvie | execv(script, argv); |
04:32.30 | Mavvie | fprintf(stdout, "verbose \"Failed to execute '%s': %s\" 2\n", script, strerror(errno)); |
04:32.43 | Mavvie | that explains why I see it on HEAD and not on the other one. |
04:33.02 | outtolunc | the reason for the double "\" crap is? |
04:33.03 | Mavvie | let's see how stat works again |
04:33.42 | mzo | trixter, if you'd help me set it up, i'll do it, i got confused when i set it up for outbound dialing |
04:34.18 | *** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net) |
04:34.35 | Mavvie | because it's supposed to be one argument. |
04:36.00 | trixter | um ... its like anything else in the context you are in |
04:36.05 | russellb | rustyb: do you have iaxtel set up where you could make a test call? |
04:36.06 | trixter | it quite literally is cut and paste |
04:36.26 | outtolunc | gee |
04:36.29 | Kizmet | trixter, wasnt here cuz im using AEL :) but eyh it sould be cut n paste |
04:36.30 | mzo | lemme go poke around, i'm using the amp, so i probably screwed something up :P |
04:36.32 | outtolunc | but and paste |
04:36.35 | outtolunc | er cut |
04:36.43 | trixter | ok I will add ael stuff tonight :P |
04:36.43 | outtolunc | hmm |
04:36.48 | mzo | what's AEL? |
04:37.01 | trixter | a different dialplan language for asterisk |
04:37.08 | Kizmet | mzo, nothing you need to worry about at this stage young padawan. |
04:37.22 | xbmodder_lappy | mzo, You are a young padawan |
04:37.23 | mzo | i have a long time until i reach padawan status :P |
04:37.33 | mzo | I'm still trying to get xp to level up to noob. |
04:37.41 | Kizmet | lol |
04:37.43 | outtolunc | i'm still trying to see where the NON-padawan |
04:37.49 | outtolunc | 's get into this |
04:37.57 | xbmodder_lappy | We do asterisk consulting, would you like to be taken under our wing. We can teach you. |
04:38.07 | Kizmet | outtolunc, no idea *sigh* |
04:38.08 | mzo | they get an invite to the sekret invite #jedi-asterisk |
04:38.12 | outtolunc | really? |
04:38.14 | mzo | xbmodder_lappy, teach me, my master. :P |
04:38.17 | outtolunc | gee |
04:38.27 | outtolunc | where do i sign up? |
04:38.33 | xbmodder_lappy | Remember I am only here to show you the door, you're the one who has to walk through it |
04:38.34 | russellb | i know someone wants to make an iaxtel test call :) |
04:38.47 | mzo | russellb, show me how to configure it :P |
04:39.01 | russellb | mzo: ha ... iaxtel.com |
04:39.13 | *** join/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net) |
04:39.18 | Mavvie | heh... C should have an -x operator :-) |
04:39.37 | outtolunc | come on |
04:39.42 | outtolunc | there has to be more than this |
04:40.18 | mzo | take the red pill, the one labelled 'iax' |
04:40.23 | Mavvie | or can I safely assume that asterisk is always ran as root? |
04:41.08 | outtolunc | umm NO |
04:41.27 | outtolunc | search the wiki for 'non-root' |
04:41.40 | *** join/#asterisk riddlebox (n=victoria@24-171-11-166.dhcp.stls.mo.charter.com) |
04:41.44 | outtolunc | what the F are you teaching these guys |
04:41.58 | xbmodder_lappy | lol |
04:42.21 | outtolunc | what do i care |
04:42.24 | outtolunc | go for it |
04:42.33 | outtolunc | have a wonderful DAY |
04:42.41 | outtolunc | party on |
04:42.51 | Mavvie | oh, access() is a good way to overcome this problem. |
04:43.10 | outtolunc | shoot mavvie in the head |
04:43.18 | outtolunc | for bs f'n crap |
04:43.28 | Mavvie | if that's the best you can come up, keep practising. |
04:43.54 | outtolunc | gee wally only another 30 years i might be able to deal with bobo |
04:44.05 | outtolunc | please |
04:44.10 | outtolunc | step up |
04:44.14 | outtolunc | take a shot |
04:44.21 | outtolunc | if you actually have one |
04:44.34 | outtolunc | hmmm |
04:44.38 | outtolunc | i'm waiting |
04:44.42 | outtolunc | still waiting |
04:44.46 | outtolunc | yet still |
04:44.58 | outtolunc | bit slow aren't ya |
04:45.27 | outtolunc | yet still, .... |
04:45.36 | *** join/#asterisk DaGeek215 (n=DaGeek21@c-71-226-252-76.hsd1.pa.comcast.net) |
04:46.51 | outtolunc | strange how all i get is 'is that the best you got' kind of crap.. after 25+ years of this crap.. thats all you come up with |
04:47.30 | outtolunc | anyone want to debate SIT tones durations? |
04:47.56 | outtolunc | aww |
04:48.29 | outtolunc | i hope someone is actually recording this to show i *was* trying to help |
04:49.05 | *** join/#asterisk Winkie (n=urmom@cpc2-stre1-6-0-cust119.bagu.cable.ntl.com) |
04:49.06 | mzo | i'm trying to sign up for that website, but i haven't gotten my password ;) |
04:52.16 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
04:52.56 | Mavvie | http://bugs.digium.com/view.php?id=6565 <- fixed! |
04:53.13 | mzo | russellb, are you the admin for that site? Send me my password :P |
04:53.46 | brookshire | mzo: which site? |
04:54.26 | russellb | mzo: ha, what username did you use |
04:54.31 | russellb | mzo: iaxtel |
04:54.33 | russellb | err |
04:54.35 | russellb | brookshire: |
04:54.48 | brookshire | hehe.. |
04:54.54 | mzo | yes iaxy |
04:54.56 | mzo | er, iaxtel |
04:54.59 | brookshire | someone seriously needs to fix iaxtel |
04:55.13 | russellb | mzo: what's your username? |
04:55.19 | mzo | mzo :P |
04:55.20 | brookshire | russellb: fix it now! |
04:55.22 | mzo | that's what i signed up as |
04:55.32 | russellb | it's not in the db :( |
04:55.40 | mzo | bleh lemme check |
04:55.42 | russellb | or the iaxtel db anyway |
04:55.49 | russellb | that might happen after you confirm it |
04:55.55 | russellb | and i have no idea how that part works |
04:55.58 | mzo | i didn't get the confirm email =( |
04:56.03 | russellb | yeahhhhh ... |
04:56.07 | russellb | i have no clue. |
04:56.09 | russellb | brookshire: !!!!!!!! |
04:56.10 | russellb | FIX IT |
04:56.11 | russellb | NOW |
04:56.20 | *** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net) |
04:56.26 | brookshire | maybe we could outsource it ;) |
04:56.29 | russellb | lol |
04:56.41 | brookshire | lol.. make it a topic |
04:56.48 | russellb | i'll fix it ... |
04:56.50 | russellb | ... later |
04:56.52 | brookshire | /msg brookshire if you want to fix iaxtel |
04:56.54 | [hC] | anyone here using the open79xxdir xml directory? |
04:58.16 | outtolunc | .. /or outsource everything and oh... whatever |
04:58.24 | outtolunc | gee |
04:58.32 | mzo | . |
04:58.43 | mzo | plz fix, i want to test :) |
04:59.40 | outtolunc | fix what? i've not seen a damn thing that looks like anything close to helpful |
05:00.19 | outtolunc | 'this looks wrong' |
05:00.33 | outtolunc | ok, yeah, when you say it that way |
05:00.43 | outtolunc | 'this looks wrong' <G> |
05:01.12 | outtolunc | anyone getting this? |
05:01.27 | outtolunc | or will the next person do the same damn thing |
05:03.05 | outtolunc | i'm sorry, but if you can't define it, i surely can't trace it down |
05:03.25 | outtolunc | )tjats |
05:03.33 | outtolunc | ekk sorry |
05:04.02 | *** join/#asterisk redondos (i=redondos@12-207-132-99.client.mchsi.com) |
05:04.32 | outtolunc | but please, anytime you want to state the 'issue' and the 'version' and the 'circumstances' well who am i to be the DICK |
05:04.55 | outtolunc | god forbid |
05:05.07 | outtolunc | anyone? |
05:07.42 | *** join/#asterisk jyukes (n=jameshot@pool-71-244-78-223.atc.east.verizon.net) |
05:07.47 | outtolunc | so you would rather i said: plz fix, i want to test :) |
05:08.02 | outtolunc | hmmm |
05:09.26 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
05:11.11 | katakefalos | does anyone know how to write dialplan? i want when a caller calls a areacode+localnumber to add a the prefix 1 and than continue with it somthing like: exten => _NXXXXXXXXX,1,Answer |
05:11.11 | katakefalos | exten => _NXXXXXXXXX,2,Wait(1) |
05:11.11 | katakefalos | exten => _NXXXXXXXXX,3,Prefix,1 |
05:11.12 | katakefalos | exten => _1NXXXXXXXXX,4,DeadAGI(a2billing.php|1) |
05:11.12 | katakefalos | exten => _1NXXXXXXXXX,5,Hangup |
05:11.19 | katakefalos | but it does not work |
05:12.41 | evilbuny | exten => _NXXNXXXXXX,1,Goto(1${EXTEN},1) |
05:12.48 | evilbuny | exten => _1NXXNXXXXXX,1,.... |
05:12.55 | *** join/#asterisk tengulre (n=tengulre@221.11.5.180) |
05:13.08 | Mavvie | Prefix(1) maybe? |
05:13.11 | outtolunc | first off |
05:13.24 | outtolunc | you need to keep 'this and that' together |
05:13.50 | evilbuny | but i might want my thises and thats seperated in the wash :) |
05:13.53 | outtolunc | meaning.. _NXX = this, and _1NXX = that |
05:13.57 | Mavvie | katakefalos: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Prefix |
05:14.07 | Mavvie | "If you switch into an extension which has no first step, Asterisk will treat it as though the user dialed an invalid extension." |
05:14.32 | outtolunc | so, JUMPING from this,3 to that,4 is not right |
05:14.34 | Mavvie | priority 4 is not a first step. |
05:14.58 | outtolunc | and if you can't see that, clean your glasses |
05:15.39 | Kizmet | Ok here is a question for you, How would i (Im using early dial on all my phones) make it as when someone dials an incorrect number they get passed to a context that tells them that the number that they have dialed is incorrect and for them to please try again later. |
05:16.09 | outtolunc | easy |
05:16.20 | outtolunc | listen closely |
05:16.37 | Kizmet | Ok NP |
05:16.47 | outtolunc | 'asterisk defaults to the [default] context.. hint f'n hint |
05:17.22 | mzo | brb i gotta reboot =( |
05:17.26 | outtolunc | meaning, anyone silly enough to use the [default] context as 'normal procedure is HIGH |
05:17.27 | outtolunc | ' |
05:17.37 | outtolunc | sorry<G> |
05:17.57 | outtolunc | OUT OF THEIR EVERLOVIN F"N MINDS |
05:18.06 | outtolunc | ok? |
05:18.32 | outtolunc | is that is "ANY WAY" unclear? |
05:19.04 | rustyb | russellb yes i can make an iaxtel test call |
05:19.08 | outtolunc | i personally HOPE not |
05:19.09 | evilbuny | outtolunc: sure it is, to people that don't do english :) |
05:19.10 | rustyb | whats your number? |
05:19.15 | katakefalos | outololunc> am new to this and try to unterstand |
05:19.34 | Kizmet | outtolunc, SIP/2.0 404 Not Found |
05:19.35 | russellb | rustyb: well, i ended up just calling myself :) |
05:19.41 | outtolunc | DO NOT EVER USE [DEFAULT} FOR ANY REASON |
05:19.48 | outtolunc | er } |
05:19.54 | outtolunc | er whatever |
05:20.04 | rustyb | mzo hello |
05:20.23 | rustyb | ok i did the same |
05:20.23 | *** join/#asterisk mzo (i=user@ool-435193b3.dyn.optonline.net) |
05:20.37 | rustyb | one number workes & the other doesn't |
05:20.38 | outtolunc | so when they ask you 'knowledgeable people that sit here daily' in the future, you will kindly pass that on, RIGHT |
05:21.04 | outtolunc | or will you be the non-responding asses you have shown yourselves to be |
05:21.11 | outtolunc | which is it? |
05:21.24 | evilbuny | can we be fence sitters instead? ) |
05:21.25 | evilbuny | :) |
05:21.38 | outtolunc | if you are, do so |
05:21.42 | outtolunc | and STFU |
05:21.45 | evilbuny | lol |
05:21.58 | Nivex | outtolunc: referring to the people who have the potential to help you as "asses" does not strenghten your posititon. |
05:22.13 | outtolunc | like i give a flying fuck |
05:22.22 | outtolunc | when have ANY of you helped me |
05:22.23 | outtolunc | ever |
05:22.24 | Nivex | clearly, you don't. |
05:22.32 | outtolunc | ever |
05:22.40 | Nivex | Please take your harsh language elsewhere. |
05:22.42 | evilbuny | outtolunc: what did you need help with? |
05:22.43 | outtolunc | i didn't studder did i |
05:22.45 | outtolunc | ever |
05:22.48 | russellb | outtolunc: have you filed bug reports? |
05:22.52 | outtolunc | haha |
05:22.53 | russellb | if you did, I'm sure I've worked on them in some way. |
05:22.57 | russellb | so there. |
05:23.23 | Kizmet | outtolunc, Ok let me make myself a little clearer. The context people are hitting when dialing out is : ael-outgoing i want to be able to redirect invalid calls to a different extention that being : ael-error inside that context i have the appropriate entrys to make it playback the default sound file pbx-invalid and loop. However i cannot get incorrect calls to hit that context wether using the 'i' ext or not, Have you guys/girls got any idea |
05:23.24 | Kizmet | <PROTECTED> |
05:23.27 | outtolunc | please if you think i'm without bugs, do a search |
05:23.35 | outtolunc | PLEASE |
05:23.42 | russellb | i know you have |
05:23.51 | outtolunc | yet you still pulled that crap |
05:23.55 | outtolunc | why? |
05:24.05 | outtolunc | more f'n BS |
05:24.06 | Mavvie | outtolunc: they have stuff against bugs at chemists. |
05:24.10 | russellb | because you said "when have ANY of you helped me" |
05:24.14 | Nivex | outtolunc: switch to decaf man. |
05:24.17 | rustyb | russellb the 2nd iaxtel no halts my * |
05:24.20 | outtolunc | when have YOU |
05:24.28 | outtolunc | russelb helped me? |
05:24.33 | outtolunc | please |
05:24.41 | outtolunc | when? |
05:24.58 | outtolunc | more f |
05:25.01 | outtolunc | 'n bs |
05:25.16 | russellb | dude, chill out :) |
05:25.21 | outtolunc | why |
05:25.27 | outtolunc | you said you helped me |
05:25.31 | outtolunc | so say so |
05:25.33 | Mavvie | mostly because you're making a fool out of yourself. |
05:25.39 | outtolunc | excuse me |
05:25.46 | mzo | russellb, please fix iaxtel :P |
05:25.46 | Nivex | outtolunc: You are excused. You may leave. |
05:25.59 | xtrvd | Kizmet: exten => i,1,Goto(ael-error,s,1) ? |
05:26.02 | mzo | ooh i'm missing drama! |
05:26.35 | Kizmet | xtrvd, as i have said i have tried using 'i' and it doesnt work. |
05:26.53 | katakefalos | thank you evilbunny! i did exten => _NXXXXXXXXX,1,Goto(1${EXTEN},1) |
05:26.53 | katakefalos | exten => _1NXXXXXXXXX,1,Answer |
05:26.53 | katakefalos | exten => _1NXXXXXXXXX,2,Wait(1) |
05:26.53 | katakefalos | exten => _1NXXXXXXXXX,3,DeadAGI(a2billing.php|1) |
05:26.53 | katakefalos | exten => _1NXXXXXXXXX,4,Hangup |
05:26.54 | russellb | http://bugs.digium.com/view.php?id=4768 |
05:26.58 | outtolunc | yeah, i'm silly, i'm a butthead, i've actually provided something back to asterisk... and yet, i'm getting flac because you buttheads can"t back up 'you own statements' <G> |
05:26.59 | russellb | there's a patch of yours that I merged |
05:27.04 | katakefalos | and it worked! |
05:27.09 | russellb | :-p |
05:27.22 | outtolunc | if you haven't realised... FU |
05:27.24 | xtrvd | Kizmet: So is the problem using the 'i', or is the problem passing to the new context? Can you isolate it a bit further? |
05:27.26 | Nivex | outtolunc: You're getting flack because you're acting like you're hyped up on speed. |
05:27.32 | outtolunc | have a nice day |
05:27.36 | mzo | hey, who has speed, and isn't sharing, plzkthx. |
05:27.58 | outtolunc | why is it i've not seen ANY of these nick's EVER |
05:28.17 | outtolunc | not in the 3+ years i've been on this project |
05:28.20 | outtolunc | hmmm |
05:28.33 | outtolunc | anyone |
05:28.44 | russellb | well, i usually go by drumkilla. |
05:28.44 | outtolunc | please show me some irc logs with you before me |
05:28.50 | outtolunc | please |
05:28.55 | Mavvie | three years on this project and only 6 little fixes? |
05:28.56 | mzo | i went to tell my friend that i'm using asterisk and he just gave me a rant that went along the lines of 'there's no hope for asterisk, ever, it's broken' =( some people are mean. |
05:29.08 | outtolunc | haha |
05:29.11 | outtolunc | ask mark |
05:29.27 | outtolunc | maybe, just MAYBE i was here BEFORE THE F |
05:29.28 | evilbuny | mzo: it's only mostly broken :) |
05:29.39 | outtolunc | "n bug tracker DUMBSHIT |
05:29.52 | mzo | haha, i just laughed when he told me, i guess i'm used to it :P |
05:30.02 | outtolunc | but excuse me, that's too HARSH |
05:30.18 | outtolunc | or is it |
05:30.21 | Mavvie | That's all I can get your history from. (you asked us to look at that ourselves) |
05:30.25 | outtolunc | i think not |
05:30.28 | Kizmet | xtrvd, The 'i' exten does not pass it to the other extention, I am using the SIP 484 response eg. Early Dial for the phones so that they simmulate a regular phone. |
05:30.42 | brookshire | mzo: he probably couldn't get it to compile |
05:30.48 | outtolunc | <PROTECTED> |
05:31.03 | outtolunc | damn, better watch out |
05:31.08 | mzo | brookshire i dunno. I kinda laughed a bit |
05:31.16 | mzo | ~amp |
05:31.17 | jbot | hmm... amp is NOT supported here! people using it should join #amportal |
05:31.21 | outtolunc | haha |
05:31.47 | xtrvd | Kizmet: But does the 'i' extension work at all? If you want to, perhaps loop your current context? |
05:31.49 | outtolunc | did you happen to notice the bug number on that first one? |
05:32.04 | outtolunc | 195 iirc |
05:32.22 | *** join/#asterisk freat (n=freat@h-72-244-84-43.chcgilgm.covad.net) |
05:32.23 | Nivex | I really hate it when people think that longevity gives them a license to assholery. |
05:32.25 | outtolunc | back then we didn't have to play the games we do now |
05:32.30 | Kizmet | xtrvd, Well i guess i could try it a little more than i have however a goto does not work. |
05:32.48 | freat | hello... anyone know how I could call an agi script when an agent answers a call from the queue? |
05:32.51 | outtolunc | no, just doing this shit for 25+ years |
05:33.05 | outtolunc | and listening to the same ol lame ass crap |
05:33.12 | outtolunc | so yeah |
05:33.16 | outtolunc | i'm a butthead |
05:33.25 | outtolunc | and you are 'lame ass crap' |
05:33.28 | Nivex | outtolunc: Admitting you have a problem is the first step toward recovery. |
05:33.44 | justinu | lol |
05:33.47 | katakefalos | LOL |
05:34.00 | mzo | stop fighing and help me fix fwd : |
05:34.05 | *** join/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net) |
05:34.13 | outtolunc | fwd is broken<G> |
05:34.14 | justinu | this channel is usually relatively light on the flame wars |
05:34.14 | evilbuny | Nivex: lol |
05:34.21 | outtolunc | omg life is failing |
05:34.33 | outtolunc | please help this man |
05:34.39 | justinu | why bother? |
05:34.46 | Nivex | I am beyond help. |
05:34.47 | outtolunc | exactly |
05:34.48 | Nivex | :-D |
05:34.56 | justinu | people are generally going to fuck it up one way or another |
05:34.59 | outtolunc | bs is bs |
05:35.07 | justinu | they'll find someone to help them do it wrong |
05:35.18 | outtolunc | so if you want my help, ask a REAL question |
05:35.27 | *** join/#asterisk Psykick (n=anon@203.167.226.250) |
05:35.29 | xtrvd | Kizmet: At the moment, I can't seem to think of how/why the goto doesn't work, sorry. |
05:35.29 | Nivex | outtolunc: What is the airspeed velocity of a swallow? |
05:35.31 | outtolunc | thats all i ask |
05:35.33 | mzo | outtolunc, it is? |
05:35.34 | evilbuny | outtolunc: what's the meaning of life, the universe and everything? |
05:35.34 | Psykick | hi all |
05:35.36 | Mavvie | outtolunc: how long will you continue with this? |
05:35.48 | justinu | is there a way I can use a sunset T1 to help diagnose/tune my tx/rxgains? |
05:35.56 | outtolunc | about 1mp[h more than you, because i'd have stuffed your silly ass in first |
05:36.06 | justinu | do I need a milliwat test number at the CO i'm terminating with? |
05:36.14 | justinu | or can I just set up one over the PSTN can call it? |
05:36.27 | outtolunc | if you don't 'think so' then maybe you need to talk to others that know me |
05:36.46 | outtolunc | just show up |
05:36.48 | Nivex | outtolunc: tsk tsk... I expected more than the same old vitriolic tripe from you |
05:36.56 | outtolunc | why |
05:37.00 | justinu | outtolunc: i could use your help, actually |
05:37.01 | outtolunc | i'm a marine |
05:37.11 | outtolunc | we don't play f'n games |
05:37.18 | mzo | it's the doom guy! |
05:37.21 | justinu | lol |
05:37.23 | russellb | mzo: lol |
05:37.23 | Nivex | I know a marine. He's not an ass like you. |
05:37.25 | Psykick | justinu: if I could help you I would |
05:37.35 | mzo | dude, do that thing with the eyebrows? |
05:37.40 | justinu | Psykick: thanks pal ;) |
05:37.55 | Psykick | mzo: he probably can't cos they're blocking his view |
05:38.03 | mzo | asterisk works mostly for me, but im not responsibel for it at work, they had a firm they use for work to deal with it, and they pay a pretty penny for it, but they won't teach me nothing. :P |
05:38.15 | Nivex | outtolunc has just earned the honor of being the third person ever placed on my /ignore list |
05:38.29 | outtolunc | oh and because your'friend' that isn't an add but a marine.. (probably from today) .... you can't believe that me, a marine of yesterday... would rather rip your f'n head off ... |
05:38.30 | outtolunc | hmm |
05:38.32 | Nivex | so, where were we? |
05:38.43 | justinu | lol |
05:38.52 | mzo | fwd not accepting my iax registration? :) help! |
05:38.56 | outtolunc | please |
05:39.05 | outtolunc | spring von |
05:39.08 | outtolunc | san jose |
05:39.10 | Nivex | mzo: That's been going on for quite awhile now. It made me sad. |
05:39.12 | Psykick | mzo: there's plenty of info on the net about registering with FWD |
05:39.17 | mzo | is it me, or their side? |
05:39.34 | mzo | Psykick, i already did everythign, but apparently it's not working atm, and i can't find out if that's true |
05:39.37 | outtolunc | show up or shut up, i'm so f'n tired of this weenie shit |
05:39.38 | Nivex | mzo: It's them. I was registered to them fine for months, then it just all of a sudden died. |
05:39.52 | Psykick | mzo: I don't know how old it is but the AAH handbook has that info (if you haven't tried that already) |
05:39.55 | mzo | so just be patient? |
05:39.58 | Nivex | interestingly I am registered to them now |
05:40.12 | mzo | Psykick, i followed it all verbatim. saysa n error code 29, and no one knows what it means |
05:40.13 | Psykick | Nivex: ain't that always the way |
05:40.31 | Psykick | mzo: which end? |
05:40.37 | mzo | lemme log |
05:40.41 | outtolunc | it's strange, no other channel i'm in thinks it's bs |
05:40.45 | outtolunc | only this one |
05:40.54 | mzo | join #linux and say linux sucks. :P |
05:41.14 | Psykick | I don't know if I dare ... but what is bs outtolunc? |
05:41.17 | mzo | actually when someone finds your body in a few hundred years... we'll be sure to ask. :P |
05:41.51 | outtolunc | the bs being how noone thinks i'm a marine that is more than willing to 'rip thier head off' |
05:42.06 | outtolunc | just ask file |
05:42.25 | outtolunc | even tho he's only been here 2 ISH years |
05:42.39 | Psykick | I wouldn't say that's bs .... I just know that you feel that you need to prove yourself to everyone and that's the best way you know how |
05:42.40 | justinu | i'm willing to pay someone to help me with this pri echo issue |
05:43.03 | outtolunc | oh yes, me, poor pitiful me.. why oh why |
05:43.19 | Psykick | outtolunc: you just don't want to admit it |
05:43.21 | mzo | nah |
05:43.25 | outtolunc | oh gee wally because it's f'n fun to kick the crap out of weenie |
05:43.26 | outtolunc | s |
05:43.31 | outtolunc | hmm |
05:43.33 | justinu | no takers? |
05:43.35 | Psykick | justinu: if its an echo issue it's quite likely that its your telco |
05:43.38 | outtolunc | shall i say more |
05:43.42 | evilbuny | Psykick: i thought most people needing to prove themselves got penis enhancments :) |
05:43.45 | mzo | http://pastebin.com/566298 |
05:43.48 | justinu | it's SBC, i'll pay someone to work with them to fix it. |
05:44.04 | Psykick | outtolunc: to be honest .... got no time for someone like you |
05:44.18 | justinu | if you're confident you can solve it, i'm interested |
05:44.19 | outtolunc | yet, instead of asking 'actual' questions, this is now a 'otl is a big meany' |
05:44.30 | mzo | that's the fwd iax debug log, i posted it on their forum but no answer yet |
05:44.36 | justinu | i asked some actual questions... was hoping for your insight... |
05:44.37 | outtolunc | so what the fuck does it matter |
05:44.49 | rustyb | justinu which echo canceler did u set when compiling zaptel? |
05:44.52 | Psykick | outtolunc: there are better things to be done than being obliging to anyone that has the inclination of ' ripping someones head off ' |
05:45.00 | justinu | rustyb: running mg2 right now |
05:45.07 | justinu | zaptel-1.2.1 |
05:45.10 | Psykick | and I can tell U ... that I'd be more than willing to rip yours off for you |
05:45.14 | Psykick | just say the word |
05:45.20 | justinu | lol |
05:45.31 | Psykick | hell I'll even make a nice stew from your remains |
05:45.36 | justinu | lol |
05:45.38 | outtolunc | oh well growing up on a farm in MN i was 'reglected' as a person so the use of arms became everyday |
05:45.43 | justinu | jeffry dahmer style |
05:45.57 | Psykick | not quite .... NZ Kiwi style |
05:45.57 | outtolunc | so please, tell me i'm strange<G> |
05:46.05 | evilbuny | there we go, penis enhancements! |
05:46.10 | justinu | Psykick: lol, you know who jeffry dahmer is? |
05:46.32 | outtolunc | oh gee, i squished a bug |
05:46.39 | outtolunc | drat |
05:46.41 | Psykick | outtolunc: I'll dig a hole just for you and steam your remains until your flesh is coming off the bones and suck every last morsel off |
05:46.48 | outtolunc | haha |
05:46.48 | *** join/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
05:46.50 | justinu | haha |
05:46.55 | outtolunc | you have no f'n clue |
05:47.08 | xtrvd | Nivex: Sorry I took so long to ask this, but: Laiden or unlaiden? |
05:47.16 | Psykick | believe me ... it's in my culture .. and my nature |
05:47.18 | evilbuny | Psykick: watching red dragon lately? |
05:47.24 | xtrvd | Nivex: I completely missed that comment up there. |
05:47.24 | FuriousGeorge | hey all |
05:47.31 | justinu | rustyb: you have any recomendations on which EC to use? are there perhaps improvements in zaptel i should be using? |
05:47.35 | Psykick | I take it none of you have seen 'Once Were Warriors' |
05:47.47 | outtolunc | so psykick, when have you done ANYTHING outside the norm |
05:47.52 | Nivex | xtrvd: I... I don't know that... AAAAAaaahhh! |
05:47.55 | evilbuny | Psykick: I have, not a very great movie though |
05:47.57 | xtrvd | =) |
05:48.05 | FuriousGeorge | so i made an account to telnet into theAPI expecting to see it spitting out all this stuff, and its just silen |
05:48.06 | evilbuny | although most hollywood garbage is worst |
05:48.26 | outtolunc | meaning if i call the FBI they have a file on you obviously |
05:48.26 | xtrvd | Nivex: Thank you good sir, you have made my night. |
05:48.26 | outtolunc | right? |
05:48.26 | Nivex | xtrvd: Glad to be of service. |
05:48.27 | Psykick | evilbuny: that is pretty much how things are/were for us/me when we were growing up |
05:48.27 | justinu | Psykick: so NZers like to eat people? |
05:48.34 | evilbuny | lol |
05:48.40 | outtolunc | well? |
05:48.43 | Psykick | NZer's or the maori used to be canibals |
05:48.55 | justinu | you a native? |
05:48.56 | Psykick | outtolunc: gee lets see |
05:49.00 | Psykick | I am a native |
05:49.05 | justinu | gotcha |
05:49.17 | *** join/#asterisk Eggplant (n=none@dsl-352.cascadeaccess.com) |
05:49.26 | outtolunc | i'm a 40+ marine that is tired of the bs from you candy ass mf'ers |
05:49.27 | evilbuny | Psykick: auckland airport = sucks, no wifi |
05:49.32 | Psykick | outtolunc: I've been in more fights than I can remember ... probably why I can't remember |
05:49.34 | outtolunc | how is that |
05:49.39 | rustyb | justinu you have a recent zaptel. i was checking to see which EC i'm using. I think its mark2 |
05:49.54 | outtolunc | i have a dd214 do you |
05:49.56 | justinu | rustyb: k... the default for my version was KB1 |
05:50.00 | Psykick | outtolunc: killed a guy for giving my g/f a hard time |
05:50.04 | outtolunc | i doubt it |
05:50.07 | justinu | heh |
05:50.10 | outtolunc | haha |
05:50.11 | Psykick | outtolunc: broke another guys legs |
05:50.15 | Psykick | outtolunc: broke another guys shoulder |
05:50.21 | outtolunc | <PROTECTED> |
05:50.23 | Psykick | outtolunc: tore 3 limbs off another |
05:50.24 | justinu | did you stew the guy? |
05:50.31 | outtolunc | right now |
05:50.33 | Psykick | not completely off |
05:50.37 | justinu | hmm |
05:50.42 | justinu | that's pretty brutal |
05:50.42 | Psykick | but enough now that he can't do jack |
05:50.54 | outtolunc | haha |
05:51.23 | Psykick | gave one guy such a beating that the hunchback of notre dame would look more appealing |
05:51.35 | justinu | you ever see casino? |
05:51.38 | outtolunc | oh yeah stay on the good side of the guy that 'might' have done something because someone/somehwhere said 'somehthing' about his gf |
05:51.38 | Psykick | I could go on and on |
05:51.49 | outtolunc | gee wally |
05:51.53 | outtolunc | FU |
05:51.57 | Psykick | outtolunc: I seriously don't care if you give my g/f a hard time |
05:51.58 | justinu | "you hear a little girl frankie?? is that a little girl? what happened to the tough guy who told me friend to stick it up his ass!" |
05:52.00 | Psykick | she's dead now |
05:52.01 | *** join/#asterisk sitexec (n=sitexec@69.62.163.65) |
05:52.03 | outtolunc | please |
05:52.37 | outtolunc | i'd already said, i'm showing up at spring von for ANYONE that thinks they are able.. to take a shot |
05:52.50 | outtolunc | if this means YOU |
05:52.50 | sitexec | just wondering how to add a local exten, i add mine in extentions.conf and i still get extention not found in local context |
05:52.54 | outtolunc | <PROTECTED> |
05:52.56 | justinu | high noon? right in front of the digium booth? |
05:53.05 | outtolunc | for all the rest of you... kiss my ass |
05:53.33 | outtolunc | but saying i'm NOT gonna do this or not gonna do that |
05:53.35 | outtolunc | haha |
05:53.36 | Psykick | so ... seriously ... go ahead ... make an arse of yourself ... I really don't give a fuck cos in all reality ... I can't be arsed to even give a shit |
05:53.39 | outtolunc | PLEASE |
05:54.08 | outtolunc | oh no, i must have studdered.. PLEASE |
05:54.09 | Psykick | I know myself that what I did was seriously fucked up |
05:54.14 | FuriousGeorge | anyone know of a good source of info to get started with the API? the wiki assumes I know stuff that i don't |
05:54.28 | Psykick | you know what |
05:54.32 | sitexec | feel free to message me if you can help |
05:54.34 | outtolunc | what the fuck part of 'i'll fuck you up' did you not understand? |
05:54.46 | xtrvd | the 'up' part |
05:54.55 | outtolunc | and i've been saying this for AGES |
05:54.55 | Psykick | I seriously hope you do rip someones head off .... mainly ... I hope you rip your own off |
05:54.57 | xtrvd | Because without it... it's kind of weird. |
05:55.15 | Psykick | but if we're all not that lucky ... you'll have a nice b/f to share a small room with |
05:55.28 | outtolunc | yet none of you 'hip young 'smart' guys' seem to make it to my front door |
05:55.47 | outtolunc | not to mention the less bugs filed |
05:56.02 | outtolunc | oh gee that must really be a bitch eh |
05:56.12 | outtolunc | any time |
05:56.19 | outtolunc | please! |
05:56.31 | outtolunc | or are you just a weenie <G> |
05:56.34 | Psykick | outtolunc: go buy yourself a carrot |
05:56.47 | outtolunc | for what? you gf? |
05:56.49 | Psykick | or any other vege you prefer |
05:56.59 | outtolunc | or are you a gf? |
05:57.08 | outtolunc | hmm |
05:57.22 | outtolunc | gee wally |
05:57.48 | outtolunc | awww |
05:57.57 | sitexec | just wondering how to add a local exten, i add mine in extentions.conf and i still get extention not found in local context |
05:58.04 | outtolunc | please don't go away mad... just f'n go away <G> |
05:58.45 | outtolunc | or is there anyone else here that 'thinks that know what i have/have not done for asterisk or other projects' that wants to be bold |
05:58.54 | xtrvd | sitexec: did you 'reload'? |
05:58.58 | outtolunc | 'please' |
05:59.24 | sitexec | sure didnt, thanks i will give it a try |
05:59.31 | xtrvd | =) No worries |
06:00.03 | Nivex | FYI: Don't need to reload the whole PBX, just 'extensions reload' |
06:00.05 | xtrvd | sitexec: just type 'reload' from the asterisk command prompt |
06:00.22 | xtrvd | sitexec: or as Nivex just explained, just 'extensions reload' |
06:00.25 | outtolunc | 'reload' does a 'base level reload' for asterisk |
06:00.32 | sitexec | xtrvd same error, no extention 'blah' in context 'local' |
06:00.41 | FuriousGeorge | anyone know a good reference for the * api |
06:01.14 | outtolunc | if you want to reload a specific part do a 'reload the_app_func_name_.so' |
06:01.22 | sitexec | hmm, i didnt see it get added |
06:02.18 | outtolunc | as for the reference of * api, i truely hate to say it ... but the source is the best |
06:02.34 | Psykick | outtolunc: genuine question .... ever do any interrogations? |
06:02.40 | *** join/#asterisk katakefalos (i=katakefa@194.214.77.65.in-addr.arpa.ethernext.com) |
06:02.59 | outtolunc | yes i have, from both sides |
06:03.09 | FuriousGeorge | outtolunc: thats kinda putting the horse before the carriage for a coding novice, dont you think |
06:03.17 | outtolunc | meaning, i've been, and have done |
06:03.24 | Psykick | oh ok |
06:03.37 | Psykick | been practicing lately? |
06:03.44 | outtolunc | fg you are not a novice, even though you 'keep saying so' |
06:03.59 | FuriousGeorge | novice=beginner right? |
06:04.15 | FuriousGeorge | am i getting my spoken languages confused? |
06:04.16 | outtolunc | after 1/5-2 of doing do either you 'get smart or you just 'hang here'' |
06:04.27 | mzo | i'm too busy breaking my asterisk to talk :P |
06:04.39 | Psykick | mzo: sounds ... very .... familiar |
06:04.40 | outtolunc | fg how long have you 'been hanging in this channel'? |
06:04.53 | outtolunc | it's been almost 2ish years |
06:04.53 | FuriousGeorge | i came in hear for the first time a year ago |
06:05.08 | outtolunc | so after 'that period of time' |
06:05.10 | FuriousGeorge | and i said coding novice |
06:05.14 | FuriousGeorge | not * novice |
06:05.20 | outtolunc | you are still just 'asking what'? |
06:05.22 | Psykick | fg: what language? |
06:05.38 | FuriousGeorge | in that time i did not learn C or Java |
06:05.54 | FuriousGeorge | Psykick: i dunno, heard python was easy, i knew pascal once |
06:06.14 | outtolunc | fg i never said you 'were' a novice, what i'm saying is you 'need not ask a question as if you were one' |
06:06.22 | Psykick | haven't gone the P Y way myself |
06:06.26 | *** join/#asterisk svenna_ (n=svenna@p548D388D.dip0.t-ipconnect.de) |
06:06.31 | outtolunc | but if you want to ask as if |
06:06.32 | Psykick | .... done the pascal thing too |
06:06.42 | outtolunc | we can go that route |
06:06.53 | sitexec | where do i put an exten in extentions.conf to get it to show up =\ |
06:07.03 | FuriousGeorge | i said i was a coding novice, which my understanding of english leads me to believe it is a step up beginner |
06:07.47 | outtolunc | and my statement is more to the point, i'm giving people more 'understanding' than i should |
06:07.53 | *** join/#asterisk salviadud (n=salviadu@201.133.209.101) |
06:07.57 | salviadud | heeeeelp |
06:08.03 | Psykick | lol |
06:08.12 | FuriousGeorge | outtolunc: have you been drinking, cuz im struggling to understand what you are getting at? |
06:08.16 | outtolunc | so lets talk asterisk <G> |
06:08.23 | *** join/#asterisk Goral (n=needsand@CPE0012172e9c9f-CM014080205433.cpe.net.cable.rogers.com) |
06:08.27 | salviadud | Feb 22 00:14:03 NOTICE[20764]: chan_iax2.c:7398 socket_read: Registration of '651692' rejected: 'Registration Refused' from: '192.246.69.186' |
06:08.35 | salviadud | whats that supposed to mean? |
06:08.37 | outtolunc | what is your 'current issue' with asterisk? |
06:09.00 | mzo | 192.246.69.186:4569 749414 <Unregistered> 60 Rejected |
06:09.00 | mzo | <PROTECTED> |
06:09.24 | FuriousGeorge | outtolunc: no issue, now that i got basic administration under some control i wanted to learn how to interface some simple scripts i could theoretically right with the API |
06:09.36 | outtolunc | (oh and btw: the best code i'd ever written i was drunk, so whats your f'n point <G>) |
06:09.40 | FuriousGeorge | but documentation on the API appears scarce, to put it nicely |
06:09.45 | *** join/#asterisk Abbas (n=Abbas@203.81.220.90) |
06:09.52 | mzo | salviadud, it's broken for me too :P |
06:10.04 | salviadud | ohhh |
06:10.10 | salviadud | i shouldn't worry then... |
06:10.11 | outtolunc | so, you are now able to admin an asterisk box (after 1 year +) |
06:10.34 | FuriousGeorge | thats more or less accurate |
06:10.42 | FuriousGeorge | insert small |
06:10.45 | outtolunc | and yet you inject yourself into a convo you think one of the parties 'might' be drunk... |
06:10.49 | outtolunc | gee wally |
06:11.01 | FuriousGeorge | im pretty convinced now |
06:11.22 | outtolunc | only took, what, 6 gee wally's |
06:11.31 | FuriousGeorge | lol |
06:11.58 | outtolunc | i hate to say it, but if you guys are 'our brightest' we are screwed |
06:12.45 | FuriousGeorge | dont hate grandpa that shit will kill you |
06:13.26 | outtolunc | i like my grandpa, he died when i was 6 |
06:13.41 | outtolunc | the other was dead before i was born |
06:14.00 | outtolunc | then my bother died when i was 12 |
06:14.16 | outtolunc | but you all think you know me |
06:14.25 | outtolunc | so who gives a shit |
06:14.44 | FuriousGeorge | you are a combative old drunk, arent you :) |
06:14.56 | outtolunc | so which one of you 'intelligent mf'ers' wants to come say 'hi to me' |
06:15.20 | outtolunc | i've not changed one iota in the last hour+ |
06:15.29 | outtolunc | at all |
06:15.31 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
06:15.40 | outtolunc | have i been attacked by others |
06:15.41 | outtolunc | <PROTECTED> |
06:15.41 | FuriousGeorge | you're "maintenence boozing" |
06:15.48 | outtolunc | in various forms |
06:16.01 | outtolunc | but yet you still want to attack 'just me' |
06:16.07 | outtolunc | ok, here we go |
06:16.11 | evilbuny | FuriousGeorge: give him a mickey fin reciepe :) |
06:16.48 | FuriousGeorge | evilbuny: would that help, him? i'm not sure what it is, tbh |
06:16.55 | outtolunc | fg in the time you have been in this channel i can NOT believe that you have only accumulated the knowledge you have... |
06:17.08 | FuriousGeorge | yeah, its tough |
06:17.13 | evilbuny | what the air stewedesses give drunk people to put em to sleep on planes :) |
06:17.24 | outtolunc | oh damn, i'm drunk, i should be more defensive |
06:17.33 | FuriousGeorge | i always say the hardest thing about asterisk is propper linux administation |
06:17.38 | outtolunc | fg i think you need to spank yourself |
06:17.48 | FuriousGeorge | ? |
06:17.58 | outtolunc | damn, thats not drunk that's just idotic |
06:18.00 | evilbuny | he likes it when you do that fg :) |
06:18.06 | outtolunc | well GEE WALLY |
06:18.08 | FuriousGeorge | dude, you can tab complete my name |
06:18.10 | evilbuny | it turns him on :) |
06:18.29 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
06:19.24 | outtolunc | i'm so glad i can come here to remind myself not to help you guys |
06:19.46 | FuriousGeorge | lol, its probably better that way |
06:19.47 | [hC] | bahaha |
06:19.57 | [hC] | there needs to be like, a divide here. |
06:20.08 | FuriousGeorge | a drunk tank? |
06:20.24 | [hC] | asterisk at home help, "What is SIP/Asterisk/VoIP" channel, and an actual asterisk channel for intelligent conversation |
06:20.28 | outtolunc | which is why when you do a lookup for mods done by FuriousGeorge you'll find MANY |
06:20.35 | FuriousGeorge | ~amp |
06:20.35 | jbot | hmm... amp is NOT supported here! people using it should join #amportal |
06:20.37 | outtolunc | insert any of you lamers |
06:20.46 | [hC] | one down. |
06:21.26 | Qwell | [hC]: heh |
06:21.29 | outtolunc | i came here to help and if you review i did, till someone thought they could come after me |
06:21.30 | Qwell | [hC]: there already is |
06:21.34 | [hC] | Sup qwell.. |
06:21.36 | FuriousGeorge | dude, i just said "im a novice coder" and "i knew pascal once", thats like moking me for not being in the winter olympics cuz i snowboard |
06:21.42 | Qwell | #amportal, #asterisk, and #asterisk-dev |
06:21.47 | Qwell | :p |
06:22.05 | [hC] | Oooh look at mr wikipedia over here... haha :P |
06:22.28 | outtolunc | you surely aren't talking about me |
06:22.30 | salviadud | gui's are for pussies! |
06:22.37 | [hC] | No, qwell. |
06:22.43 | outtolunc | i've NEVER posted to ANY wiki |
06:22.44 | outtolunc | ever |
06:22.50 | Qwell | outtolunc: shh |
06:22.54 | Qwell | not everything is about you |
06:22.56 | Qwell | go play |
06:22.58 | outtolunc | k |
06:24.23 | mzo | i broke my asterisk, oops :) |
06:24.36 | FuriousGeorge | did it hurt |
06:24.41 | mzo | yah |
06:24.43 | mzo | someone called |
06:24.47 | mzo | and i got like 3000 pages of scroll |
06:24.57 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
06:25.00 | FuriousGeorge | unfortunate, that |
06:26.10 | mzo | yay |
06:27.25 | outtolunc | this gets better right? |
06:27.26 | FuriousGeorge | so anyway, i telnet into my asterisk box's API today for the first time, and i expect to see it spitting out all this info, events and whatnots, and its silent |
06:27.43 | outtolunc | guess not |
06:28.00 | outtolunc | afk |
06:29.39 | *** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de) |
06:29.44 | exstatica | anyone seen this? |
06:29.44 | exstatica | Starting Zap/4-1 at internal,s,1 still failed so falling back to context 'default' |
06:29.59 | outtolunc | haha |
06:30.03 | exstatica | i'm trying to get incoming calls from pstn to work |
06:30.18 | outtolunc | karma is a bitch |
06:31.02 | [av]bani | FuriousGeorge: you need to login first |
06:31.29 | FuriousGeorge | http://pastebin.ca/42743 |
06:31.37 | FuriousGeorge | [av]bani: check it out i tried |
06:31.48 | outtolunc | put simply, asterisk will 'fall thru' to the [default] context, the best suggestion is to have 'playback .. what are you doing here' in the [default] context |
06:32.41 | outtolunc | and if 'others' tell you it's ok to play in '[default]' land, you are on your own <G> |
06:32.43 | [av]bani | FuriousGeorge: its wrong |
06:32.50 | FuriousGeorge | exstatica: put your zap channel in the "incoming calls context", whatever you wanna call it |
06:32.53 | FuriousGeorge | [av]bani: duh :) |
06:33.06 | [av]bani | Action: login |
06:33.12 | [av]bani | Username: blabla |
06:33.17 | [av]bani | Secret: password |
06:33.31 | FuriousGeorge | then it will acknowledge me? lets see |
06:33.32 | [av]bani | Events: on |
06:33.38 | [av]bani | and two CR's |
06:34.15 | FuriousGeorge | wow, exactly how i thought it should work, i cant belive i didnt think to try that :) |
06:34.28 | outtolunc | gee wally |
06:34.31 | exstatica | FuriousGeorge: what do you mean? |
06:34.33 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
06:34.42 | outtolunc | send me to [default] PLEASE |
06:35.13 | outtolunc | sad thing is he probably didn't get that |
06:35.59 | FuriousGeorge | exstatica: in your zapata.conf file you see context=default somewhere? |
06:36.26 | FuriousGeorge | [av]bani: are the commands case sensitive |
06:36.40 | FuriousGeorge | Action vs action |
06:36.50 | [av]bani | FuriousGeorge: try it and find out! |
06:37.03 | FuriousGeorge | [av]bani: maybe i will |
06:37.23 | exstatica | FuriousGeorge: i see context incomming |
06:38.08 | *** join/#asterisk CKGC (i=CKGC@202.8.86.162) |
06:38.11 | FuriousGeorge | you spelled incoming wrong :) |
06:38.15 | FuriousGeorge | well thats ok |
06:38.32 | FuriousGeorge | just put in extensions.conf |
06:38.33 | outtolunc | so ok, you felt you had to say something <G> |
06:38.34 | exstatica | question, i think my channels might be mixed up but i'm not sure |
06:38.40 | *** part/#asterisk CKGC (i=CKGC@202.8.86.162) |
06:38.49 | exstatica | the signaling is the opposite of what the modules is? |
06:38.53 | FuriousGeorge | outtolunc: me and you both, brother |
06:39.04 | outtolunc | hehe, not sure of that |
06:39.15 | outtolunc | i'm a bit darker |
06:39.51 | outtolunc | remember, my brother died when i was 12 |
06:40.05 | FuriousGeorge | anyway, exstatica, every channel below that line is gonna default to incomming context |
06:40.33 | exstatica | http://pastebin.com/566318 |
06:40.39 | exstatica | i think i have them mixed up |
06:40.42 | FuriousGeorge | so make sure in extensions.conf you have a [context] called [incomming] |
06:40.55 | exstatica | module 1 is my fxs card, and 4 is fxo |
06:41.33 | outtolunc | so glad i wasn't in that convo |
06:42.21 | outtolunc | all that for a 4 port fxs/fxo board gone arye |
06:42.36 | outtolunc | bloody f'n hell |
06:45.09 | FuriousGeorge | http://pastebin.com/566320 |
06:45.25 | FuriousGeorge | define it once, its good for every channel defined below |
06:45.31 | FuriousGeorge | easy when you think about it |
06:46.01 | outtolunc | that is unless you look at inherited channel vars |
06:46.24 | FuriousGeorge | so [from-pstn] is the context that needs to have an s extension which eventually calls you |
06:46.33 | outtolunc | but i'm sure you were being 'general' kinda like 'default' <G> |
06:46.37 | FuriousGeorge | (some people like to answer & wait first) |
06:48.05 | outtolunc | i want to ask you |
06:48.33 | outtolunc | do you really understand why 'fall thru to [default]' is a bad thing |
06:48.41 | outtolunc | ? |
06:49.15 | outtolunc | if you don't, there is no reason to continue on my part |
06:49.17 | FuriousGeorge | outtolunc: shhhh, busy |
06:49.31 | outtolunc | ah ok, you are busy.. gotchya |
06:49.53 | dudes | Busy getting your internet gf hooked up? |
06:50.09 | FuriousGeorge | exstatica: dont forget to reload extensions |
06:50.22 | outtolunc | dudes, hush <G> |
06:50.34 | outtolunc | ty, but hush <G> |
06:51.15 | outtolunc | well, actaually that isn't fair to you |
06:51.26 | outtolunc | since you have been though this |
06:51.38 | outtolunc | you poor scared being you |
06:53.29 | FuriousGeorge | outtolunc: what are your thoughts on a good solution for state monitoring of parking spots w/ 1.2.4. any experience with bri stuff and app_devstate? if your gonna blather incessantly we may as well try to channel you |
06:53.45 | outtolunc | state monitoring |
06:54.05 | FuriousGeorge | w/ devstate |
06:54.05 | outtolunc | is not state monitoring in any sense in any version |
06:54.10 | FuriousGeorge | bristuff |
06:54.17 | outtolunc | regardles |
06:54.37 | outtolunc | t1/pri/bri/whatever |
06:54.37 | FuriousGeorge | what? |
06:55.01 | FuriousGeorge | make your point man |
06:55.03 | outtolunc | trying to get 'channel states' in a 'nice manner' is not happening |
06:55.15 | outtolunc | you can understand that right? |
06:55.17 | FuriousGeorge | how hard can it be? |
06:55.34 | FuriousGeorge | you write all these mods and yet we cant monitor 5 parking spots? |
06:55.41 | outtolunc | the reason why is that for it to be helpfull, state change happens ALOT |
06:55.52 | outtolunc | nonono |
06:56.03 | outtolunc | 'monitor 5 parking spots' is simple |
06:56.08 | FuriousGeorge | that get used say 50 times a day |
06:56.19 | FuriousGeorge | so you gotta set devstate 10 times for each |
06:56.22 | FuriousGeorge | where does it get hard? |
06:56.27 | outtolunc | monitor all 500 channels in those 5 parking LOTS is a bitch |
06:56.42 | outtolunc | which part is hard for you to understand? |
06:56.46 | FuriousGeorge | spots not lots |
06:56.50 | outtolunc | 5 spots |
06:56.56 | outtolunc | is like 5 channels |
06:57.09 | outtolunc | is like why is there a f'n prob? |
06:57.13 | outtolunc | it's 5 |
06:57.30 | FuriousGeorge | font size="2">(01:56:27) outtolunc: monitor all 500 channels in those 5 parking LOTS is a bitch <--- who's not understanding |
06:57.38 | outtolunc | not 96, times 2 or 4 or 6 |
06:58.28 | FuriousGeorge | sip-so and so was transfered to parking spot 1, set devstate. sip-so and so was transfered out of parking spot 1, set devstate |
06:58.36 | FuriousGeorge | i dont see where the disconnect is |
06:59.06 | outtolunc | fg, if you really have issue with keeping track of 5 'things' and for some reason you missed my 'sendevent' patch (yeah i know i'm just a f'n moron) but anyways... then you might need to look for another aspect to do) |
06:59.43 | FuriousGeorge | my bad for encouraging you to speak |
06:59.57 | FuriousGeorge | wont happen again |
07:00.00 | outtolunc | hahah |
07:00.09 | outtolunc | have a nice f'n day |
07:00.19 | Juggie | ladies, ladies |
07:00.53 | outtolunc | oh btw, please feel free to try and make my feel inferior in ANY WAY SHAPE OF FORM |
07:00.58 | outtolunc | hah |
07:01.24 | outtolunc | er OR (since you didn't catch that) |
07:01.31 | Juggie | everyone give outtolunc a hug |
07:01.49 | FuriousGeorge | no, he reaks of cheap rum and feet |
07:02.03 | outtolunc | tosses a slurpie (big ugly green sickie thing) at juggie |
07:02.08 | outtolunc | hehe |
07:02.18 | outtolunc | my luck you'ld catch it <G> |
07:02.25 | *** join/#asterisk Goral (n=needsand@CPE0012172e9c9f-CM014080205433.cpe.net.cable.rogers.com) |
07:02.33 | Goral | sorry about that all |
07:02.40 | X-Rob | outtolunc, please ignore FuriousGeorge, he seems to believe that applcation hints are easy. |
07:02.53 | X-Rob | Even though I've pointed him at metermaid and various other stuff. |
07:02.56 | outtolunc | hints, damn, i knew i forgot something |
07:03.00 | jeebusroxors | is there anyway to see if my cdr is getting put into mysql? |
07:03.09 | Goral | is there any other module for webmin other than thirdlane? |
07:03.20 | dpryo | jeebusroxors: Make a call, and look in the table. |
07:03.31 | jeebusroxors | dpryo; it should be any call correct? |
07:03.39 | outtolunc | is it me, or does that question seem. out there |
07:03.44 | jeebusroxors | then what? select * from cdr |
07:03.46 | dpryo | jeebusroxors: yes, unless you've defined some special cdr-events |
07:03.58 | jeebusroxors | outtolunc; my question? heh |
07:04.20 | FuriousGeorge | X-Rob: metermaid = multiparking patch by oej? we've been over this, trunk isnt a viable alternative, and its just a pet project of mine |
07:04.22 | *** join/#asterisk justnulling2 (i=justnull@ool-18bab443.dyn.optonline.net) |
07:04.22 | Juggie | jeebusroxors, mysql status |
07:04.23 | outtolunc | thirdlane did NOT invent, mysql, nor anything that happened between that and the 'advent of thirdlane' |
07:04.26 | Juggie | think the command line is |
07:04.41 | Juggie | or it might be cdr mysql status |
07:04.42 | outtolunc | so you MIGHT want to look for useful tools, THEY DO EXIST |
07:04.43 | Juggie | i forget |
07:04.44 | Juggie | look |
07:05.13 | jeebusroxors | i will Juggie - thanks |
07:05.14 | FuriousGeorge | X-Rob: you know im using bristuff patch with this right? |
07:05.36 | X-Rob | Yeah, because you want devstate. |
07:05.42 | jeebusroxors | fyi im trying to get asterisk-stat setup and i dont have any calls listed |
07:06.13 | FuriousGeorge | X-Rob: perhaps im having a fundamental misunderstanding of how this works, but isnt the api gonna spit something out when i park someone and unpark them? |
07:06.20 | FuriousGeorge | cant i set a devstate based on that? |
07:06.23 | outtolunc | jeebus.. i've not seen an actual question |
07:06.34 | X-Rob | FuriousGeorge, no, you can't. |
07:06.43 | outtolunc | at most, i have 'this installed' and it's 'not working' |
07:06.44 | X-Rob | you want application hints. Metermaid gives you that. |
07:06.53 | justnulling2 | what is the best was to setup vm for smallbusiness that wants to look like largebusiness by having more ext then ppl? |
07:06.56 | X-Rob | If you don't want to use trunk, then you can't use metermaid, so you're stuck. |
07:07.03 | jeebusroxors | outtolunc; i was asking if there was a way to check if my CDR was in mysql |
07:07.11 | X-Rob | And, really don't use trunk in a production environment |
07:07.13 | dudes | do a query and see |
07:07.18 | outtolunc | well cdr in mysql is a prob |
07:07.34 | X-Rob | So what you want to do can't be done, yet. Give it 6 months |
07:07.35 | outtolunc | what version you using of everything |
07:07.37 | FuriousGeorge | X-Rob: im not trying to disagree, but im not getting why i cant call an extension from the api that sets devstates when it gets info about park and unpark |
07:07.42 | X-Rob | you know, I did say that the very first time you joined and asked that question |
07:07.55 | jeebusroxors | asterisk is 2.4 |
07:07.59 | X-Rob | because you can't get info about park and unpark. |
07:07.59 | *** join/#asterisk Gir19 (n=JDepp@67.189.110.174) |
07:08.02 | jeebusroxors | sql is .18 |
07:08.10 | X-Rob | Woo. |
07:08.13 | outtolunc | meaning, mysql cdr's has NOT, and will NOT be standard for what, the last 1.75 years |
07:08.20 | X-Rob | jeebusroxors has a time machine and has asterisk 2.4! |
07:08.23 | FuriousGeorge | X-Rob: i dont remember you saying that, tbh |
07:08.30 | outtolunc | but noone is counting |
07:08.32 | jeebusroxors | hehe |
07:08.42 | X-Rob | Does it have application hints? FuriousGeorge wants a copy. |
07:08.43 | jeebusroxors | outtolunc; so what- postgres is standard? |
07:08.49 | FuriousGeorge | X-Rob: np, ill use meetme's, itll be a bit harder but ill learn more and thats the point anyway |
07:08.58 | outtolunc | so, you have what asterisk, and what asterisk-addons versions? |
07:09.08 | brookshire | use odbc cdr :) |
07:09.25 | outtolunc | jeebus, yes, pgsql IS STANDARD nowdays |
07:09.31 | jeebusroxors | brookshire; ive never played with odbc and am having errors with it heh |
07:09.43 | brookshire | http://www.voip-info.org/wiki-Asterisk+cdr+odbc |
07:09.45 | brookshire | ?? |
07:09.46 | outtolunc | if you aren't aware of that, you HAVE been gone 'awhile' |
07:10.13 | jeebusroxors | outtolunc; im a newb to asterisk heh |
07:10.38 | outtolunc | by any chance do you remember the gpl/lgpl mysql fiasco almost 2 years ago |
07:10.39 | jeebusroxors | brookshire; i read that - im getting errors on connecting to my ds - isql isnt working |
07:10.49 | FuriousGeorge | X-Rob: the only question becomes can i use the api to allow my peers to x-fer outside party to a meetme if and only if there isnt another outside party there |
07:10.55 | outtolunc | well, thats how long it's been |
07:10.58 | outtolunc | so |
07:11.21 | outtolunc | either, read some more, or dare i say, figure something else out |
07:11.37 | brookshire | but.. odbc is the way to go with asterisk |
07:11.41 | outtolunc | it's been almost 2 freakin years for petesake |
07:12.18 | brookshire | i'm sure it's just a simple configuration misconfiguration |
07:12.32 | jeebusroxors | brookshire; im sure too - just gotta find out where heh |
07:12.34 | brookshire | he |
07:12.35 | brookshire | heh |
07:12.57 | Gir19 | I have a copy of ppc-asterisk0.1 |
07:13.06 | outtolunc | sweet |
07:13.12 | outtolunc | you gonna share? |
07:13.29 | Gir19 | I'll share when I get the bugs out of it. |
07:13.34 | *** join/#asterisk jeebusroxors (n=jeebus@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
07:13.48 | Gir19 | It tends to lockup my ppc every couple of hours. |
07:14.01 | outtolunc | isn't life great |
07:14.30 | Gir19 | but I wanted to be able to make iax2 calls from my ppc's wifi. |
07:14.38 | outtolunc | i get to listen to assholes all night give me shit about asterisk and 'i've given back already' and here is you |
07:14.47 | outtolunc | gee wally |
07:15.04 | outtolunc | but i'll say cool and move on |
07:15.13 | outtolunc | i'm just not it the f'n mood |
07:15.41 | Gir19 | ah, havin some clients that are not pleased with asterisk? |
07:15.57 | outtolunc | haha clients, hell ASTERISK PEOPLE HERE |
07:16.08 | Gir19 | oh, lol |
07:16.11 | outtolunc | people i'm trying to HELP |
07:16.43 | outtolunc | strange eh |
07:17.07 | Gir19 | yeah, I was helpin earlier and some just couldn't believe me or just refused. |
07:17.31 | outtolunc | you are real aren't you? <G> |
07:17.41 | Juggie | outtolunc, seinfeld: serenity now |
07:17.48 | outtolunc | damn |
07:17.53 | Juggie | nothing good will come from venting, some people are just slow they cant help it :) |
07:18.03 | outtolunc | oh oh |
07:18.13 | outtolunc | i got a good one (today even) |
07:18.24 | outtolunc | lets see if i get this correct |
07:18.34 | Juggie | we want even the people who dont try, to learn |
07:18.39 | Juggie | they will get it eventually |
07:18.40 | Gir19 | it's all good. I get vented on and sometimes I need to be the one venting, it all balances out eventually. |
07:18.49 | outtolunc | actually i think i already said it tonight |
07:19.12 | Juggie | we dont want to turn people away from * just because we dont like their question. |
07:19.21 | outtolunc | i never have |
07:19.25 | Juggie | sometimes just pointing people at voip-info and telling them to go read |
07:19.28 | Juggie | is a good answer |
07:19.43 | Gir19 | I also point them to asteriskguru |
07:20.46 | Gir19 | the problem I have with some of those sites is that alot of the questions and answers haven't been updated and are way out dated. |
07:21.09 | outtolunc | the prob with that site, they want help with ' |
07:21.18 | outtolunc | thier software' |
07:21.27 | outtolunc | but it's conditional |
07:22.07 | Gir19 | I'm still trying to workout a problem on one of my recent servers and the poping noise on the sip side, but not on the pots side. |
07:22.30 | outtolunc | meaning, i wanted to help with the one app (the one that showed statuses), but i was told, 'when we get back to that' (this was what, 8 months ago) |
07:23.01 | outtolunc | so , i'm not too happy to 'be thankful' |
07:24.26 | Gir19 | anyone know much about how well asterisk runs on an AMD system? I ask cause I am considering switching from Intel chips to AMD. |
07:26.09 | brookshire | pretty well :) |
07:26.39 | Gir19 | any major differences with the 64 chips? |
07:26.53 | outtolunc | the thing that i'd watch for is the threading |
07:27.32 | Juggie | why would threading be effected? |
07:27.33 | Qwell | yeah, AMD processors don't thread at all |
07:27.40 | outtolunc | i run it on a DC intel with HT disabled just fine, with HT enabled it gets alittle freaky sometimes |
07:27.54 | Juggie | outtolunc, i run asterisk on dual EMT64 xeon systems |
07:28.06 | Juggie | (2 real, 4 virtual processors total) |
07:28.09 | Juggie | and i have no problems |
07:28.12 | outtolunc | which ones? |
07:28.19 | Juggie | Xeon 3.2 EMT64 |
07:28.28 | outtolunc | i ran mine with HT enabled for the first 3 weeks |
07:28.31 | X-Rob | Mmmmshiny xeon |
07:28.37 | Juggie | running centos 4.2 64bit |
07:28.42 | Juggie | 5 gigs of ram |
07:28.44 | brookshire | qwell: they can thread better than mac processors.. they only have one thread |
07:28.45 | outtolunc | but there were issue after issue |
07:28.45 | Qwell | eww! centos?! |
07:28.46 | brookshire | lol |
07:28.47 | Juggie | 500gigs of hdd. |
07:28.50 | Qwell | brookshire: haha |
07:29.04 | Juggie | Qwell, its corporate, centos is close to rhel |
07:29.13 | Qwell | Juggie: Just messing with you. :) |
07:29.14 | Juggie | and also all the tools for the servers eg the HP server tools |
07:29.22 | Juggie | work on centos |
07:29.26 | Juggie | because they were desgned for rhel |
07:29.30 | Qwell | centos is fine, as long as it isn't *@~ |
07:29.37 | Juggie | haha |
07:29.40 | Juggie | common |
07:29.42 | Juggie | what do you think |
07:29.46 | Qwell | :p |
07:29.46 | Juggie | you met me |
07:29.50 | Qwell | newb :D |
07:29.50 | Juggie | do i see like a @home kinda guy |
07:30.16 | Juggie | guys who run @ home dont drink like 6 pitchers at a hick bar and still go to the 9am presentation the next day :P |
07:30.21 | Qwell | heh |
07:30.31 | Qwell | You had cheap American beer. ;] |
07:30.38 | brookshire | i bet centos only has one thread |
07:30.39 | Juggie | 5$ for a pitcher |
07:30.43 | Juggie | is good though |
07:30.48 | Juggie | even the good beer was like 6$ |
07:30.48 | Qwell | brookshire: centos has like 2 threads |
07:30.53 | Qwell | 1 is for the kernel |
07:30.57 | brookshire | ahh.. right.. |
07:31.11 | Qwell | That's seriously the cheapest beer I've ever seen |
07:31.11 | Juggie | [root@TRN-HTTP-SV01 ~]# free |
07:31.11 | Juggie | total used free shared buffers cached |
07:31.11 | Juggie | Mem: 5015180 4582256 432924 0 55440 4293280 |
07:31.16 | Juggie | umm |
07:31.18 | outtolunc | $5/pitchers.... damn starts having a party |
07:31.22 | brookshire | free -m ! |
07:31.42 | Juggie | [root@TRN-HTTP-SV01 ~]# free -m |
07:31.42 | Juggie | total used free shared buffers cached |
07:31.42 | Juggie | Mem: 4897 4474 422 0 54 4192 |
07:31.48 | brookshire | :) |
07:31.53 | Qwell | -m? |
07:31.57 | Juggie | megs |
07:31.57 | brookshire | free naked is sooo over rated |
07:32.07 | Qwell | ahh...silly free |
07:32.09 | Qwell | should use -h |
07:32.09 | outtolunc | he could have done -h <G> |
07:32.14 | brookshire | hah |
07:32.17 | outtolunc | hehe |
07:32.23 | Qwell | -h doesn't exist on free though |
07:32.23 | brookshire | qwell: we should right a patch |
07:32.32 | Qwell | brookshire: we should |
07:32.39 | brookshire | and submit it to them |
07:32.43 | Qwell | though... |
07:32.44 | brookshire | i wonder what they would say |
07:32.46 | Qwell | I'd write one |
07:33.13 | brookshire | i can just see it now... |
07:33.26 | brookshire | "here's for not following standard unix -h" |
07:33.32 | brookshire | lol |
07:33.36 | Juggie | my free does -h |
07:33.42 | Qwell | Juggie: what version? |
07:33.51 | *** part/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net) |
07:33.56 | Qwell | procps version 3.2.5 |
07:34.09 | Juggie | [root@TRN-HTTP-SV01 ~]# free -V |
07:34.09 | Juggie | procps version 3.2.3 |
07:34.13 | Qwell | wtf |
07:34.14 | Qwell | those newbs |
07:34.16 | Juggie | well, |
07:34.19 | Juggie | i see now |
07:34.24 | Juggie | it says invalid option |
07:34.25 | Juggie | and prints help |
07:34.27 | Qwell | heh |
07:34.30 | brookshire | haha |
07:34.35 | Qwell | is it in the manpage? |
07:34.40 | outtolunc | are you two done yet <G> |
07:35.05 | Qwell | it should totally follow like how du does it |
07:35.17 | X-Rob | Oooooh |
07:35.21 | Qwell | with GNU opts |
07:35.29 | brookshire | qwell: and df |
07:35.33 | Juggie | jeeze, i havnt updated centos in a while |
07:35.34 | Qwell | yeah |
07:35.37 | Juggie | 39 updates avail |
07:35.41 | X-Rob | (all I did was tick 'use multiple burners' in nero, but I still feel proud of myself) |
07:35.47 | Juggie | haha.... anyone watch king of the kill |
07:35.51 | Juggie | the best eppisode is on fox |
07:35.52 | Qwell | should be so easy to do |
07:35.58 | Juggie | "thats my purse, i dont know you" |
07:36.08 | Qwell | hell...it probably uses the same code for -b, -k, -m, -g |
07:36.18 | Qwell | erm, no -g |
07:36.26 | Juggie | mine has -g |
07:36.33 | brookshire | they are going to have to add a -t one day |
07:36.33 | Qwell | erm...yeah, so does mine |
07:36.37 | Qwell | not in the man page though |
07:36.39 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-21.claranet.co.uk) |
07:36.40 | brookshire | see.. it just keeps grown |
07:36.53 | brookshire | growing |
07:36.57 | Juggie | hmmm |
07:37.02 | Qwell | -t is already used in free |
07:37.03 | Juggie | how can i schedule yum to auto update |
07:37.05 | outtolunc | sorry, bad 'purse' thing |
07:37.16 | Qwell | Juggie: cron? |
07:37.19 | outtolunc | imagine that <G> |
07:37.31 | outtolunc | tag every purse in 'a city' |
07:37.41 | outtolunc | now track |
07:37.42 | outtolunc | 'em |
07:37.47 | Qwell | haha, this rocks |
07:37.58 | outtolunc | everyone, everyplace, all the time |
07:37.59 | Qwell | yum has been running a "yum remove xorg-x11-libs" on my router for like... |
07:38.02 | Qwell | 3 hours |
07:38.09 | outtolunc | eww |
07:38.13 | Juggie | i think theres a yum service |
07:38.13 | brookshire | juggie: /etc/init.d/yum start |
07:38.14 | Qwell | and it's still calculating deps |
07:38.48 | Qwell | that freaking rocks |
07:38.48 | outtolunc | well rock on, i'm gonna get some rest |
07:38.48 | Qwell | chkconfig yum on |
07:38.48 | outtolunc | gnight |
07:38.48 | Juggie | yah, i just found that, thanks |
07:38.53 | Juggie | only problem is i need to use a http server to get out |
07:39.01 | Juggie | i wonder if the proxy env var will exist |
07:39.07 | Qwell | should |
07:39.19 | Juggie | i think i'm setting it in .profilerc |
07:39.21 | Qwell | if not, you can put it in the /etc/sysconfig/ file it calls |
07:39.23 | Juggie | i forget where i put it |
07:39.27 | Juggie | yah true |
07:39.37 | brookshire | qwell: here is one for you |
07:39.39 | salviadud | hey, what does . mean? |
07:39.40 | Qwell | it should run a shell though |
07:39.44 | salviadud | as in period? |
07:39.45 | Qwell | salviadud: in what context? |
07:39.51 | salviadud | for example... |
07:39.53 | Qwell | in unix, it means current dir |
07:39.56 | brookshire | mv /etc/rc3.d/K01yum /etc/rc3.d/S01yum |
07:39.57 | salviadud | dialing an extension |
07:40.06 | Qwell | in Asterisk dialplan, it means "match any number of anything" |
07:40.07 | Juggie | 64bit linux screams |
07:40.07 | brookshire | mv /etc/rc5.d/K01yum /etc/rc5.d/S01yum |
07:40.08 | Juggie | its so nice |
07:40.09 | brookshire | :D |
07:40.16 | Qwell | brookshire: silly debian ;) |
07:40.23 | salviadud | ohhhh |
07:40.24 | Juggie | brookshire, i only run in level3, 5 is evil :) |
07:40.26 | salviadud | thanx |
07:40.30 | Qwell | chmod +x /etc/rc.d/S01yum |
07:40.40 | salviadud | i use slackware... no probs here... |
07:40.50 | Juggie | 5 is xwin right? |
07:40.52 | Qwell | I prefer `rc-update add service default` |
07:40.56 | Qwell | Juggie: centos? yeah |
07:41.03 | Juggie | yah, i dont run xwin |
07:41.03 | brookshire | yum remove yum |
07:41.07 | X-Rob | or 'chkconfig yum on' |
07:41.12 | Qwell | X-Rob: see above |
07:41.29 | FuriousGeorge | ive heard some dude from digium talking about a "registration manager" in asterisk to better handel my *.dynu.com dynamic ips. anyone know wtf he was talking about? i cant make google tell me |
07:41.47 | brookshire | juggie: can you actually run in any other level? |
07:41.59 | Qwell | FuriousGeorge: never heard of a registration manager, but you can use externhost |
07:42.02 | Juggie | brookshire, i'm sure you can |
07:42.09 | camonz | hi, if i want to send a call to voicemail after it has ringed 10 secs to a extension how would i do that? |
07:42.15 | Qwell | RH distros have 2, 3, 5 |
07:42.16 | camonz | with n+101 priority? |
07:42.17 | brookshire | init 11 |
07:42.19 | brookshire | lol |
07:42.21 | Qwell | (and 0, 1, 6, of course) |
07:42.22 | FuriousGeorge | qwell, thanks ill check into that |
07:42.27 | brookshire | 2 is single right? |
07:42.30 | brookshire | 6 is reboot |
07:42.31 | Qwell | 1 is single |
07:42.34 | Juggie | look in ... /etc/inittab |
07:42.35 | brookshire | 0 is powerdown? |
07:42.36 | Juggie | they are all in there. |
07:42.36 | Qwell | 0 is startup, 6 is shutdown |
07:42.42 | brookshire | oh yeah |
07:42.43 | Qwell | 2 is no network CLI |
07:42.56 | Juggie | # Default runlevel. The runlevels used by RHS are: |
07:42.56 | Juggie | # 0 - halt (Do NOT set initdefault to this) |
07:42.56 | Juggie | # 1 - Single user mode |
07:42.57 | Juggie | # 2 - Multiuser, without NFS (The same as 3, if you do not have networking) |
07:42.57 | Juggie | # 3 - Full multiuser mode |
07:42.59 | Juggie | # 4 - unused |
07:43.00 | brookshire | i knew this at one point |
07:43.01 | Juggie | # 5 - X11 |
07:43.02 | Qwell | yeah |
07:43.03 | Juggie | # 6 - reboot (Do NOT set initdefault to this) |
07:43.07 | brookshire | but it's kinda pointless |
07:43.10 | Qwell | oh...0 is halt |
07:43.15 | brookshire | HAHA |
07:43.19 | Juggie | who cares, use 3 :) |
07:43.22 | brookshire | juggie: # init 6 |
07:43.37 | Qwell | echo "telinit 6" >> /etc/local.start |
07:43.37 | Qwell | :D |
07:43.44 | Juggie | if i actually want to reboot a server, can i do init 6 |
07:43.47 | brookshire | everyone should use init 69 man! |
07:43.49 | Qwell | That's a real bitch to figure out |
07:44.11 | brookshire | turn your server into a "love machine" |
07:44.21 | Juggie | the only love machine in the lab is me |
07:44.31 | Juggie | i dont need any competition |
07:44.53 | brookshire | depends on who you're loving.. i guess.. lol |
07:45.26 | salviadud | is anybody else having trouble with FWD? |
07:45.35 | salviadud | im getting a bunch of *CLI> Feb 22 01:44:58 NOTICE[21019]: chan_iax2.c:7398 socket_read: Registration of '651692' rejected: 'Registration Refused' from: '192.246.69.186' |
07:46.14 | brookshire | i guess they don't like you anymore :( |
07:46.29 | salviadud | they never liked me! im a prankster... |
07:46.41 | salviadud | i call rehab clinics and lie that im drunk |
07:46.47 | salviadud | with a redneck voice |
07:46.53 | FuriousGeorge | Qwell: thats not the same as extenip? it appears thats for sip, the problem im having is with iax. i register=> with eachother, and then they have their respective freinds, but when i get a new ip they lose eachother for a while |
07:47.18 | FuriousGeorge | i just thought there should be some way to have it not cache the ip, which appears to be what * is doing |
07:47.19 | IronHelix | ahlll tell yoo whnnn ivh had enufffff.... |
07:47.37 | salviadud | hahaha, yeah, iron thats the spirit |
07:48.03 | IronHelix | hehe |
07:48.10 | salviadud | one guy wanted to call an ambulance, and i told him "no son.... im over here *hick* attt. walmart, im getting more booze" |
07:48.31 | FuriousGeorge | ~externhost |
07:48.32 | salviadud | i want to mess around with the monitor function |
07:48.41 | *** join/#asterisk P0L0 (n=n0n3@62-43-65-175.user.ono.com) |
07:48.44 | salviadud | tape all my pranks and social engineering hacks |
07:49.25 | salviadud | wish me luck... cya guys in the mornin' |
07:49.31 | IronHelix | cya |
07:49.35 | IronHelix | also- dont abuse hotlines |
07:49.37 | IronHelix | yeah |
07:49.38 | IronHelix | thats bad |
07:49.43 | FuriousGeorge | the wiki page is down but it looks like its for sip |
07:49.52 | salviadud | haha, hotlines are funny too |
07:50.15 | *** join/#asterisk vgster (n=vg@host217-45-221-53.in-addr.btopenworld.com) |
07:50.31 | brookshire | what's a hotline? |
07:50.34 | brookshire | haha.. just kidding |
07:50.37 | IronHelix | hehe |
07:54.19 | glm2k | brookshire: what's a batline? hehe |
07:54.38 | brookshire | we need one of those at work |
07:54.39 | *** join/#asterisk pengyong (n=lala@222.188.134.10) |
07:55.10 | IronHelix | setup a red phone on the wall |
07:55.21 | IronHelix | att slimline style if you can find one |
07:55.28 | IronHelix | put a removable plastic cover over it |
07:55.49 | IronHelix | then explicitly tell everybody to not ever, ever pick up or use that phone, no matter what |
07:55.56 | brookshire | cortelco! |
07:56.12 | FuriousGeorge | exten => s,1,dial(${POTS_OUT}/${COMISSIONER_GORDON}) |
07:56.18 | brookshire | lol.. yeah.. that would be greaet |
07:56.25 | brookshire | i wonder if i can find one of those on ebay |
07:57.05 | IronHelix | then whip up some automation stuff so it dials a script as soon as it goes off hook; which turns off all the lights in the immediate area, and intercom's every phone in the office with a RED ALERT type message |
07:57.17 | glm2k | lol |
07:57.18 | FuriousGeorge | i use a slimline mounted on my front foor to replace the functionality of the doorbell my landlord wont fix |
07:57.23 | IronHelix | as well as monitor()ing everything for future amusement |
07:57.56 | FuriousGeorge | batphone mode, of course |
07:58.15 | trixter | but does it ring to commisioner gordon? |
07:58.23 | *** join/#asterisk astar` (n=astar@ANantes-154-1-27-237.w81-53.abo.wanadoo.fr) |
07:58.37 | FuriousGeorge | no, it actually calls the local cluck-u chicken |
07:58.52 | FuriousGeorge | which really confuses them when they deliver |
07:58.53 | IronHelix | (then if you get bored, call the red phone, which will be wired to flash brightly when it rings)... immediately fire whoever answers it |
08:00.26 | sl16 | Feb 22 10:05:46 NOTICE[3952]: chan_sip.c:3593 process_sdp: No compatible codecs! |
08:00.29 | sl16 | how to fix this |
08:00.35 | Qwell | sl16: use compatible codecs |
08:00.49 | sl16 | i have those 2 licenses |
08:00.53 | sl16 | of g729 |
08:01.10 | sl16 | and the other side also has g729 NetCentrix or sort of |
08:02.16 | sl16 | i also get: SIP/2.0 488 Not acceptable here , when trying incoming calls |
08:02.55 | Qwell | yes...ypu arem |
08:03.00 | Qwell | 't using the same codecs they are |
08:03.06 | Qwell | need to allow the right ones |
08:03.29 | sl16 | i am only allowing g729 which is the one my VoIP provider working with |
08:03.52 | sl16 | 0/0 encoders/decoders of 2 licensed channels are currently in use |
08:03.56 | sl16 | the module is loaded |
08:04.04 | sl16 | i don't understand ... |
08:09.59 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
08:11.02 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
08:11.34 | X-Rob | sl16, use 'allow=all' |
08:11.44 | X-Rob | you've got ever codec, let them pick which ever one they want |
08:11.48 | X-Rob | every |
08:12.38 | sl16 | ok, just a second |
08:13.55 | sl16 | no compatible again |
08:14.19 | sl16 | the interesting thing i that a phone behind * can do the calls |
08:14.19 | X-Rob | then you didn't reload sip, or they are broken. |
08:14.29 | sl16 | but * can't |
08:14.35 | sl16 | i've reload it |
08:14.45 | sl16 | what is "broken" mean ? |
08:14.45 | X-Rob | sip debug |
08:14.48 | X-Rob | post to pastebin |
08:14.52 | X-Rob | pastebin.ca |
08:17.33 | *** part/#asterisk anandbabu (i=ab@69-12-132-138.dsl.static.sonic.net) |
08:17.52 | camonz | i'm having a prob with voicemail conf |
08:18.08 | camonz | i'm creating a voicemail account like this : 1002 => 0000,Simon,camonz@localhost,attach=no |
08:18.19 | sl16 | http://pastebin.ca/42746 |
08:18.22 | sl16 | X-Rob: |
08:18.31 | camonz | but when entering voicemail main and entering the pw the authentication fails |
08:19.16 | X-Rob | Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263) |
08:19.23 | camonz | except that when testing the example account 1234 it works allright |
08:19.25 | X-Rob | try 'show translations' |
08:19.51 | Abydos313 | hi everyone |
08:20.02 | {zombie} | camonz: did you "reload" asterisk after making changes to voicemail.conf? |
08:20.05 | camonz | yep |
08:20.16 | camonz | several times |
08:20.27 | Abydos313 | looking for some help on equipment choice |
08:20.49 | camonz | do i have to create a special directory for the mailbox for it to work? |
08:20.51 | {zombie} | and are you creating it in the [default] context in voicemail.conf? not the [other] context it creates? |
08:21.00 | camonz | yep |
08:21.15 | sl16 | X-Rob: http://pastebin.ca/42747 |
08:21.17 | {zombie} | asterisk creates the directories itself |
08:21.19 | camonz | actually i first had it in other context and then i put it on the default context |
08:21.45 | Juggie | camonz, pastebin.ca your voicemail.conf |
08:22.32 | camonz | thanks!, had the same login in the 2 contexts |
08:22.49 | camonz | removed it from the non default context and it worked |
08:23.03 | {zombie} | cool |
08:23.26 | Juggie | sl16, your problem is simple |
08:23.30 | camonz | i also want to dial an extension for 15 secs and then if no answer leave a voicemail |
08:23.32 | Juggie | your phones are set to g729 |
08:23.40 | Juggie | thats why they can talk via passthrough |
08:23.45 | sl16 | X-Rob: i've put allow=all at the beginning sip.conf |
08:23.53 | sl16 | and i have more capabilities |
08:23.56 | Juggie | but asterisk doesnt have a g729 codec |
08:24.03 | Juggie | allow=all = bad |
08:24.05 | Juggie | does wacky things |
08:24.18 | sl16 | ah. ok |
08:24.35 | Juggie | but your not listning |
08:24.42 | Juggie | your phone is set to g729 only |
08:24.47 | sl16 | Juggie: the phones are not set to g729 only ... |
08:24.52 | Juggie | asterisk does not understand g729 by default |
08:25.06 | camonz | http://pastebin.ca/42748 |
08:25.23 | sl16 | Juggie: 0/0 encoders/decoders of 2 licensed channels are currently in use |
08:25.35 | camonz | is this the way to do it?, i'm using Dial(sip/${EXTEN},15,j) |
08:25.53 | camonz | hoping it will fall to n+101 |
08:26.10 | Juggie | well, sl16, asterisk is presenting |
08:26.22 | Juggie | (gsm|ulaw|alaw|h263) |
08:26.37 | Juggie | and the phone is presenting |
08:26.38 | Juggie | peer - audio=0x100 (g729) |
08:26.50 | Juggie | no match |
08:28.49 | Juggie | try this :) |
08:28.50 | Juggie | sip.conf |
08:28.52 | Juggie | disallow=all |
08:28.54 | Juggie | allow=ulaw |
08:28.57 | Juggie | allow=g729 |
08:29.02 | Juggie | in [general] |
08:29.22 | Juggie | then try again |
08:29.58 | sl16 | Juggie: ok, tnx, just a second |
08:31.40 | *** join/#asterisk WasPhantom (n=neil@203-86-197-11-lightning.thepacific.net) |
08:31.42 | *** join/#asterisk k3y (n=avati@59.92.149.246) |
08:31.46 | sl16 | Juggie: http://pastebin.ca/42749 |
08:32.42 | k3y | i have a problem, i have an iaxy (digium) hardware behind a NAT talking to a remote asterisk server, when the DSL reboots and gets new IP, the digium stops working |
08:32.56 | Juggie | sl16, well now this problem is clearly new |
08:33.03 | Juggie | not codec related |
08:33.41 | Juggie | theres no match in whatever context is associated with that peer |
08:33.47 | Juggie | for the number its trying to dial |
08:33.50 | Juggie | hence the 404 |
08:34.43 | Juggie | theres probally some other output to the console to tell you exactally what context it was looking in |
08:35.38 | Juggie | none the less, its a simple context matching problem |
08:36.18 | Juggie | hurry |
08:36.22 | Juggie | because i'm going to sleep |
08:36.31 | sl16 | could it work putting all in one context |
08:36.33 | sl16 | default |
08:36.35 | sl16 | ?? |
08:36.50 | sl16 | Juggie: ok, sorry |
08:38.18 | Juggie | sl16, it can but for organization purposes that would not be reccomended |
08:38.20 | Juggie | none the less |
08:38.27 | Juggie | if its default |
08:38.50 | Juggie | you need to match the incomming number |
08:38.53 | Juggie | To: <sip:4912332@193.68.217.37> |
08:39.01 | Juggie | so the incomming number is 4912332 |
08:39.02 | *** join/#asterisk jaike (n=a@203.131.137.76) |
08:39.21 | Juggie | so you could do exten=>4912332,1,Dial(SIP/somesiphone) |
08:39.22 | Juggie | OR |
08:39.32 | Juggie | exten=>491233X |
08:39.36 | Juggie | and so on |
08:39.38 | Juggie | whatever |
08:39.51 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
08:39.53 | Juggie | i would recommend keeping things seperated |
08:39.54 | sl16 | could i in register say : user:PASS@provder/ext |
08:40.02 | brookshire | k3y: email digium support support@digium.com |
08:40.11 | brookshire | k3y: sounds like a configuration problem really |
08:40.20 | *** join/#asterisk anandbabu (i=ab@69-12-132-138.dsl.static.sonic.net) |
08:40.28 | k3y | brookshire when my nat session expires, it continues to work |
08:40.46 | Juggie | sl16, the register doesnt determine the context in which the call is received |
08:40.51 | Juggie | thats determined in the peer configuration |
08:41.14 | Juggie | look on the wiki and read the docs on sip.conf |
08:41.32 | Juggie | regardless for the purpose of making it work for the moment, add a pattern match into default for the number dialed |
08:41.38 | Juggie | you can learn more later :) |
08:41.39 | sl16 | it worked |
08:41.41 | sl16 | :))))))))))) |
08:41.44 | sl16 | thank youuuuuuuuuuuuu |
08:41.59 | Juggie | idealy the best thing is that incomming sip calls, and iax calls and calls from your TDM/zap interface all go into seperate contexts |
08:42.00 | Juggie | eg |
08:42.02 | brookshire | oh yeah.. iaxy doesn't support dns |
08:42.07 | Juggie | you should have sip-in |
08:42.10 | brookshire | so.. |
08:42.11 | Juggie | and iax-in |
08:42.13 | Juggie | and zap-in |
08:42.14 | Juggie | and so on |
08:42.18 | Juggie | to keep everything seperated |
08:42.39 | Juggie | www.voip-info.org |
08:42.45 | Juggie | lots of info, now i must sleep, work in 5hrs |
08:42.52 | Juggie | and i have to drop my skis to the shop on the way to work |
08:42.55 | Juggie | so i have to get up early |
08:45.35 | k3y | brookshire server address is static... iaxy is behind NAT... the NAT ip changes |
08:47.20 | brookshire | hmm.. dunno |
08:47.27 | *** join/#asterisk miller7 (n=none@gige-2.office-nl.irismedia.gr) |
08:47.44 | *** part/#asterisk miller7 (n=none@gige-2.office-nl.irismedia.gr) |
08:47.48 | Juggie | k3y, set qualify=yes |
08:47.51 | Juggie | for the iax peer |
08:48.20 | Juggie | and if you can on the iaxy set the registration time low |
08:48.26 | Juggie | i dont know about those devices |
08:49.54 | brookshire | i don't think you can set the registration time on the iaxy |
08:50.16 | trixter | it all depends on skill level |
08:50.20 | trixter | and time |
08:50.22 | Juggie | i dont see it in iax.conf either |
08:50.35 | Juggie | but you can set qualify=yes |
08:50.35 | k3y | Juggie my iaxy device sends packets from src port 4569 |
08:50.38 | k3y | then it works |
08:50.51 | k3y | but whan IP changes, it starts going with src port 1024 |
08:51.00 | k3y | that what my NAT (linux, netfilter) does |
08:51.08 | k3y | i'm suspecting its the port problem |
08:51.12 | Juggie | k3y, the iaxy source port isnt changing |
08:51.13 | anandbabu | k3y, i have set qialify=yes now in iax.conf |
08:51.17 | Juggie | netfilter is writing |
08:51.24 | k3y | Juggie yes, that's what i meant |
08:51.24 | Juggie | er, rewriting it |
08:51.33 | jaike | iax2 problems with nat? |
08:51.44 | Juggie | nat w/ changing ip |
08:51.51 | Juggie | what kinda isp changes its ip anyways |
08:52.02 | Juggie | k3y, did you set qualify=yes |
08:52.09 | brookshire | bastard ones :( |
08:52.13 | Juggie | with that * sends contant keep allive packets |
08:52.16 | anandbabu | Juggie, will qualifysmoothing = yes also help? |
08:52.23 | Juggie | no |
08:52.24 | k3y | Juggie dsl |
08:52.28 | jaike | isp with a super paranoid network administrator |
08:52.30 | anandbabu | Juggie, i have access to k3y's server. configuring it |
08:52.38 | Juggie | just qualify=yes |
08:52.43 | Juggie | that will have * send packets |
08:52.54 | Juggie | which has the effect of A) keep nat connection open |
08:53.02 | Juggie | B) will notice when the iaxy isnt responding |
08:53.21 | anandbabu | k3y, done, restarted asterisk. try now |
08:53.43 | Juggie | anandbabu, type 'iax2 show peers' |
08:53.49 | Juggie | to see the status of his iaxy |
08:54.56 | kippi | hey |
08:55.04 | k3y | currently it is sending with src port 4569 as the udp session expired |
08:55.05 | k3y | hangon |
08:55.13 | anandbabu | Juggie, bharathi/bharat 59.92.149.246 (D) 255.255.255.255 4569 Unmonitored |
08:55.14 | k3y | its come online |
08:55.33 | kippi | can anyone help me with this error? Feb 22 08:54:47 NOTICE[15770]: chan_iax2.c:3918 register_verify: No registration for peer '1001' (from 10.6.10.149) |
08:55.55 | brookshire | is that fwd? |
08:56.28 | brookshire | nm... lol |
08:56.33 | Juggie | anandbabu, qualify=yes should make it monitored |
08:56.47 | anandbabu | Juggie, thanks i have set it now |
08:56.58 | *** join/#asterisk saftsack (n=saftsack@p54A7EBE7.dip.t-dialin.net) |
08:57.14 | saftsack | are some germans here? |
08:58.53 | *** join/#asterisk welles (n=welles@61.150.12.136) |
09:01.49 | *** join/#asterisk maruz (n=maumar@adsl-123-3.38-151.net24.it) |
09:03.01 | maruz | flexibel rate not heavily tested! i get this error, i modified my mp3 but still i have it, can someone point me to a source of mp3 certified asterisk? |
09:05.45 | jaike | maruz: have you tried removing mp3 ID info? |
09:06.32 | maruz | i have no skill on mp3 and i dunno which sw to use to do it; someonehere tried in win to change rate of mp3 |
09:07.10 | maruz | so i thought that the fastest solution is download by some url an mp3 that is good for asterisk |
09:07.43 | maruz | but in www.asterisk.org there is any mp3 suitable for moh? |
09:08.06 | jaike | 1.2.4 comes with 3 mp3s for MOH |
09:08.07 | *** join/#asterisk marktt (n=marktt@203.217.18.2) |
09:08.23 | marktt | ring.ring |
09:08.34 | jaike | fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3 |
09:09.11 | maruz | -r--r--r-- 1 root src 1939812 2006-02-07 18:35 ./sounds/fpm-calm-river.mp3 |
09:09.11 | maruz | -r--r--r-- 1 root src 2217563 2006-02-07 18:35 ./sounds/fpm-world-mix.mp3 |
09:09.11 | maruz | -r--r--r-- 1 root src 2582496 2006-02-07 18:35 ./sounds/fpm-sunshine.mp3 |
09:09.21 | maruz | are them? |
09:09.25 | jaike | yes |
09:09.27 | brookshire | maruz: http://www.freeplaymusic.com/ |
09:09.31 | maruz | the size is right? |
09:09.40 | maruz | bronze: let's try :) |
09:09.45 | anandbabu | music from this link worked for mehttp://www.sounddogs.com/catsearch.asp?Type=2 |
09:09.48 | jaike | yes |
09:10.26 | anandbabu | also after replacing mpg321 with mpg123 in my debian box, all mp3s played well |
09:10.59 | *** join/#asterisk corruptor (n=andrew55@www.tae.ru) |
09:11.08 | jaike | maruz: mpg123 installed? |
09:11.40 | anandbabu | debian sym links mpg321 as mpg123 - be aware |
09:11.44 | maruz | yes, i removed soft links to mpg321 |
09:11.54 | maruz | in /etc/aslternatives |
09:11.59 | maruz | and so on |
09:12.00 | maruz | :) |
09:16.43 | *** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.190) |
09:16.48 | Kernel_core | hi all |
09:19.42 | Kernel_core | anybody used asterisk 1.2.4 with h323 module on debian ? |
09:21.09 | justnulling2 | how can i send user to a specific folder in vm (so that msg is not saved in a defualt folder)? |
09:23.02 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
09:23.36 | x86 | i'm trying to setup my softphone to connect to my asterisk server, but it doesnt appear to even try to register |
09:23.59 | x86 | i have sip debug on in my asterisk console, but i'm not seeing anything |
09:25.06 | x86 | any ideas? |
09:25.14 | x86 | the asterisk box and the softphone are on the same box |
09:25.18 | ChrisUK | software firewall perhaps? |
09:25.24 | ChrisUK | oh |
09:26.14 | x86 | i just flushed the firewall just to make sure |
09:27.03 | dpryo | Is there any softphones without stupid sci-fi GUI? |
09:28.31 | x86 | hmm |
09:28.39 | jaike | maybe the softphone cant register to its own ip |
09:28.46 | x86 | asterisk automagically acts as a sip proxy right? |
09:29.59 | x86 | also, i'm trying to IAX2 trunk to freeworlddialup |
09:30.05 | x86 | my asterisk server is behind a NAT |
09:30.07 | x86 | is this possible? |
09:31.08 | *** join/#asterisk Sloboda (n=slob@194.42.196.254) |
09:31.35 | Sloboda | Hi! I'm looking for soft-phone for Linux that supports usb-phones. |
09:33.35 | trixter | ok |
09:33.36 | camonz | maruz did you compiled ztdummy? |
09:33.39 | trixter | good to know |
09:33.44 | camonz | you can do it without MP3Player |
09:33.48 | maruz | camonz: well question :) |
09:34.02 | maruz | camonz: i am having many troble about that |
09:34.11 | jaike | yup..1.2 has native MOH player |
09:34.30 | camonz | maruz: how so? |
09:34.46 | trixter | asterisk-addons format_mp3 |
09:34.54 | maruz | yes, taht i have installed |
09:35.06 | trixter | just use the native moh then :) |
09:35.15 | trixter | its a lot better and more friendly to your system |
09:35.21 | camonz | it's easier to use native MoH |
09:35.30 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
09:35.32 | maruz | but i stil miss timing source, i have visdn driver and i have posted on mialing list |
09:35.38 | camonz | just uncomment the # ztdummy line in the makefile |
09:35.43 | camonz | for zaptel |
09:35.53 | camonz | remove the # |
09:36.00 | maruz | sure? i just do it now |
09:36.07 | jaike | moh needs timing source? though that was for meetme |
09:36.10 | camonz | then make clean && make && make install |
09:36.11 | jaike | thought |
09:36.29 | camonz | also for MoH, i just resolved that issue on sunday thanks to [av]bani |
09:36.43 | maruz | Feb 20 11:08:13 WARNING[3486] res_musiconhold.c: Unable to open pseudo channel for timing... Sound may be choppy. |
09:36.48 | vgster | is busydetectmartin much better than regular busydetect? |
09:36.59 | jaike | really? good thing i got ztdummythingny compiled too |
09:37.00 | camonz | yep that's because you don't have ztdummy.so module loaded on kernel |
09:37.01 | maruz | yes, for moh too ... |
09:37.57 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
09:38.07 | camonz | after you've made and installed zaptel do a modprobe ztdummy |
09:38.09 | jaike | wish mixmonitor bugs would be fixed soon too. shell+soxmix really eats up a lot of resources |
09:38.17 | camonz | that should do it |
09:38.37 | camonz | oh..., in musiconhold.conf in the default context set mode=files |
09:38.51 | camonz | then reload and you're all set |
09:39.02 | jaike | maruz: using kernel 2.6? |
09:39.36 | maruz | jaike yes |
09:39.39 | maruz | 2.6.15 |
09:39.49 | maruz | by ftp.kernel.org |
09:39.54 | maruz | not that of distro |
09:39.54 | jaike | ztdummy should load with no problems |
09:40.33 | maruz | MODULES:=zaptel tor2 torisa wcusb wcfxo wctdm wctdm24xxp \ |
09:40.33 | maruz | <PROTECTED> |
09:40.34 | maruz | <PROTECTED> |
09:40.36 | maruz | eh eh eh |
09:40.39 | maruz | here it is |
09:40.41 | camonz | yep |
09:40.43 | camonz | remove the # |
09:40.44 | maruz | great! |
09:41.10 | *** join/#asterisk jorgito (n=jorge@82.113.32.241) |
09:41.23 | jorgito | cant uninstal asterisk , neither upgrade . bleee tfuj shit soft |
09:41.34 | maruz | maybe i can fix flexible rate error, too ;) |
09:42.00 | pcm | hey there |
09:42.07 | camonz | i had a similar issue, the makefile checks for a shell variable called KVERS |
09:42.14 | pcm | i need to sell a couple of tormenta2 quad t1/e1 cards, anyone interested ? |
09:42.18 | camonz | wich should be linux26 for 2.6 kernel ver |
09:42.32 | prh | KVERS is set by make-kpkg (the debian kernel build system) |
09:42.46 | camonz | i don't have that var on my shell, so it defaulted to kernel 2.4 |
09:42.50 | camonz | i'm running SuSe |
09:43.01 | prh | and should be the full kernel version to compile against - ie 2.6.15-rc2-localwibble-rev1 or whatever |
09:43.06 | prh | ooh |
09:43.13 | camonz | should KVERS get a value? |
09:43.49 | camonz | that's why the makefile for zaptel didn't compile ztdummy when the # ztdummy line was commented |
09:45.17 | *** join/#asterisk Mavvie (n=edwin@252-131-222-203.rev.techex.net.au) |
09:45.22 | jorgito | have proglems with asterisk Unable to connect ro remote asterisk (does /var/run/asterisk.ctl extist?0 |
09:45.25 | camonz | actually the KVERS var is the result of uname -r wich on my system turns out to be 2.6.11.4-21.9-default |
09:45.36 | camonz | jorgito: are you running as root? |
09:45.41 | jorgito | yes |
09:45.49 | jaike | jorgito: asterisk is not running |
09:45.58 | camonz | yep |
09:46.13 | jaike | asterisk -vc |
09:46.13 | camonz | did you do asterisk -vvvc before going asterisk -r? |
09:46.25 | jorgito | i did asterisk -p |
09:47.12 | *** join/#asterisk froguz (n=froguz@188-142-222-201.adsl.terra.cl) |
09:47.14 | camonz | hmm |
09:47.18 | jorgito | Asterisk Dynamic Loader Starting: |
09:47.18 | jorgito | <PROTECTED> |
09:47.18 | jorgito | Feb 22 10:40:42 WARNING[26339]: loader.c:499 load_modules: Loading module app_system.so failed! |
09:47.24 | *** join/#asterisk ambriento (n=ambrient@www.cobranet.com.br) |
09:48.25 | pcm | do ldconfig |
09:48.53 | vgster | does anyone know if busydetectmartin is much better than regular busydetect? |
09:49.09 | pcm | vgster: it was I believe |
09:49.21 | pcm | vgster: but recently it was improved I think ... funny that you ask |
09:49.39 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
09:50.06 | *** join/#asterisk moreece (n=m@196.46.142.23) |
09:50.17 | vgster | ok, i need to comment the original line out but do i need to remove the + from the BUSYDETECT+= #-DBUSYDETECT_MARTIN |
09:50.34 | vgster | in order to use it? |
09:50.35 | maruz | camonz: jaike : Notice: Configuration file is /etc/zaptel.conf |
09:50.35 | maruz | line 0: Unable to open master device '/dev/zap/ctl' |
09:50.52 | *** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.190) |
09:50.52 | vgster | have you made any udev changes? |
09:50.56 | jaike | udev |
09:50.57 | maruz | i have no zaptel hw, so i think there no problem out there |
09:51.08 | maruz | or not? |
09:51.27 | pcm | vgster: comment the outher one and uncomment the 'MARTIN' one by removeing '#' |
09:51.30 | camonz | there shouldn't be, it worked really well for me |
09:51.37 | maruz | camonz: ok :) |
09:52.01 | camonz | i'm a newb at * and linux too sorry i cannot be of much help :-> |
09:53.04 | pcm | anyone need 4xe1 or 4xt1 card ? I need to sell a few |
09:53.53 | vgster | pcm: thanks |
09:54.03 | jaike | maruz: go through README.udev inside your zaptel folder |
09:54.17 | pcm | vgster: np |
09:54.50 | Mavvie | if it just me, or is the editline in asterisk configured as emacs instead of as vi? |
09:54.56 | vgster | next question, :D how many channels/ how small a cache in order to use CONFIG_CALC_XLAW in zaptel? |
09:55.11 | jaike | didnt need to do that though, since with Fedora you just do 'make config', which creates a zaptel init file |
09:55.52 | vgster | fedora used to take a few seconds to make all the devs on my system |
09:57.22 | jorgito | shit asterisk Makefile is a mess |
09:57.44 | jaike | i think it does need some cleanup |
09:58.34 | jorgito | no doesnt have uninstall |
09:58.53 | jaike | maruz: if this is becoming too complex just for MOH, just use the mp3 that came with asterisk |
09:58.58 | jaike | ang mpg123 |
09:59.20 | jaike | i on the other hand really needed ztdummy for meetme |
09:59.23 | camonz | just delete everything in var/lib/asterisk , usr/lib/asterisk /etc/asterisk |
09:59.32 | camonz | and where you keep the executable |
09:59.36 | trixter | app_conference doesnt require a timer |
10:00.11 | jaike | hmm...will read on that |
10:00.21 | vgster | do you need to config the zaptel.conf file before you load ztdummy? |
10:00.22 | *** join/#asterisk lorinc (n=ang@caracas-3121.adsl.interware.hu) |
10:00.47 | camonz | vgster: i didn't had to |
10:00.51 | vgster | ok |
10:00.52 | jaike | vgster: i didnt need to..default conf worked fine |
10:00.53 | camonz | just modprobe ztdummy from bash |
10:01.28 | *** join/#asterisk sch19 (n=sch19@adsl-9-107-161.mia.bellsouth.net) |
10:02.15 | jaike | trixter: i dont think its with stable yet |
10:03.55 | trixter | dont think what is? |
10:05.02 | maruz | camonz: yes sir, i did it 2 times, the first time i got error, teh 2nd it got looaded |
10:05.06 | maruz | loaded |
10:05.25 | *** part/#asterisk anandbabu (i=ab@69-12-132-138.dsl.static.sonic.net) |
10:06.24 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
10:06.49 | *** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.190) |
10:07.30 | Delvar | ~ping |
10:07.31 | jbot | pong |
10:09.33 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
10:09.45 | vgster | does anyone know how many channels/ how small a cache in order to use CONFIG_CALC_XLAW in zaptel? |
10:17.12 | *** join/#asterisk Bambr (n=Bambr@213-35-238-17-dsl.end.estpak.ee) |
10:17.41 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
10:17.44 | *** join/#asterisk h3x (n=h3xor@64.192.116.16) |
10:23.53 | areski | good morning everybody ! |
10:24.18 | pcm | good |
10:24.24 | vgster | hello |
10:27.49 | *** join/#asterisk apardo (n=apardo@87.218.45.71) |
10:30.27 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
10:30.45 | jorgito | i have noload => chan_modem.so but still have problem [chan_modem.so]Feb 22 11:24:40 WARNING[26892]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_modem.so: cannot open shared object file: No such file or directory |
10:30.45 | jorgito | Feb 22 11:24:40 WARNING[26892]: loader.c:499 load_modules: Loading module chan_modem.so failed! |
10:33.38 | pcm | chan_modem exports some functions that are NEEDED |
10:33.44 | pcm | you shouldn't do noload for it |
10:35.06 | jorgito | well the problem is that i specified INSTALL_PREFIX?=/usr/local/asterisk |
10:35.43 | jorgito | when i run /usr/local/asterisk/usr/sbin/asterisk -vc , it took configuration from /etc |
10:35.48 | jorgito | /etc/asterisk |
10:36.44 | vgster | i didnt think chan_modem got built with 1.2 |
10:37.18 | pcm | then you need to maybe add /usr/local/asterisk |
10:37.21 | pcm | to /etc/ld.so.conf |
10:37.24 | pcm | and do ldconfig |
10:38.18 | jorgito | well i am not missing lib but configuration file |
10:40.36 | *** join/#asterisk fulgas (n=fulgas@82.102.2.254) |
10:47.25 | marktt | Can someone confirm that I can what seems plausable with asterisk.... |
10:48.16 | marktt | I have an ISP provided SIP 'address'.... which is probably asterisk with a card out to copper... the number of which they have assigned to me. |
10:48.51 | marktt | So I have a modest little network on a fixed IP where I have NAT running behind a firewall. |
10:49.14 | marktt | THere is a Cisco IP phone on this as well as softPhones. |
10:49.34 | marktt | Can I use my asterisk to proxy to the isp 'asterisk'? |
10:52.16 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
10:52.47 | *** join/#asterisk mut (n=animenod@65.111.201.79) |
10:53.31 | mut | <PROTECTED> |
10:53.35 | mut | what causes that? |
10:53.45 | mut | seems randomly all the bchannels restart |
10:54.29 | jaike | marktt: should be possible, if youre able to sort out sip-nat problems |
10:59.00 | camonz | i'm gonna go to sleep |
10:59.06 | *** join/#asterisk Vyeperman (n=Vye@ip68-8-174-154.sd.sd.cox.net) |
10:59.09 | camonz | cya |
10:59.13 | maruz | bye |
11:04.18 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
11:11.08 | Falle | Folks, are there any irclogs for this channel on the internet somewhere? |
11:11.43 | tzafrir | Falle, yes |
11:12.26 | tzafrir | http://www.asteriskgeeks.com/ |
11:12.30 | Falle | the only pages o found containing logs had the logpart of the page down. :S |
11:12.48 | *** join/#asterisk chrisvarns (i=chris@ACCB96E7.ipt.aol.com) |
11:12.52 | tzafrir | Falle, that's the link from the wiki |
11:13.20 | Falle | tzafrir: that one shows HTTP Error 404 when i click the loglink :) |
11:15.38 | *** join/#asterisk jwu (n=jwu@221.221.11.37) |
11:17.55 | *** join/#asterisk skeffling (n=chatzill@andrew.1ec.aaisp.net.uk) |
11:25.39 | jwu | <PROTECTED> |
11:25.48 | iDunno | yes, now infact. |
11:25.57 | iDunno | in 1 second, a second will have passed! |
11:26.10 | jwu | oh.. |
11:26.34 | jwu | idunno... you speak to me? |
11:29.11 | dpryo | ;P |
11:29.12 | jwu | idunno, can you help? |
11:37.34 | *** join/#asterisk Samoied (n=Samoied@201.14.162.82) |
11:42.14 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
11:45.04 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
11:47.48 | *** join/#asterisk Modcuts (n=info@proporta.gotadsl.co.uk) |
11:48.40 | RoyK | ~realtime |
11:48.41 | jbot | it has been said that realtime is http://www.voip-info.org/wiki-Asterisk+RealTime |
11:48.42 | RoyK | ~docs |
11:48.43 | jbot | rumour has it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
11:48.48 | RoyK | ~rtfm |
11:48.48 | jbot | it has been said that rtfm is Read The F*cking Manual (TM) |
11:48.50 | RoyK | oh |
11:48.52 | RoyK | he quit :P |
11:50.57 | RoyK | hmmmmmmm |
11:50.58 | RoyK | exten => s,n,Set(DIALOPTS=${IF($[ "${CCMAXSECS}" = "-1" ]?90:90\,S(${CCMAXSECS}))}) |
11:51.02 | RoyK | that doesn't work.... |
11:51.59 | tuxinator_linux | guess jwu didn't want an answer ;-) |
11:53.30 | *** part/#asterisk maruz (n=maumar@adsl-123-3.38-151.net24.it) |
11:54.45 | RoyK | can someone please help me out with this? the above Set(DIALOPTS.... just sets DIALOPTS to 90 |
11:54.46 | RoyK | <PROTECTED> |
11:54.56 | RoyK | even though I've escaped the , |
11:57.11 | thazza | jaike: Bugger thats a few hours of recorded phone calls. |
11:57.24 | jaike | thats in wav49 |
12:02.11 | Modcuts | with the callwaiting function *70 can i make it so that it does not ring on the line when a second call is coming in? |
12:05.04 | *** part/#asterisk Sloboda (n=slob@194.42.196.254) |
12:10.40 | *** join/#asterisk apardo (n=apardo@87.218.44.157) |
12:17.29 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
12:24.45 | enemy^x | exten => t,1,Queue(teknisk|tT|||0) |
12:24.54 | enemy^x | how can this actually go further to t,2,.... |
12:25.03 | enemy^x | when the timeout is set to 0? |
12:27.57 | jaike | 100gigs to go tomorow..am outta here |
12:28.01 | jaike | bye all |
12:28.05 | *** part/#asterisk jaike (n=a@203.131.137.76) |
12:29.46 | RoyK | enemy^x: #asterisk-no |
12:31.09 | RoyK | hmmmmmm |
12:31.20 | RoyK | seems the S flag in Dial doesn't work :( |
12:34.41 | *** join/#asterisk mountie (n=mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
12:36.02 | *** join/#asterisk Skarmeth (n=Skarmeth@201009039176.user.veloxzone.com.br) |
12:36.19 | remiss | w00t.. ata-box in da house :D |
12:43.10 | *** join/#asterisk fugitivo (n=ajf@201.255.177.92) |
12:43.15 | fugitivo | hello |
12:49.43 | saftsack | Feb 21 21:56:24 debian kernel: Power alarm on module 2, resetting! |
12:49.49 | saftsack | what is that? |
12:49.51 | *** join/#asterisk _Paulo_ (n=Paulo@200-168-112-132.dsl.telesp.net.br) |
12:50.06 | fugitivo | some problem with your power supply |
12:50.51 | saftsack | my asterisk server said this |
12:50.54 | *** join/#asterisk }btorch{ (n=kvirc@208.63.19.184) |
12:51.03 | saftsack | Feb 22 02:30:50 debian kernel: qozap: dropped audio card 1 cardid 255 bytes 8 z1 103 z2 79 |
12:51.17 | }btorch{ | is it possible for * |
12:51.18 | saftsack | this is a aftereffect of this or? |
12:52.03 | }btorch{ | has anyone here tried to setup asterisk to talk to a second PRI on a proprietary PBX box ? |
12:52.25 | fugitivo | }btorch{: lot of people did that |
12:52.35 | }btorch{ | and be able to make calls to the outside through the first PRI that is connected to the PSTN |
12:53.34 | }btorch{ | I got it working so that I can make calls to the internal 4 digit extension numbers that the PBX controls but I can't receive calls from those regular phones nor make calls out |
12:53.48 | x86 | heh, i managed to make the voicemail extension crash asterisk when called :) |
12:53.49 | }btorch{ | calls to the outside ... any howtos out there ? |
12:54.41 | fugitivo | }btorch{: what kind of cable did you use? |
12:54.42 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
12:55.08 | }btorch{ | I'm not siemens PBX specialist btw ... good on * though ... |
12:55.17 | }btorch{ | fugitivo: I'm using a crossover T1 |
12:55.57 | }btorch{ | * and the PBX can see each other but I guess rules or routes or something may have to be setup on the siemens !!! |
12:56.23 | fugitivo | }btorch{: i didn't make a setup like that yet, i have to do it this weekend :) |
12:56.39 | }btorch{ | good luck |
12:56.53 | *** join/#asterisk SomePBXUser (n=neil@96.Red-80-38-99.staticIP.rima-tde.net) |
12:57.12 | fugitivo | thanks |
12:57.22 | }btorch{ | I'm already thinking about getting a nother TE110P card for my asterisk box and place between the PSTN and the PBX so that I can save all this headache |
12:57.57 | }btorch{ | It would be nice if I could make it work with e&m but I can't |
12:57.57 | fugitivo | what do you get when calling from the pbx to asterisk? |
12:57.58 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
12:57.59 | fugitivo | busy tone? |
12:58.37 | luke-jr_ | How well-tested is the AEL stuff? |
12:58.43 | }btorch{ | voicemail |
12:58.55 | *** join/#asterisk pengyong (n=lala@218.93.154.172) |
12:58.58 | }btorch{ | pbx voicemail not * |
12:59.32 | *** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com) |
12:59.34 | fugitivo | }btorch{: so it never reaches asterisk? |
12:59.36 | luke-jr_ | * is a pbx |
12:59.39 | }btorch{ | no |
12:59.42 | luke-jr_ | ;) |
12:59.57 | Egonis | I have load => pbx_gtkconsole.so in modules.conf, but don't get a gtk console, am I missing something? |
13:00.13 | }btorch{ | I think it's because the siemens sees the call come in but doesn't know where to route too |
13:00.17 | luke-jr_ | Egonis: check the commandline options |
13:00.22 | fugitivo | luke-jr_: well, it's hard to talk that way, it's better * and pbx :) |
13:00.32 | Egonis | luke-jr_: i.e. asterisk --help? |
13:00.37 | fugitivo | }btorch{: what model of siemens? |
13:00.39 | luke-jr_ | Egonis: sure |
13:00.53 | Egonis | luke-jr_: lol, okay.. ty! |
13:01.10 | }btorch{ | 150E officepro |
13:01.14 | luke-jr_ | so has anyone here used AEL yet? |
13:01.25 | RoyK | erm |
13:01.37 | RoyK | anyone here tried the S(timeout) flag with dial? |
13:03.06 | fugitivo | }btorch{: i think the problem is with the siemens config, but i know nothing about siemens pbx |
13:03.08 | *** join/#asterisk lunaphyte_ (n=lunaphye@c-71-193-101-146.hsd1.mi.comcast.net) |
13:04.40 | }btorch{ | fugitivo: yeah I thought me niether and the guys who give us support to the siemens just keeps saying it can't be done |
13:04.51 | fugitivo | why not? |
13:05.01 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
13:05.51 | }btorch{ | fugitivo: they say that the siemens can't route calls from on PRI to another |
13:06.15 | *** join/#asterisk Garak_ (i=1000@209.5.171.170) |
13:06.16 | fugitivo | that sucks if it's true |
13:06.37 | }btorch{ | like for example a number comes in from the PSTN to PRI1 and send the call through PRI2 so that * can get it |
13:07.29 | Garak_ | Is it possible to make a call from the console and have it play a few recordings to who ever answers |
13:07.29 | }btorch{ | in order for that to work they told me that had to setu pthe card as an analog T1 using e&m wink but when they did that I kept getting the red alarm on the Te110p |
13:07.34 | Egonis | I have built asterisk w/ gtk, but cannot find how to open the gtk console |
13:07.37 | Egonis | I am using gentoo |
13:07.55 | }btorch{ | anyway got go setup a sonicwall now |
13:08.14 | saftsack | fugitivo, do you really think, that this is a psu issue? |
13:08.48 | *** join/#asterisk stse (n=stse@muedsl-82-207-237-090.citykom.de) |
13:08.59 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
13:09.00 | RoyK | bummer |
13:09.10 | RoyK | S(timeo) flag surely doesn't work |
13:09.53 | fugitivo | saftsack: maybe, didn't see that message in my whole life (11 years of linux) |
13:11.39 | RoyK | it's bill gates fault, all of it |
13:11.41 | saftsack | ok. what is module2? is this the second module which i can see in lsmod? |
13:13.19 | *** join/#asterisk f7950qs0 (n=redhotte@61.17.213.99) |
13:13.31 | f7950qs0 | hi |
13:13.34 | f7950qs0 | anybody talking |
13:13.38 | f7950qs0 | yea yea I know it would be best to ask |
13:13.44 | f7950qs0 | I am using A@H |
13:13.56 | f7950qs0 | and have configured a trunk a route and an extension successfully |
13:14.14 | fugitivo | ~amp |
13:14.14 | jbot | extra, extra, read all about it, amp is NOT supported here! people using it should join #amportal |
13:14.34 | f7950qs0 | what is amportal? |
13:14.48 | fugitivo | the web interface of a@h |
13:15.10 | f7950qs0 | oh yea I forgot |
13:15.11 | f7950qs0 | hehehe |
13:16.02 | stse | Hi, I need some help with Asterisk 1.2.4 (Bristuff) and Snome phones (320/360). |
13:16.33 | fugitivo | stse: what's the problem? |
13:16.35 | stse | I can't get the LEDs to show the ringing/used state. |
13:17.16 | stse | sip show subscriptions shows the subscriptions, show hints shows the correct hints/watchers, but asterisk doesn't send notify messages. |
13:17.29 | TheCops | stse, It was working before and not anymore with the new 1.2 ? |
13:17.44 | TheCops | I that problem here |
13:17.49 | *** join/#asterisk backblue (n=igor@87-196-36-85.net.novis.pt) |
13:17.51 | TheCops | I have that problem here |
13:18.11 | stse | TheCops: It never worked with 1.2.x. I didn't have 1.0.x |
13:18.14 | backblue | hi, ppl, why use tdm and not iax2 trunk?? which its better? |
13:20.51 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
13:21.12 | Garak_ | How do I get asterisk to make a call and play a recording when someone answers it |
13:21.53 | remiss | answer(), playback(recording) |
13:22.04 | stse | If I understand correctly, Asterisk should send notify messages to the subscriber, but if I trace with ethereal, asterisk doesn't send any messages. |
13:22.16 | Garak_ | remiss: yea but where, its not an exten... |
13:22.30 | remiss | Garak_: it is... |
13:22.41 | remiss | oh.. make a call |
13:22.51 | remiss | still.. create a context with it |
13:23.14 | remiss | hmm.. good question.. |
13:23.39 | Garak_ | their is no onanswer => |
13:23.43 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
13:24.43 | Garak_ | Has anyone ever implemnted a collect call system with asterisk? |
13:25.11 | f7950qs0 | what is the best way to learn dialling rules in asterisk? |
13:25.24 | remiss | LIMIT_CONNECT_FILE=filename |
13:25.25 | remiss | Specifies which file to play when call begins |
13:25.27 | tuxinator_linux | Has anyone figured out the appeal to the Olympic game, curling? |
13:25.36 | remiss | as an option to dial |
13:26.21 | Modcuts | tuxinator_linux:to entertain old people |
13:26.41 | f7950qs0 | I have the asterisk book |
13:26.56 | Garak_ | tuxinator_linux: try playing it, then you will understand |
13:26.56 | fugitivo | f7950qs0: editing the files and not using web interfaces |
13:26.59 | f7950qs0 | I need a good reference to asterisk |
13:27.23 | fugitivo | ~docs |
13:27.24 | jbot | well, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
13:27.54 | RoyK | ~wiki |
13:28.07 | RoyK | ~wiki asterisk |
13:28.59 | f7950qs0 | thanks jobt |
13:29.01 | f7950qs0 | jbot |
13:29.28 | Egonis | I'm using asterisk 1.0.10, although I have gtk_console.so set to load in modules, nothing shows... I can't find a howto anywhere on google. Can someone help me? |
13:32.57 | *** join/#asterisk _deg_ (n=deg@200-233-51-145.corp.ajato.com.br) |
13:33.11 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
13:34.25 | Modcuts | how do you make sure that a sip extension is unregistered so voicemail goes to offline not busy? |
13:34.30 | *** join/#asterisk coppice (n=chatzill@62.166.17.210.dyn.pacific.net.hk) |
13:35.10 | fugitivo | Modcuts: DIALSTATUS |
13:35.30 | Modcuts | ah i see and how do i set that? |
13:35.49 | fugitivo | when you dial you have the status of the call in ${DIALSTATUS} |
13:36.07 | vgster | Egnois have you tried starting asterisk from x? |
13:37.46 | *** join/#asterisk laichzeit (n=ahuman@dsl-145-185-135.telkomadsl.co.za) |
13:38.10 | *** part/#asterisk laichzeit (n=ahuman@dsl-145-185-135.telkomadsl.co.za) |
13:38.20 | Modcuts | fugitivo: and how can i set the dial status of all extensions to be offline when a call comes in after a certain time say with gotif? |
13:39.22 | fugitivo | Modcuts: what do you need to do? |
13:43.13 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
13:44.26 | f7950qs0 | bye all |
13:44.35 | f7950qs0 | i will read the asteriskbook and come back |
13:47.33 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
13:48.26 | *** join/#asterisk trelane_ (n=trelane@209.43.90.13) |
13:51.09 | *** join/#asterisk p0g0_ (n=pogo@madwifi/support/p0g0) |
13:51.17 | *** join/#asterisk oej (n=oej@swissco012231-3-3.clients.easynet.fr) |
13:51.30 | *** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net) |
13:53.59 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
13:56.50 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
13:58.55 | Modcuts | fugitivo: Well we have VM setup for all the exten users, now when we are not in the office the extenions setup with VM to record messages goes to the busy message not the offline message.? |
14:01.17 | Garak_ | With IAX2, can you setup an outgoing connection to another asterisk box and then accpect calls back through that link, i.e. for asterisk machines behind nat where forwarding ports is not an option |
14:02.03 | RoyK | register => |
14:02.47 | Garak_ | so that keeps a connection open? |
14:03.32 | stse | No one here with a hint for my SNOM/Asterisk/Notify problem? |
14:06.04 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
14:11.31 | shmaltz | why would I get clicks (like a DTMF tone) in middle of a conversation, using Zap to Zap on the same single span T1 Digium card, but it's NOT internaly bridged. |
14:15.15 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
14:16.31 | luke-jr_ | Is there any documentation/howto for moving an Asterisk server to a different system? |
14:16.49 | luke-jr_ | eg, obviously move the config files and voicemail directories... what else? |
14:16.52 | mitcheloc | just copy your config and voicemail/sound files |
14:17.06 | mitcheloc | maybe your cdr records (cdr-csv) if you want that |
14:17.10 | luke-jr_ | ah |
14:18.15 | shmaltz | luke-jr, if you were using dhcpd, and it's going to be on the same network, then you might want to copy dhcpd.conf, as well as the leases |
14:18.40 | shmaltz | luke-jr, if you did any mass provisioning using http/ftp/tftp then move those as well |
14:18.45 | mitcheloc | heh what do you need the leases for? i can never find that file when i want to move to another server... |
14:19.05 | mitcheloc | tftp i've forgotten once or twice ;), don't forget that! i had to manually redo the directory *sigh* |
14:19.15 | shmaltz | mithceloc, that way your dhcpd doesnt give out the same ip again ;( |
14:19.17 | luke-jr_ | shmaltz: what would dhcpd be used for asterisk-specific? o.O |
14:19.20 | *** join/#asterisk adaro1 (n=adaro@62-213-205-18.colo.kangaroot.net) |
14:19.32 | shmaltz | luke-jr, automatic provisioning |
14:19.39 | luke-jr_ | oh |
14:19.48 | mitcheloc | it's more like for the infrastructure of your network, your phones specifically |
14:20.11 | luke-jr_ | Can Asterisk handle IPv6, I wonder |
14:20.12 | mitcheloc | allthough, shame on us using the asterisk server for dhcp, tftp, AND asterisk... |
14:20.26 | shmaltz | luke-jr, I don't think so |
14:20.34 | shmaltz | mitcheloc, why? |
14:20.50 | shmaltz | I do it all of the time, and apache, webmin as well |
14:21.06 | mitcheloc | i know, it's just a major point of failure... |
14:21.08 | shmaltz | and sometimes mysql for CDR as well |
14:21.18 | shmaltz | mitcheloc, what is? |
14:21.19 | mitcheloc | if the asterisk server dies, so do your workstations (if they lose their leases), etc, etc, theres a lot of reasons |
14:21.28 | mitcheloc | dhcp on that machine |
14:21.44 | shmaltz | mitcheloc, what are you talking about, why would the asteirsk server die quicker than Windowz? |
14:21.54 | mitcheloc | i never mentioned windows... |
14:22.09 | shmaltz | mitcheloc, the other option for dhcpd would be windowz |
14:22.36 | mitcheloc | no, the other option is another server with dhcp on it, putting it on a layer 3 switch, or a decent firewawll/edge device that can run it |
14:22.49 | Nugget | "the other". How quaint. |
14:23.00 | shmaltz | <PROTECTED> |
14:23.11 | *** join/#asterisk danzig (n=chatzill@130.226.169.177) |
14:23.18 | shmaltz | mitcheloc, and that supports vendor specific dhcp options????????????????? |
14:23.26 | Nugget | uptime is irrelevant. it's *downtime* that matters. |
14:23.33 | mitcheloc | 06:23:24 up 369 days, 11:36, 1 user, load average: 0.00, 0.00, 0.00 |
14:23.52 | shmaltz | Nugger, you are right, that box never had downtime without *me* bringing it down |
14:24.11 | mitcheloc | thats not the point shmaltz, if it goes down, you are screwed |
14:24.17 | mitcheloc | you have more work and more services to bring back online |
14:24.30 | asteriskmonkey | shmaltz: your probably getting clicks cause your audio is cliping |
14:24.31 | shmaltz | mitcheloc, but where doesn asterisk come into this "if it goes down..." |
14:24.37 | mitcheloc | specifically workstations can be affected without dhcp...so your users don't have any computers to use! |
14:24.45 | shmaltz | asteriskmonkey, meaning? |
14:24.48 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
14:25.05 | flujan | hi all. |
14:25.07 | mitcheloc | shmaltz, i'm just saying, instead of having the workstations go down as well as the phones, it's better to have one or the other, (if you have a choice) |
14:25.16 | shmaltz | mitcheloc, what you say has *nothing* to do with asterisk on the same machine |
14:25.17 | flujan | I have a analogic server which has 3 E1 interfaces. Now, I intent to use asterisk with e1 cards but integrate both servers in a migration process... |
14:25.17 | flujan | flujan Can I connect a E1 interface from the analogic server to a E1 interface from the asterisk server? |
14:25.27 | asteriskmonkey | shmaltz: meaning your hitting the audio threshold.. use ztmonitor on the channel you should see it peak when you hear a click |
14:25.33 | Nugget | actually it has everything to do with it. |
14:25.35 | asteriskmonkey | they you can adjust it accordingly |
14:25.49 | mitcheloc | shmaltz, no i'm not blaming asterisk, i'm just worried about too many services on one machine |
14:25.51 | austinnichols101 | flujan: yes, with a crossover cable |
14:25.59 | shmaltz | asteriskmonkey, how do I adjust it? tx/rxgains in zapata.conf? |
14:26.09 | shmaltz | mitcheloc, you are over worried |
14:26.18 | Nugget | or over experienced. |
14:26.19 | asteriskmonkey | yes |
14:26.23 | shmaltz | lol |
14:26.30 | mitcheloc | ;) |
14:26.31 | shmaltz | asteriskmonkey, thanks I'll monitor it |
14:26.36 | asteriskmonkey | remember you have to restart asterisk after you make the changes though or load/unload zap modules |
14:26.46 | adaro1 | I'm having a problem trying to overflow calls from one queue to another |
14:26.50 | mitcheloc | well when you have a server failure, and an entire business offline, you'll understand what an additional peace of mind can mean for you |
14:26.50 | asteriskmonkey | np its in /usr/src/zaptel-xx/ |
14:27.18 | adaro1 | Queue1 has only one extension when that is busy I want the call to go to Queue2 which has 2 extensions |
14:27.41 | adaro1 | I'm getting please try again later instead of the overflow |
14:27.46 | mitcheloc | i had a server up for over a year and a half, and then it went down..just one day, no way to get the thing back up either quickly, it requires a custom kernel compile with hdlc... |
14:28.13 | shmaltz | asteriskmonkey, why am I getting this: |
14:28.14 | asteriskmonkey | ive had one up for 8months now running asterisk and centos :D |
14:28.15 | shmaltz | Unable to open /dev/dsp: No such device or address |
14:28.16 | shmaltz | Cannot open audio ... |
14:28.24 | adaro1 | any suggestions on the best setup for this? |
14:28.47 | *** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac) |
14:28.48 | asteriskmonkey | shmaltz: it cant find a device.. if youve recently recompiled zap you have to re modprobe |
14:29.12 | shmaltz | asteriskmonkey, this is running right now |
14:29.21 | asteriskmonkey | odd then |
14:29.33 | asteriskmonkey | dont know would have to poke around the box |
14:29.51 | asteriskmonkey | google for some possible things to look for :) |
14:29.56 | shmaltz | asteriskmonkey, it's a bug, I did ztmonitor 9 -vvvvvvvvvvvvvvv |
14:30.01 | shmaltz | so it gave me that error |
14:30.09 | shmaltz | if I do: |
14:30.11 | shmaltz | ztmonitor 09 -vv |
14:30.12 | shmaltz | its ok |
14:30.18 | adaro1 | IS it possible to overflow one queue to another |
14:30.23 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
14:31.16 | shmaltz | adaro1, based on what conditions? |
14:32.06 | adaro1 | shmaltz - just if the extension in queue1 ( it only has one extension ) is busy |
14:32.20 | *** part/#asterisk flujan (n=flujan@internet.nube.com.br) |
14:32.47 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
14:32.47 | *** mode/#asterisk [+o anthm] by ChanServ |
14:33.37 | shmaltz | adaro1, so I guess that if you do: |
14:33.39 | shmaltz | exten => dial(extenthatsbusy,1) |
14:33.41 | shmaltz | exten => jumphere if busy and use second queue |
14:33.42 | shmaltz | exten => if not busy call queue |
14:33.44 | shmaltz | this should sure work |
14:34.02 | sergeus | ~seen jvu |
14:34.17 | jbot | sergeus: i haven't seen 'jvu' |
14:34.20 | adaro1 | I'll give it a go - thanks |
14:34.38 | *** join/#asterisk oej (n=oej@fuckoff012231-1-3.clients.easynet.fr) |
14:34.46 | sergeus | ~seen jwu |
14:34.48 | jbot | jwu <n=jwu@221.221.11.37> was last seen on IRC in channel #asterisk, 3h 5m 36s ago, saying: 'idunno, can you help?'. |
14:36.32 | shmaltz | looks like dells ftp server is down |
14:39.26 | *** join/#asterisk ManxPower (n=ewieling@stirprop-S4-0-0-21.ndcr2.datasync.net) |
14:39.53 | Garak_ | what dose facility not subscribed mean in idefisk |
14:41.03 | *** join/#asterisk iCEBrkr (n=icebrkr@6532244hfc169.tampabay.res.rr.com) |
14:41.08 | iCEBrkr | werd! |
14:41.41 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
14:42.41 | *** join/#asterisk need_sccp_help (n=none@198.60.73.171) |
14:43.06 | Blackthorn | Hey guys/gals. This is off-topic but I seem to always get more help in here than on the Cisco channel. I have our wireless unit that services the town connected into catalyst switch and the port keep's reseting alot. i show some errors such as "False carrier errors" Anyone know what that means? |
14:43.48 | Blackthorn | This town wireless is what we run our voip services on and is connected to our * box. Voip cut-offs :( |
14:44.08 | need_sccp_help | Anybody familiar with sccp setup in asterisk for cisco 12sp phones? |
14:44.16 | mitcheloc | black, out of curiousity what town? |
14:46.32 | Blackthorn | Marion VA |
14:48.47 | lunaphyte_ | what is 'n' in 's,n,Wait,2' ? |
14:49.39 | need_sccp_help | "n" means next priority |
14:50.24 | need_sccp_help | Instead of ordering priorities 1,2,3,4, etc.. . You can order then 1n,n,n,n, etc. . . |
14:50.54 | lunaphyte_ | simply meaning next in the list? |
14:50.58 | need_sccp_help | exactly |
14:51.00 | RoyK | yeah |
14:51.08 | greendisease | luckyduck: it is helpful if you ever end up needing to change the dialplan, you dont need to go back and renumber everything |
14:51.14 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
14:51.22 | need_sccp_help | Makes life easier |
14:51.53 | lunaphyte_ | ah - so if i didn't feel like numbering things, i could just keep them in the order i wish and they would essentially number themselves...? |
14:52.03 | need_sccp_help | correct |
14:52.08 | lunaphyte_ | thanks. |
14:52.35 | need_sccp_help | But remember that the first priority still must be numbered '1' |
14:52.45 | lunaphyte_ | was just about to ask that. ;) |
14:52.50 | need_sccp_help | After that you can use 'n' |
14:52.54 | *** part/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
14:54.26 | lunaphyte_ | s,n(value) - does 'value' name the priority so i can still goto it without it having a number? |
14:54.46 | *** join/#asterisk NDT (n=me@cpe-24-195-218-134.nycap.res.rr.com) |
14:59.23 | *** join/#asterisk asteriskNewb1e (n=chatzill@144.92.25.196) |
14:59.30 | backblue | anyone works with tdm? |
15:00.02 | asteriskNewb1e | I have a question about Asterisk@home with a X100P card for POTS. Anyone familiar with that setup? |
15:00.10 | Garak_ | backblue: I've done alittle bit |
15:00.20 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
15:00.42 | backblue | Garak_: why use tdm, if you can use a iax2 trunk between 2 asterisk servers? |
15:00.44 | asteriskNewb1e | I can make calls out and my phone rings when someone calls, but I would like to have voicemail pick up after 3 rings (~5 sec) |
15:01.02 | iCEBrkr | asteriskNewb1e: |
15:01.03 | iCEBrkr | ~amp |
15:01.04 | jbot | methinks amp is NOT supported here! people using it should join #amportal |
15:01.12 | asteriskNewb1e | thanks |
15:01.35 | Garak_ | backblue: TDM is for connecting to a telco or connecting to channel banks |
15:02.07 | asteriskmonkey | shmaltz: ztmonitor is not the same as asterisk :) it has a 2 v limit 1 shows the bar graph 2 shows with numeric indicators |
15:02.39 | backblue | Garak_: can we talk a litle in pvt? |
15:03.18 | RoyK | ~wtf is amp |
15:03.26 | RoyK | ~wtf amp |
15:03.44 | GerbilWrk | would it be advantageous to connect two sip phones to an asterisk box that talks to another asterisk box over an isdn link and uses that as the main gateway or two just have the two sip phones talking over the ISDN link to the one asterisk box? |
15:03.45 | shmaltz | asteriskmonkey, I know, but giving me that error when I enter -vvvvvvvvvvvvv is a bug |
15:03.46 | asteriskNewb1e | asterisk management portal |
15:04.40 | *** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net) |
15:07.06 | asteriskmonkey | shmatlz: yes good bug observation :D |
15:07.52 | asteriskmonkey | anyone experience any issues with any firewalls and iax2? i seem to be having an issue in which i can get a ring though but no audio and it mysteriously disconnects after 8 seconds |
15:08.17 | shmaltz | asteriskmonkey, what firewall is it? |
15:08.23 | asteriskmonkey | sonicwall |
15:08.24 | brad_mssw | asteriskmonkey: what v of asterisk? |
15:08.28 | fugitivo | asteriskmonkey: sounds like a codec problem |
15:08.34 | asteriskmonkey | 1.2 as of yesterday |
15:08.44 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
15:08.44 | brad_mssw | 1.2.???? |
15:08.49 | asteriskmonkey | ah ok :) ill go force ulaw to be safe |
15:08.57 | brad_mssw | or do you mean as of svn yesterday ? |
15:09.06 | fugitivo | if it rings, it should work (with iax) |
15:09.10 | brad_mssw | yeah, ulaw would be a good choice |
15:09.14 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
15:09.22 | asteriskmonkey | brad_mssw .. i dont do the 1.2.??? i compile by revision i think im like SVN-branch-1.2-r10368 |
15:09.55 | asteriskmonkey | i always use ulaw its nice but so damn heavy on bandwidth |
15:10.01 | brad_mssw | asteriskmonkey: ah, ok ... i usually don't do svn for production stuff |
15:10.07 | tdonahue | what causes "Stopping retransmission on '568a0d65068cb83838de831b1542f89b@208.51.101.194' of Request 102: Match Found |
15:10.08 | brad_mssw | (and I really only do production stuff) |
15:10.16 | tdonahue | " to be spammed to the debug log? |
15:10.56 | *** join/#asterisk UlbabraB (n=salama@host-84-222-46-106.cust-adsl.tiscali.it) |
15:11.04 | *** join/#asterisk bkw_ (n=bkw_@ip-207-145-170-139.lax.megapath.net) |
15:12.17 | asteriskmonkey | brad_mssw: thats branches not trunk branches is stable trunk is not |
15:12.38 | fugitivo | well, a release is not necessary "stable" ;) |
15:12.54 | asteriskmonkey | is anything made be the open source considered to be stable :) |
15:13.01 | fugitivo | apache |
15:13.05 | fugitivo | linux |
15:13.20 | fugitivo | sendmail |
15:13.20 | asteriskmonkey | dude id say they have some serious amount of years on them compared to asterisk |
15:13.28 | asteriskmonkey | well FreeBSD for the win |
15:13.35 | fugitivo | well, you asked for opensource stable :) |
15:14.22 | asteriskmonkey | lol sorry was asking for grief when i said that :) |
15:14.36 | asteriskmonkey | mmm ... odd it dosnt seem to like gsm.. |
15:15.02 | fugitivo | did it work with ulaw? |
15:15.05 | RoyK | asterisk will prolly stabilise one day.... |
15:15.17 | asteriskmonkey | switching to ulaw atm |
15:15.20 | asteriskmonkey | ill tell you in a sec |
15:15.25 | RoyK | why ulaw? |
15:15.36 | asteriskmonkey | its nackedest codec you can use |
15:16.07 | RoyK | µlaw is american evilness :) |
15:16.27 | tdonahue | no one knows what causes the "Stopping retransmission" message? |
15:16.34 | asteriskmonkey | ok so ulaw dosnt work :( |
15:16.37 | asteriskmonkey | ok so its not codec |
15:16.56 | fugitivo | weird |
15:17.05 | asteriskmonkey | here is my tree so to say cubixsoftclient(iax)=>asterisk=>pstn |
15:17.18 | asteriskmonkey | i can call numbers fine just incomming numbers get rapped and dropped |
15:17.25 | RoyK | asteriskmonkey: pastebin the config and verbose and debug outputs and ask again :) |
15:17.54 | *** join/#asterisk litecode (n=andrewb@12-217-30-205.client.mchsi.com) |
15:18.14 | asteriskmonkey | RoyK: there is nothing man in the verbose log that says anything why its ditching the call |
15:18.22 | litecode | I've been messing with this for a bit, but I upgraded to 1.2.4 and i get "No application 'Dial' for extension" any ideas |
15:18.37 | asteriskmonkey | its got to be a fireweall thing |
15:18.56 | asteriskmonkey | but i though iax was impervious to firewall .. (this works everywhere else) just not at this office |
15:19.14 | zoneout | How do I write a GotoIf() which matches against the number, is there a way to do this making use of the dialplan _XXX format? |
15:19.42 | mikefoo | asteriskmonkey: iax is only firewall friendly because its in/out ports stay the same. |
15:19.42 | asteriskmonkey | zoneout : yes |
15:20.13 | asteriskmonkey | mikefoo: so if the ports are no longer 5060 then i can assume the firewall is blocking them |
15:20.28 | asteriskmonkey | sorry 4569 |
15:20.29 | *** join/#asterisk Delvar (n=irc@host-83-146-53-46.bulldogdsl.com) |
15:20.55 | mikefoo | iax is on 5060? |
15:20.58 | mikefoo | ahh ok yah.. |
15:21.03 | asteriskmonkey | for some reason my iax connection is on port 47925 on the server.. so does this mean the firewall is blocking |
15:21.21 | mikefoo | asterisk: is it behind a nat? |
15:21.22 | asteriskmonkey | sorry iax is support to be 4569.. i quoted you a sip port |
15:21.26 | asteriskmonkey | no |
15:21.31 | asteriskmonkey | asterisk is on a public ip |
15:21.36 | zoneout | asteriskmonkey: how? :) |
15:21.56 | mikefoo | So.. disable firewall and test? |
15:22.18 | asteriskmonkey | dude :) ist a corporate firewall there is non of those disable options available to me |
15:22.34 | asteriskmonkey | so ill be happy if its just it dont work for this reason sorta answer :D |
15:22.50 | mikefoo | hahah |
15:23.34 | asteriskmonkey | but yea if its not on port 4569 can i can say look its the firewall things should be on that port not some whacked out one like 47925 |
15:24.33 | asteriskmonkey | ah wait |
15:24.45 | asteriskmonkey | i notice there is another iax connection from this location on that port |
15:24.46 | asteriskmonkey | :P |
15:24.49 | asteriskmonkey | grr |
15:25.06 | asteriskmonkey | so is this an iax port related problem |
15:25.06 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.8) |
15:25.27 | Dr-Linux | i forgot my sipura-2100 amdin password any idea? |
15:25.33 | RoyK | mikefoo: 5060 is sip |
15:26.06 | zoneout | How do I write a GotoIf() expression which matches against a number using _XXXX???? |
15:27.26 | *** join/#asterisk stormfr (n=StorM@sgc91-2-82-237-76-2.fbx.proxad.net) |
15:28.41 | asteriskmonkey | zoneout: if you google for zapteller youll see a nice example on the voip wiki with the code structure you require |
15:29.58 | mikefoo | RoyK: yes I know.. |
15:30.04 | remiss | what is the best way to call asterisk from an outside script? |
15:30.10 | mikefoo | i was responding to asteriskmonkey |
15:30.23 | remiss | or java |
15:30.24 | mikefoo | Hey, can anyone suggest a toll free did provider? |
15:30.53 | iCEBrkr | mikefoo: Good luck with that |
15:31.00 | mikefoo | lol.. |
15:31.06 | iCEBrkr | mikefoo: But asterlink.com has really cheap 800 DID's |
15:31.18 | stormfr | hello, is there any dsp card compatible with asterisk for G729 or G723 ? |
15:31.20 | iCEBrkr | mikefoo: $1.95/mo at like .02/min |
15:31.25 | mikefoo | yeah? niice |
15:31.35 | zoneout | asteriskmonkey: cheers |
15:31.37 | mikefoo | how are the for an inbound provider? |
15:31.44 | ManxPower | Anyone here familiar with Nortel? I need information on what to dial from a Nortel (Meridian?) analog line to access call pickup and all-station-page |
15:31.57 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
15:32.00 | iCEBrkr | mikefoo: I haven't used them exclusively yet. I just got the account. |
15:32.03 | Darwin35 | DR Linux Call Sipura |
15:32.13 | mikefoo | cool.. |
15:32.22 | ManxPower | zoneout, exten => _XXXX,1,Goto(context,extension,priority) |
15:32.29 | litecode | well, it looks like 1.2.4 is making PaX angry. |
15:32.30 | Darwin35 | or open it up and find the reset button |
15:32.55 | ManxPower | stormfr, NO. |
15:32.57 | iCEBrkr | mikefoo: If you're looking for a tollfree number, I'd try them. It's a very low risk, cheap investment. |
15:33.13 | *** join/#asterisk CoolAcid (n=jason@216.99.98.39) |
15:33.47 | *** join/#asterisk mhnoyes (n=mhnoyes@user-38lc0c2.dialup.mindspring.com) |
15:34.15 | RoyK | ManxPower: exten => _X.,1,Goto([[context,]extension,]priority)? |
15:34.59 | ManxPower | RoyK, no, that would match too much. He only wanted to match exactly 4 digits |
15:34.59 | mikefoo | iCEBrkr: thanks :) |
15:35.02 | RoyK | ManxPower: i just though about the goto syntax..... |
15:35.22 | ManxPower | RoyK, It's easier to tell people to always use all three |
15:38.05 | asteriskmonkey | ha thats odd sip with stun dosnt work but sip without stun does behind the sonicwall :P |
15:38.10 | stormfr | hello, i have several problem today with iax. Expanded trunk go from 12K to 128K in 1 second and finally said : "chan_iax2.c: Maximum trunk data space exceeded to x.x.x.x". Did i have to modify define of MAX_TRUNKDATA ? |
15:38.20 | *** join/#asterisk danzig (n=chatzill@130.226.169.177) |
15:38.29 | mikefoo | asteriskmonkey: oh its a sonicwall? |
15:38.46 | asteriskmonkey | yes |
15:39.04 | mikefoo | they have tons of features of packet analysis, which interfers with alot of crap, I would do any xml requests with those turned on. |
15:39.21 | mikefoo | I couldnt* |
15:39.49 | asteriskmonkey | whacky good to know though :) so anymore issue like that i can point at the sonicwall :D |
15:40.13 | mikefoo | hah, yeah |
15:41.10 | lunaphyte_ | when a peer makes a sip call to a number on my proxy, how can i give them a dialtone? |
15:41.42 | [TK]D-Fender | lunaphyte : Lookup "DISA" on the WIKI |
15:42.09 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
15:42.22 | dezent | hm strange.. when i compile zaptel i get this error "/bin/sh: line 1: scripts/mod/modpost: No such file or directory" |
15:42.29 | lunaphyte_ | [TK]D-Fender: thanks. |
15:42.31 | *** join/#asterisk firestrm (n=no@S01060013105c1c69.gv.shawcable.net) |
15:42.37 | dezent | anyonek know what could be wrong ? |
15:44.57 | *** join/#asterisk Nebukadneza (n=daddel9@i3ED6E1A6.versanet.de) |
15:45.09 | Nebukadneza | hihi |
15:45.48 | asteriskmonkey | DISA the bomb :D |
15:46.18 | asteriskmonkey | i have a php program where you go to a page enter your number and your pbx calls you so you can use its disa feature to dial back out :D |
15:48.32 | *** part/#asterisk mhnoyes (n=mhnoyes@user-38lc0c2.dialup.mindspring.com) |
15:49.39 | *** join/#asterisk saftsack (n=saftsack@p54A7EBE7.dip.t-dialin.net) |
15:50.16 | saftsack | is it possible to run a pri card and 30 - 60 VOIP telephones on one computer? |
15:52.01 | asteriskmonkey | is it possible to fit 12 people into a van |
15:52.06 | asteriskmonkey | depends on the pc man :D |
15:52.20 | asteriskmonkey | if its a p4 no worries |
15:53.36 | asteriskmonkey | i can get 40 ulaw calls on a linksys 54g router running linux |
15:53.57 | asteriskmonkey | so 60 calls shouldnt be an issue unless you want to do something mega funky |
15:54.22 | saftsack | ^^ |
15:54.32 | saftsack | ok |
15:54.37 | Delvar | in some recent testing on a dell dule xeon 1gb ram i had over 800 channels with no transcodeing |
15:54.41 | asteriskmonkey | anyone going to VON this year in toronto? |
15:55.05 | Delvar | i got about 300 channels when transcoding |
15:55.11 | asteriskmonkey | yes a dual 3 gig xeons craps at around 100channels tops g729 coding |
15:55.23 | Delvar | didnt try g729 |
15:55.29 | asteriskmonkey | Delvar: what gsm? |
15:55.37 | Delvar | yes |
15:55.38 | saftsack | Delvar, wtf? do you have 800 telephones or were that simulated channels? |
15:55.43 | Delvar | nooo |
15:55.50 | Delvar | i had 5 asterisk boxes |
15:56.03 | saftsack | :) |
15:56.04 | Delvar | 1,2 started callls, though 3 to 4 and 5 |
15:56.06 | docelm0 | dezent, YOU BROKE IT! |
15:56.21 | docelm0 | Delvar, I have 12 in my cluster right now |
15:56.23 | lunaphyte_ | so if i want ext 7508 to be for disa access, is something like this appropriate? exten => 7508,1,DISA(1234|home) ? |
15:56.25 | Delvar | so not a real world test by any means |
15:56.46 | dezent | docelm0: SHIY !!! |
15:56.49 | Delvar | cool |
15:56.56 | dezent | docelm0: that sux :/ |
15:57.11 | docelm0 | My real world test is 800 calls so far :) |
15:58.39 | iCEBrkr | 1 MILLLLLLLLLLLLION calls! </doctor evil> |
15:58.50 | RoyK | now many concurrent calls? |
16:00.16 | docelm0 | Im figuring around 5000 top's.. |
16:00.26 | docelm0 | 3000 w/ transcoding |
16:01.19 | asteriskmonkey | dude if you have cards with dsps your pci slots are the limit :D |
16:02.53 | remiss | suggestions on how to make asterisk do something from an external application? |
16:03.06 | *** join/#asterisk redondos (n=redondos@190.48.41.62) |
16:03.14 | asteriskmonkey | remiss - AGI's |
16:03.15 | remiss | e.g. if i want to call santa claus from a java-application.. |
16:03.28 | asteriskmonkey | agi agi agi |
16:03.34 | remiss | asteriskmonkey: isn't that the other way? O_o |
16:03.45 | *** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com) |
16:03.55 | asteriskmonkey | sos |
16:04.02 | remiss | e.g. asterisk calls up the agi-script? |
16:04.17 | asteriskmonkey | just make a php script or java scrpt that dumps calls into files for the asterisk manager to dial |
16:04.24 | remiss | ahhhhhh |
16:04.27 | remiss | of course :D |
16:04.37 | docelm0 | hay iCEBrkr I had 250 calls up last night at once.. its kinda impressive.. :p |
16:04.42 | asteriskmonkey | then in does dumpfiles specify a context in which you want it to execute and have your funckyness there :D |
16:05.11 | remiss | you can send me flowers, but i won't arrange them |
16:05.15 | Egonis | I just finished a new asterisk install... etc -- when I call extension 101 from extension 100, 101 can hear me, but I can't hear 100 |
16:05.24 | Egonis | it's via SIP |
16:05.25 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
16:05.43 | remiss | Egonis: if you want ppl to help you, you have to provide more information |
16:06.01 | remiss | e.g. nat or not |
16:06.07 | *** join/#asterisk salviadud (n=ralfalfa@dsl-201-129-72-144.prod-infinitum.com.mx) |
16:06.34 | mikefoo | remiss: dude, give me santa's number |
16:07.08 | asteriskmonkey | here is the most uberist asterisk box http://www.williamsglobal.com/fuseworxv2000.jpg 12terabytes storage space 32gig ram 4 cpu :D |
16:07.36 | asteriskmonkey | hehehe |
16:07.39 | mikefoo | I have a bone to pick with him from 1993, i got the fak MyBuddy and was mad fun of |
16:07.44 | remiss | mikefoo: i'm not gonna give you my mommas number :-/ |
16:07.44 | Egonis | remiss: Which information would be useful? |
16:08.28 | [TK]D-Fender | asteriskmonkey : Holy McFuck..... |
16:08.41 | [TK]D-Fender | asteriskmonkey : that box is just... WRONG.... |
16:08.53 | asteriskmonkey | thats the beast i be building now :) |
16:09.11 | [TK]D-Fender | asteriskmonkey : And that SuperMicro case is FUGLY. |
16:09.22 | [TK]D-Fender | Who needs that much for a telephony server? |
16:09.25 | asteriskmonkey | its for 8 ds3's |
16:09.37 | asteriskmonkey | banks |
16:09.50 | [TK]D-Fender | asteriskmonkey : can it ration off the CPUS to handle it properly? |
16:09.56 | zoneout | in a macro how do you do a Goto based upon the number of digits in a variable? I want something like: GotoIf(${VAR} MATCHES _XXXX?10:20)... How do you do this?? |
16:09.56 | Egonis | When I make a call to another SIP phone, there is a 1 second delay when I speak, and it indicates: Attempting native bridge of SIP/106-e6c5 and SIP/100-181b |
16:10.00 | asteriskmonkey | yes |
16:10.03 | [TK]D-Fender | DS3 *data* I take it, not voice? |
16:10.06 | Egonis | but I can't hear the person I am calling |
16:10.09 | asteriskmonkey | it can even run in multi jail mode |
16:10.12 | fugitivo | THAT'S "HEAVY" |
16:10.19 | asteriskmonkey | voice :D |
16:10.27 | [TK]D-Fender | asteriskmonkey : What card? |
16:10.30 | asteriskmonkey | going to be using next gen d3 cards with g729 dsps |
16:10.34 | asteriskmonkey | hehehe |
16:10.41 | asteriskmonkey | the new ones comming like i said :D |
16:10.46 | [TK]D-Fender | ahhh.. the "unmentionable" ones :) |
16:10.48 | dasuberdavid | who makes it? |
16:10.57 | asteriskmonkey | the box? |
16:11.01 | *** part/#asterisk redondos (n=redondos@190.48.41.62) |
16:11.01 | dasuberdavid | the card |
16:11.21 | asteriskmonkey | cant tell you cause there in distro negotiations atm |
16:11.30 | dasuberdavid | ah |
16:11.49 | fugitivo | sangoma |
16:11.54 | asteriskmonkey | lol no |
16:12.00 | fugitivo | microsoft |
16:12.03 | salviadud | is a red alarm on my zap channels a bad thing? |
16:12.13 | fugitivo | salviadud: no, green is a bad thing |
16:12.34 | asteriskmonkey | salviadud: no it usually means something is misconfigured or not plugged in |
16:12.50 | asteriskmonkey | it menas its going to blow up |
16:12.57 | salviadud | damn |
16:13.06 | asteriskmonkey | if it blinks slow at first then starts blinking faster..run |
16:13.35 | salviadud | well, im using an old config |
16:13.38 | salviadud | from asterisk 1.06 |
16:13.45 | salviadud | has zaptel changed that much? |
16:14.01 | mzo | wtf, still getting rejected from FWD? FWD do es not love me =( *cry* |
16:14.07 | fugitivo | you upgraded and now you get red light? |
16:14.15 | salviadud | yeah |
16:14.20 | fugitivo | weird |
16:14.32 | fugitivo | what card? |
16:14.39 | salviadud | generic clone |
16:14.42 | salviadud | tp100 |
16:14.48 | fugitivo | is the module loaded? |
16:14.53 | salviadud | yeah |
16:14.59 | salviadud | zaptel, and wcfxo |
16:15.00 | fugitivo | ztcfg |
16:15.14 | fugitivo | are the channels actually available? |
16:15.21 | salviadud | yeah, channel 01 and 01 |
16:15.23 | salviadud | i mean |
16:15.26 | salviadud | 01 and 02 |
16:15.33 | salviadud | i got 2 of those cards |
16:15.40 | salviadud | fxs kewlstart |
16:16.12 | mzo | http://pastebin.com/566905 ;p |
16:16.31 | fugitivo | are the cables ok? do you have dialtone? |
16:16.51 | [TK]D-Fender | ahhh.. the "unmentionable" ones :) |
16:17.09 | salviadud | no, the cables are not plugged |
16:17.15 | fugitivo | well |
16:17.28 | fugitivo | plug them |
16:17.37 | salviadud | errrrrr |
16:17.56 | salviadud | can't get a descent place to test this damn thing |
16:17.58 | fugitivo | [TK]D-Fender: the chinesse ones? |
16:18.04 | zoneout | in a macro how do you do a Goto based upon the number of digits in a variable? I want something like: GotoIf(${VAR} MATCHES _XXXX?10:20)... How do you do this?? |
16:18.22 | salviadud | just one more question |
16:18.26 | salviadud | in signalling |
16:18.37 | salviadud | if its fxs |
16:18.42 | *** join/#asterisk sch19 (n=sch19@adsl-223-232-80.mia.bellsouth.net) |
16:19.01 | salviadud | should it go signalling=fxsks or signalling=fxs_ks? |
16:19.16 | fugitivo | fxs_ks |
16:19.31 | fugitivo | fxsks=channel in zaptel.conf |
16:19.34 | salviadud | even for zaptel.conf? |
16:19.37 | fugitivo | signalling=fxs_ks in zapata.conf |
16:19.49 | fugitivo | ^^ |
16:19.50 | *** join/#asterisk feist (n=feist@nat-pool-msp.redhat.com) |
16:20.00 | salviadud | thanx man |
16:20.42 | *** join/#asterisk Skwid_ (n=Skwid___@bas1-montreal42-1177928372.dsl.bell.ca) |
16:22.28 | Katty | hi lads. |
16:22.32 | *** join/#asterisk lobstu (n=mippy@dsl-146-240-254.telkomadsl.co.za) |
16:22.37 | lobstu | join #rubyonrails |
16:22.38 | lobstu | :P |
16:22.38 | *** part/#asterisk lobstu (n=mippy@dsl-146-240-254.telkomadsl.co.za) |
16:22.38 | Katty | i've come to collect hugs. |
16:22.45 | Katty | and mushy food. |
16:22.52 | *** join/#asterisk saftsack (n=saftsack@p54A7DAAD.dip.t-dialin.net) |
16:22.53 | ChrisUK | >:P |
16:22.53 | saftsack | hi |
16:23.06 | saftsack | is the Elmeg IP290 a snom telephone? |
16:23.10 | Katty | hi ChrisUK (= |
16:23.12 | Katty | mister fender! |
16:23.28 | Katty | fugitivo: i'm /almost/ to a tolerable pain level without the pain killers :> |
16:23.42 | Katty | :< |
16:23.54 | lunaphyte_ | mmm. steak... |
16:24.15 | [TK]D-Fender | Katty: Can you send me some drugs by PayPal? |
16:24.39 | Katty | [TK]D-Fender: uhmm, no. |
16:24.43 | lunaphyte_ | why does my disa dialtone go away after a second or so? |
16:24.52 | Katty | [TK]D-Fender: they told me to flush the narcotics i couldn't take. |
16:25.02 | Katty | 13 pills of oxy-contin down the drain. |
16:25.13 | *** part/#asterisk Nebukadneza (n=daddel9@i3ED6E1A6.versanet.de) |
16:25.44 | Katty | fugitivo: odd you say that, because my surgeon told me to stop taking ibeuprofen and start taking extra strenght tylenol instead. |
16:26.23 | mzo | icky bad surgery :P |
16:26.41 | fugitivo | Katty: careful, don't drive when you take those pills |
16:26.51 | Katty | fugitivo: oh? |
16:26.59 | Katty | fugitivo: they don't make me sleepy |
16:27.08 | fugitivo | PM will make you sleep |
16:27.13 | Katty | i'm not taking PM :) |
16:27.18 | Katty | just extra strength |
16:27.18 | fugitivo | oh :) |
16:27.32 | fugitivo | i took PM once by mistake |
16:27.33 | fugitivo | at work |
16:27.38 | FlyboySR22 | PM is gond during the PM, add a nice glass of wine and it really works !! |
16:27.46 | Katty | FlyboySR22: you're awful. |
16:27.50 | FlyboySR22 | :-) |
16:28.03 | Katty | someday i'm going to drink wine. |
16:28.06 | FlyboySR22 | We hope you feel better Katty |
16:28.08 | FlyboySR22 | !! |
16:28.12 | fugitivo | Katty: you should |
16:28.15 | fugitivo | wine is good |
16:28.16 | Katty | thanks, but i'm afraid it's going to take awhile. |
16:28.22 | FlyboySR22 | :-( |
16:28.24 | Katty | fugitivo: i'll get around to it eventually :) |
16:28.27 | FlyboySR22 | That is no fun |
16:28.31 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
16:28.35 | Katty | hi justinu! |
16:28.38 | fugitivo | try argentinian red wine |
16:28.49 | kpettit | anyway to kill a call to a confrence room? |
16:28.53 | Katty | i've had a few sips of wine...and the ones i like the most are super super sweet. |
16:28.59 | Katty | almost like nectar |
16:29.06 | fugitivo | kpettit: unload app_meetme.so ? |
16:29.08 | kpettit | getting weird artifacts in a meetme room, seems the only way to fix it is to reset asterisk |
16:29.11 | Katty | bicardi apple is pretty good |
16:29.13 | justinu | hey catty, how's the mouth |
16:29.17 | justinu | katty even |
16:29.20 | Katty | justinu: eh, it's healing. |
16:29.20 | kpettit | fugitivo, worth a try. |
16:29.33 | justinu | bicardi makes wine? |
16:29.38 | justinu | bacardi? |
16:29.41 | Katty | justinu: the top ones are pretty much ok...i'm off the pain pills most of the time...taking one every now and then, mostly at night |
16:29.42 | fugitivo | kpettit: tell me if that works :) |
16:29.44 | Katty | justinu: it's not a wine |
16:29.49 | Katty | justinu: more like a wine cooler maybe... |
16:29.52 | justinu | ah, right |
16:30.06 | justinu | flavored beer ;) |
16:30.12 | kpettit | Feb 22 10:29:51 WARNING[7252]: loader.c:135 ast_unload_resource: Soft unload failed, 'app_meetme.so' has use count 3 |
16:30.17 | Katty | fugitivo: i have this weird alchol = EVIL thing stuck in my head from the JWs. |
16:30.44 | *** join/#asterisk danzig (n=chatzill@130.226.169.177) |
16:30.46 | kpettit | fugitivo, know of anyway to force the unload? |
16:30.52 | fugitivo | Katty: well, some alcohol is evil, like some databases (mysql) |
16:30.56 | FlyboySR22 | ah...my sister is JW...she drinks wine coolers :-) |
16:31.06 | fugitivo | kpettit: hmm, no |
16:31.21 | fugitivo | kpettit: if you enter the conf as admin and kick everybody? |
16:31.22 | justinu | katty: whats your association with witnesses? |
16:31.23 | Katty | FlyboySR22: yeah JWs say it's all good in moderation |
16:31.28 | Katty | justinu: i was raised as one |
16:31.33 | Katty | justinu: for..uhhm, around 18 years |
16:31.34 | kpettit | how can i kick everybody in the confrence |
16:31.40 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
16:31.40 | FlyboySR22 | Katty, which is smart by anoyone's standards :-) |
16:31.46 | Katty | justinu: i've spent my last two years away from home redoing my programming :) |
16:31.47 | fugitivo | kpettit: no idea, i'd never did that |
16:31.58 | fugitivo | brb |
16:31.58 | Katty | FlyboySR22: indeed, but my mother is one of the extreme JWs. |
16:32.08 | Katty | FlyboySR22: and she leans towards all alchohol being bad. |
16:32.09 | ManxPower | god hates me |
16:32.15 | Katty | FlyboySR22: so that's how i was raised. |
16:32.35 | FlyboySR22 | Katty, Ah..understood..all of my brothers and sisters are JWs and some of them are ilke your mom |
16:32.39 | justinu | katty: interesting |
16:32.51 | Katty | justinu: yeah, it is pretty interesting. |
16:32.57 | Katty | justinu: the first couple months were major culture shock |
16:33.12 | FlyboySR22 | Katty, I was raised in a Lutheran boys home far far away from them, so I ended up lutheran instead of JW...I guess its all how you were raised! |
16:33.16 | justinu | the witnesses often come to harass my hindu neighbors |
16:33.51 | salviadud | fugitivo |
16:33.58 | salviadud | i connected the line |
16:34.03 | salviadud | on alarms i get an OK |
16:34.16 | salviadud | i suppose it's cool |
16:34.19 | salviadud | is it? |
16:34.42 | Hmmhesays | da da da dad da da daldsadadfasdfadsf;lksjfd;laskdjf |
16:34.52 | justinu | i have a curious fascination with cults |
16:34.55 | Hmmhesays | I need a new domain name |
16:35.03 | justinu | not really sure if the witnesses are defined as a cult, but they share a lot of traits |
16:35.09 | [TK]D-Fender | I have perfected the ultimate anti-JW door-to-door technique : Take one in and grab every piece of material they can leave you with. Then draw a chalk outline on your porch, a red inverted pentagram on your door and litter the porch with their pamplets :) |
16:35.10 | mzo | astersik is a cult |
16:35.19 | Katty | Hmmhesays: you should come take care of me. |
16:35.22 | Hmmhesays | voipfister just isn't working out |
16:35.36 | Hmmhesays | feeling down? |
16:35.40 | Katty | Hmmhesays: we can do theraputic shopping and such |
16:35.42 | mzo | when JW's come to my door, i open the door wearing leather, and i ask them to join the orgy |
16:35.45 | Katty | Hmmhesays: more like cut up |
16:35.52 | Katty | Hmmhesays: and swollen....my face is still bruised. |
16:35.54 | mzo | not once do they ever stay, even when it's a cute girl =( |
16:36.02 | kippi | is there a user manual for the voicemail? |
16:36.03 | Hmmhesays | awww that sucks |
16:36.12 | FlyboySR22 | Katty, What happened..? |
16:36.15 | Katty | Hmmhesays: i don't think 'sucks' starts to cover it :/ |
16:36.27 | Katty | FlyboySR22: i had all four of my wisdom teeth cut out on friday. |
16:36.35 | FlyboySR22 | Katty, OUCH !!! |
16:36.42 | Katty | FlyboySR22: yeah, it's defnately been ouch. |
16:36.44 | *** part/#asterisk Skwid_ (n=Skwid___@bas1-montreal42-1177928372.dsl.bell.ca) |
16:36.48 | FlyboySR22 | Katty, forget the wine, go right to burbon!! |
16:36.52 | FlyboySR22 | :-) |
16:37.00 | Katty | FlyboySR22: the pain killer/narcotic they gave me for the pain....i was either allergic too, or it was too strong. |
16:37.10 | Katty | FlyboySR22: so..i couldn't take the pain killer. |
16:37.11 | FlyboySR22 | Katty, so crap on top of crap |
16:37.22 | Katty | FlyboySR22: yeah, lots of crappy crap this weekend. |
16:37.37 | FlyboySR22 | Katty, well hopefully you start to feel better very soon |
16:38.04 | Katty | FlyboySR22: i'm hoping so (= |
16:38.18 | Katty | FlyboySR22: currently at the stage where i've been on a liquid diet for almost a week, and my body is seriously complaining. |
16:38.33 | Katty | FlyboySR22: getting nauseas everytime i eat...really weak, etc. |
16:38.36 | FlyboySR22 | Katty, wow - it is taking that long to heal...? Is that normal..? |
16:38.43 | Hmmhesays | yes |
16:38.46 | Katty | FlyboySR22: well, it's healing......but... |
16:38.50 | FlyboySR22 | Katty, its been so long since I had mine out I cannot remember |
16:38.56 | Katty | FlyboySR22: after that severe pain i had..i'm almost afraid to try to eat anything. |
16:38.57 | Hmmhesays | it takes quite awhile |
16:39.00 | Katty | FlyboySR22: i don't want to be in pain. |
16:39.06 | FlyboySR22 | Katty, I don't blame you |
16:39.20 | justinu | i need help.... who has experience tuning EC on a single span T1 card (no DSP) |
16:39.28 | justinu | i'm willing to pay if you're good |
16:39.34 | *** join/#asterisk stse (n=stse@muedsl-82-207-237-090.citykom.de) |
16:40.02 | Hmmhesays | sounds kind of kinky |
16:40.03 | Katty | justinu: i'm experienced in hugging, does that count? |
16:40.34 | justinu | yes - but unfortunatly in this situation hugs aren't the suolution |
16:40.36 | justinu | :( |
16:40.36 | Katty | kthx. |
16:40.37 | stse | Hi! It's me again (my problem are the missing notify messages from asterisk to the subscriber (SNOM phones), so no blinking LEDs) |
16:40.37 | *** join/#asterisk zaf (n=tfournet@wsip-68-228-9-79.br.br.cox.net) |
16:40.43 | Katty | justinu: :< |
16:40.51 | Katty | justinu: all you need is hugs, dobedobedo |
16:40.57 | ManxPower | stse, how do you know they are being sent? |
16:41.11 | zaf | hi, can anyone recommend a good softphone for Windows? |
16:41.20 | ManxPower | All softphones suck |
16:41.21 | stse | ManxPower: they are not sent, I don't see them with ethereal. |
16:41.25 | justinu | stse: make sure mailbox= is set in sip.conf |
16:41.33 | ManxPower | stse, what is your mailbox= line in sip.conf? |
16:41.39 | [TK]D-Fender | I have perfected the ultimate anti-JW door-to-door technique : Take one in and grab every piece of material they can leave you with. Then draw a chalk outline on your porch, a red inverted pentagram on your door and litter the porch with their pamplets :) |
16:41.50 | [TK]D-Fender | dammit... stupid scroll-back errors.. |
16:41.50 | justinu | i just opened the door naked |
16:41.52 | [TK]D-Fender | jsdhfhjlasjkdfhiu34y5ljk3hq45 |
16:42.02 | [TK]D-Fender | where's /clear when I need it... |
16:42.42 | stse | No, I mean the notify messages to indicate that a watched channel is ringing/busy. |
16:42.56 | justinu | hints have to be set in the dialplan |
16:42.57 | stse | voicemail is working. |
16:43.19 | Katty | [TK]D-Fender: or you could just /politely/ ask them to not return |
16:43.35 | stse | justinu: there are hints, show hints and sip show subscriptions are showing the correct things (or so I think ;-). |
16:43.40 | Katty | [TK]D-Fender: never underestimate the power of being polite (= |
16:43.44 | [TK]D-Fender | Katty : That was a double paste, but it was COMEDY, not experience :) |
16:43.44 | ManxPower | stse, Ah. No idea. That's a hint thing. |
16:43.47 | lunaphyte_ | what is the difference between -v and -d ? |
16:44.06 | [TK]D-Fender | Katty : I am far too polite.... I've got to start cracking down on phone-spammers though... |
16:44.26 | justinu | stse: i got that presence notification stuff working on my snom360 |
16:44.38 | idpromnut | Katty: never underestimate the power of a 12gauge and a rocking chair on the porch :) |
16:44.39 | justinu | just so you know, it's possible |
16:44.51 | stse | justinu: *sigh* others too, but I don't have success. |
16:45.26 | justinu | #if LINUX_VERSION_CODE >= KERNEL_VERSION(2,6,3) |
16:45.27 | justinu | #define HDLC_MAINTAINERS_ARE_MORE_STUPID_THAN_I_THOUGHT |
16:45.27 | justinu | #endif |
16:45.29 | justinu | heh |
16:46.02 | Katty | idpromnut: i don't like guns. |
16:46.06 | justinu | ECHO_CAN_KB1? MG2? MARK3? |
16:46.12 | justinu | what should I be using? |
16:46.13 | *** join/#asterisk IOscanner (n=IOscanne@c-24-0-183-125.hsd1.tx.comcast.net) |
16:46.14 | stse | justinu: after I restart asterisk, for every channel I watch, I get a notify message, when someone rings this channel, telling me, the channel is terminated. ;-( |
16:47.08 | jbalcomb | anyone have a telrad phone system tied into asterisk via the PRI card? I'm looking for a proper zaptel and zapata config for this scenerio. |
16:47.40 | justinu | jbalcomb: we know nothing about telrad, but a standard PRI config, with signalling set to pri_net should do it |
16:47.47 | justinu | and perhaps a T1 cross over cable |
16:48.56 | saftsack | howto connect a pri card to the line? with a normal cat5 cable? |
16:50.38 | saftsack | is PRI supported by the zapteldrivers? |
16:51.14 | Goral | if i was going aftera book on asterisk from O'Reilly which one would it be? |
16:51.20 | *** join/#asterisk Skymarshal (n=Skymarsc@p54AF496E.dip0.t-ipconnect.de) |
16:51.24 | ManxPower | saftsack, no, it's supported by libpri and libpri uses zaptel |
16:51.36 | ManxPower | Goral, um, there should be only 1 |
16:51.47 | saftsack | sounds good :) does it run well? |
16:51.51 | Bambr | i got following error |
16:51.52 | Bambr | Feb 22 18:50:23 NOTICE[8291]: chan_sip.c:10851 handle_request_register: Registration from '321 <sip:321@192.168.1.205>' failed for '192.168.1.12' - Username/auth name mismatch |
16:51.54 | saftsack | or is it unstable like BRI? |
16:51.59 | Bambr | what's wrong? :) |
16:52.03 | ManxPower | Any carrier would use PRI |
16:52.04 | Skymarshal | Does a Global variable overrules a Channel variable or do they somehow coexist? |
16:52.22 | ManxPower | Bambr, you do not have the correct info in the [321] section of sip.conf |
16:52.28 | Goral | bambr i'm comming up with a few |
16:52.46 | ManxPower | Skymarshal, the channel variable would override the global vatriable, I thingk |
16:52.59 | Goral | but the first in line is Asterisk: The Future of Telephony is that it? |
16:53.12 | Skymarshal | ManxPower: I think that too, but can I verify this somewhere? |
16:53.13 | ManxPower | ~docs |
16:53.14 | jbot | it has been said that docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
16:53.24 | ManxPower | http://www.oreilly.com/catalog/asterisk |
16:53.31 | Bambr | ManxPower, actually i got like [term_default_321] section, and file name=321 there and also username=321 |
16:53.34 | ManxPower | Skymarshal, try it and see |
16:53.37 | Goral | TY ManxPower |
16:53.41 | ManxPower | Bambr, that's won't work |
16:53.46 | ManxPower | you need a [321] |
16:53.50 | ManxPower | name is not a valid option |
16:54.11 | ManxPower | [whater] is used for incoming auth, username=whatever is used for outgoing auth |
16:56.02 | Bambr | can i add any variable to that section, just for my own use? |
16:56.21 | Bambr | like myvar=myval ? |
16:59.53 | *** join/#asterisk need_sccp_help (n=none@198.60.73.230) |
17:00.01 | clyrrad | I would like to know that too, is it possible to make your own variables in that manner? |
17:01.33 | clyrrad | or better yet, could you be in a users conext in sip.conf and make myvar=firstname lastname? And then be able to use myvar in extensions.conf, is that possible? |
17:02.39 | xtrvd | All vars in 'extensions.conf' can be created by using: VARNAME => SIP/100 for example. and then you just have to use VARNAME in any of the configurations and it will insert what the variable equals. |
17:03.04 | xtrvd | I'm not sure if that's too elementary of an answer, but I don't understand what else you are saying. Sorry. |
17:03.37 | xtrvd | oh, and those exist in the [globals] context. |
17:04.27 | clyrrad | What i mean is.... can you make a variable like you do in [globals] but in the users context of sip.conf instead of making it global to all extensions and channels |
17:04.36 | clyrrad | so [sip_account] |
17:04.41 | clyrrad | myvar=foobar |
17:04.55 | clyrrad | accountcode=123 |
17:04.56 | clyrrad | etc... |
17:05.04 | *** join/#asterisk trailhead26 (n=flombard@hou-nat129.novolink.net) |
17:05.09 | xtrvd | Ahh, in the sip.conf... Wow, I wish I could tell you. I have no idea. =) |
17:05.15 | xtrvd | Sorry. |
17:05.18 | clyrrad | LOL |
17:05.36 | clyrrad | I would love to know how to do that (if its even possible) |
17:05.43 | Hmmhesays | I so love it when gateways don't work right |
17:06.34 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
17:08.00 | stormfr | hello, i have several problem today with iax. Expanded trunk go from 12K to 128K in 1 second and finally said : "chan_iax2.c: Maximum trunk data space exceeded to x.x.x.x". Did i have to modify define of MAX_TRUNKDATA ? |
17:10.40 | *** join/#asterisk djMax (n=chatzill@artsalliancelabs.com) |
17:10.58 | djMax | is it possible to simultaneously ring two extensions but with different CID? |
17:11.09 | FuriousGeorge | is "exterhost" only for sip? the wiki page is down. * caches my ip with my dynamiv *.dyndns.org addresses |
17:11.40 | FuriousGeorge | very annoying, must be some way to change |
17:12.08 | *** join/#asterisk Defraz (n=t0tal@tim.mychoice.cc) |
17:12.12 | remiss | moahahha.. now i can generate calls from my web-server :D |
17:12.34 | *** join/#asterisk fu3 (n=kka@234-200-29-134.hcc.mnscu.edu) |
17:12.38 | fu3 | hi |
17:13.32 | fu3 | I am currently experiencing problems with hangups. My asterisk box wont detect them :) - I had a guy from the phone company (qwest) come out here to check things out, and he said my line is signalled with loopstart. |
17:13.43 | fu3 | I have loopstart signalling configured and still, no dice. |
17:13.57 | fu3 | I understand loopstart sends a busy signal down the line to indicate a hangup, correct? |
17:14.14 | fu3 | When a hangup occurs, there is no voltage drop, no polarity shift, no busy signal, etc.. just dialtone. |
17:14.20 | justinu | fu3: try kewlstart |
17:14.22 | fu3 | Any ideas? Does that even make senese? |
17:14.25 | fu3 | I've tried it |
17:14.37 | Skid | *whooooooosh*, damn did you see that? right over my head |
17:14.38 | justinu | it's uncommon |
17:14.40 | Skid | :-) |
17:14.41 | *** join/#asterisk dca[laptop] (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
17:14.46 | fu3 | I thought ks was just a modified version of groundstart? |
17:15.20 | brad_mssw | dca[laptop]: rockynet, eh ... you don't happen to be with teliax, do you? |
17:15.49 | tronix | fu3: ks = loopstart with far end remote disconnect supervision |
17:15.51 | justinu | ks is loopstart with the battery drop detection, iirc |
17:15.55 | *** join/#asterisk tholo (n=tholo@nat.sigmasoft.com) |
17:16.06 | tholo | What's the secret to make NMI problems disappear with TE410P...? |
17:16.09 | fu3 | ahhh |
17:16.32 | fu3 | so I should see a battery drop on my multimeter when a hangup occurs correct? |
17:16.36 | tholo | I have some new servers, and when I load the zaptel drivers I get that, then the kernel goes bye-bye... |
17:16.57 | tholo | Regardless of which slot I put the card in etc... |
17:17.11 | fu3 | As soon as the asterisk box goes offhook, the line goes to around 8VDC, and stays there until it hangs up, even after the far end has disconnected and a dialtone returned. |
17:17.42 | tronix | tholo: good start would be to disable everything you can live without... usb ports, serial, irda, parallel port, etc. in BIOS |
17:17.57 | fu3 | the asterisk box play's its introduction message right over the dialtone. |
17:18.12 | justinu | fu3: that's not good |
17:18.14 | tholo | Already done that. Not that the IBM servers gives you much to change in the first place... |
17:18.15 | fu3 | I know :) |
17:18.22 | justinu | sounds like you're connected to a funky switch |
17:18.27 | justinu | that doesn't do the battery drop |
17:18.37 | fu3 | I've got the phone guys coming back out later.. so I hope to have something to ask/tell them. |
17:18.54 | fu3 | justinu.. sounds like it to me too. Like I said, I dont see any voltage fluxuation upon hangup. |
17:19.12 | tronix | fu3: please update #asterisk with what comes out of the telco guys visit. I'd be interested in hearing what they have to say about their switch |
17:19.19 | fu3 | you got it! |
17:19.21 | tronix | sweet, thanks |
17:19.41 | tronix | tholo: ahh.. hmm.. in that case, there are various stuff you can do, but don't recall details. I've got an URL in my notes... digging |
17:19.55 | jbalcomb | justinu: ok, thanks. I guess I'll know tomorrow morning. |
17:20.02 | justinu | you mentioned qwest...ask them if you're connected to a DMS100, or a 5ESS |
17:20.12 | justinu | if the answer is neither, i'm afraid you're on some vintage hardware ;) |
17:20.18 | fu3 | I believe I am :) |
17:20.19 | tronix | 1AESS? ;) |
17:20.21 | fu3 | I will ask though. |
17:20.22 | *** join/#asterisk skkip (n=Skipper@216.160.91.91) |
17:20.23 | justinu | heh |
17:20.24 | tholo | I just seem to recall a specific workaround for NMI issues being mentioned -- this is without any spans plugged in, even. Just loading the driver. So... |
17:20.30 | justinu | No 5. Crossbar ;) |
17:20.32 | tronix | hahaha |
17:20.45 | justinu | but I think the xbar switches will still do the loop current drop for disco sup |
17:21.14 | fu3 | When I say to the telco guys about far end disco sup (I like that) they look at me funny.. |
17:21.23 | fu3 | then they say "yeah" |
17:21.26 | fu3 | and then nothing. :| |
17:21.34 | fu3 | I've got 350 lines! you'd think they'd care ;) |
17:22.05 | justinu | 350 analog lines? |
17:22.15 | fu3 | yeah.. they come in over fiber |
17:22.21 | fu3 | to a channel bank of some sort, then to the demarc. |
17:22.25 | *** part/#asterisk UlbabraB (n=salama@host-84-222-46-106.cust-adsl.tiscali.it) |
17:22.28 | justinu | wow... you should just ditch the analog and go w/ a DS3 |
17:22.36 | justinu | PRIs over DS3 |
17:22.42 | fu3 | im planning on it, but i Have to make it work in a test environment first |
17:22.50 | fu3 | and I need to sort out the other problems im having here in the mean time. |
17:22.56 | Nugget | nah, just buy 350 clone X100Ps off ebay. |
17:23.04 | fu3 | that seems unreasonable :) |
17:23.23 | fu3 | I'm also having to deal with trying to get all my same numbers, but it looks like i'll have to get a whole new set. |
17:25.50 | salviadud | how about channelbanks? |
17:25.51 | skkip | anyone suggest a wireless sip phone on the cheap. I need at least 100 for trade show type events. |
17:26.23 | kpettit | skkip, good luck |
17:26.49 | kpettit | haven't heard of a wireless sip phone that works decent yet, let alone one that is cheap |
17:26.55 | asteriskmonkey | i have :) |
17:27.01 | kpettit | do tell |
17:27.02 | asteriskmonkey | ive got one that works great |
17:27.07 | asteriskmonkey | its the wip5000 :) |
17:27.16 | asteriskmonkey | but its not cheap :( |
17:27.24 | skkip | from where monkey? |
17:27.36 | RoyK | does it work if you sit > 5m from the access point? |
17:27.39 | asteriskmonkey | the wililamsglobal place i work at |
17:27.45 | asteriskmonkey | they have like 4 different ones |
17:27.57 | asteriskmonkey | there is a quad band gsm with wifi sip that does handoff too :D |
17:28.03 | asteriskmonkey | but that one is uber expensive :( |
17:28.12 | skkip | nice |
17:28.20 | justinu | nokia is coming out with a quadband GSM/wifi voip phone |
17:28.21 | asteriskmonkey | you can put 2 sim cards in though :) |
17:28.24 | skkip | but I just need them for local |
17:28.24 | justinu | E61 i think |
17:28.29 | skkip | check in check out deal |
17:28.30 | justinu | treo form factor |
17:28.55 | asteriskmonkey | cool, this one i know you can talk on your cell and as soon as you get in wifi it passes the call seamlessly and you dont know :P |
17:29.00 | asteriskmonkey | i dont know how it does that though |
17:29.03 | kpettit | http://www.voip-info.org/wiki/view/Hitachi |
17:30.38 | kpettit | asteriskmonkey, how many different wireless sip phones have you tried? Do you like the wip5000 the best? |
17:31.07 | asteriskmonkey | ive tried 5 the wip is the best but i think the gsm/wifi ones comming in march will be superior |
17:31.23 | kpettit | gsm/wifi will be killer |
17:31.39 | asteriskmonkey | they have it now |
17:31.44 | twisted[asteria] | no, murderers will be killer |
17:31.46 | kpettit | is 320 standard price for those? |
17:31.47 | asteriskmonkey | we carry 2 phone that do that with passoff |
17:31.53 | twisted[asteria] | gsm/wifi will be needed :) |
17:32.03 | jbalcomb | Anyone using the ARI web interface for voicemail and have a solution for the permissions of the VM files being root.root? |
17:32.12 | asteriskmonkey | the new ones comming in march look like the razor but have full video suppport so streaming video conversations :D |
17:32.33 | twisted[asteria] | lame |
17:32.35 | *** join/#asterisk fulgas (n=fulgas@82.102.2.254) |
17:32.56 | *** join/#asterisk caio1982_ (i=caio1982@CAcert-br/caio1982) |
17:33.16 | twisted[asteria] | i say fax over single frame photos |
17:33.21 | twisted[asteria] | i'll make a flipbook and then you can call me |
17:33.42 | asteriskmonkey | lol |
17:33.47 | kpettit | with RTA do you have to restart asterisk if you change extconfig.conf or can I just reload? |
17:33.56 | *** part/#asterisk stse (n=stse@muedsl-82-207-237-090.citykom.de) |
17:35.49 | asteriskmonkey | just reload |
17:36.11 | kpettit | sweet. Trying to do voicemail with it now. |
17:36.39 | *** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar) |
17:37.38 | Hmmhesays | i need a free app for turning a divx video into dvd format |
17:38.58 | *** join/#asterisk epablo (n=epablo@200.75.139.188) |
17:39.09 | epablo | Hi people11 |
17:39.14 | fu3 | hi |
17:39.40 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
17:40.03 | epablo | Any idea on why I can be getting a "ZT_SPANCONFIG failed on span 1: Invalid argument (22)" when doing a ztcfg? |
17:40.18 | fu3 | lets see the config file |
17:40.23 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
17:40.23 | shmaltz | My Cisco 7960 phones are freezing frequently, and my users are telling me that they see something Phase Error on the screen when it happens, anybody have a clue what it might be? |
17:40.40 | *** join/#asterisk Math` (n=math@modemcable148.4-81-70.mc.videotron.ca) |
17:40.41 | kpettit | Hmmhesays, look at dvdrip |
17:41.21 | Hmmhesays | dvdrip? |
17:41.28 | epablo | fu3: http://pastebin.ca/42791 |
17:41.48 | kpettit | Hmmhesays, google for it or look for it on freshmeat.net. |
17:43.34 | epablo | fu3: it's suposed to be a TE400P (tor2) card I bought on ebay |
17:43.41 | fu3 | hmm |
17:43.44 | *** join/#asterisk ful|work (n=fulgas@82.102.2.254) |
17:44.02 | GerbilWrk | has anyone experienced clicking sounds during calls through Teliax? |
17:44.50 | fu3 | that looks ok |
17:44.52 | fu3 | strange |
17:45.42 | jbalcomb | shmaltz i heard there is an issue with the firmware that causes something like that |
17:45.49 | fu3 | well |
17:45.55 | shmaltz | jbalcomb, which one the latest? |
17:46.06 | shmaltz | jbalcomb, or is it with 7.1 as well? |
17:46.50 | jbalcomb | shmaltz I think so but I am not sure. There was a fellow in hear last week asking for the firmware so he could stop his 7940/7960s from locking up |
17:47.35 | jbalcomb | s/hear/here |
17:47.53 | shmaltz | jbalcomb, he wanted what firmware? |
17:48.04 | shmaltz | I got 6.x, and the latest 7.x |
17:49.07 | jbalcomb | shmaltz i'm not sure which firmware he was looking for. I had P003-07-5-00 and he seemed to think that would do him some good. |
17:49.28 | shmaltz | jbalcomb, does it happen to you? |
17:49.47 | epablo | shmaltz: i haven't used that provider.. but the 6.x firmware works better for me |
17:49.50 | jbalcomb | shmaltz I've only had my cisco phones for two weeks but so far no problems |
17:50.28 | *** part/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
17:50.33 | shmaltz | jbalcomb, how many simul phone calls you handle on those? |
17:50.40 | shmaltz | epablo, thanks |
17:50.46 | *** join/#asterisk stoffell (n=stoffell@d5153FE41.access.telenet.be) |
17:50.56 | jbalcomb | shmaltz we have the 7940s so 2 would be my guess |
17:51.21 | shmaltz | jbalcomb, b/c they telling me it happnes mostly when 3 or more calls |
17:52.09 | jbalcomb | shmaltz could be. we only have 2 of the phones and they are only 2 lines so we're not likely to have '3 or more' vey often |
17:53.16 | jbalcomb | shmaltz on the side, do you know of a solution for making the auto-answer actually ring or make any noise before it picks up? |
17:54.40 | *** join/#asterisk loick (n=loick@APuteaux-151-1-82-111.w86-205.abo.wanadoo.fr) |
17:54.50 | shmaltz | jbalcomb, someone had a script that logs into the Cisco using Telnet and the test function of the cisco phone which actualy answers the phone, and NOT the auto answer feature, and the script is called with M from the Dial command, using a delay in the script you could have it first ring 2 seconds or just one, you decide |
17:55.17 | *** join/#asterisk sevard (i=sev@24-179-181-160.dhcp.dlth.mn.charter.com) |
17:55.38 | *** join/#asterisk g4m (n=g4m@dsl253-014-003.sba1.dsl.speakeasy.net) |
17:55.56 | jbalcomb | shmaltz do you have a copy of the script and do you know if its perhaps using AddSIPHeader(call-info) to tell the call to pick up? |
17:57.16 | sevard | docelm0: you around? |
17:57.19 | shmaltz | jbalcomb, no it's not, its using a telnet library from perl that logs in using telnet into the phone, and then uses the test function to answer the phone, the test functions allows you to press any key on the phone over the network |
17:57.42 | shmaltz | jbalcomb, this script is somewhere in the user list archives from around a year ago |
17:57.47 | jbalcomb | shmaltz ah, ok. im looking for it now. |
17:58.29 | sevard | Does anyone have logs for this channel from now to about 3 weeks ago? |
17:58.50 | *** part/#asterisk trailhead26 (n=flombard@hou-nat129.novolink.net) |
17:59.11 | shmaltz | jbalcomb, http://lists.digium.com/pipermail/asterisk-users/2005-February/088614.html |
18:01.44 | fu3 | ok |
18:01.48 | *** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar) |
18:01.50 | fu3 | I just got off the phone with qwest |
18:01.57 | fu3 | they have no idea what "disconnect supervison" is |
18:02.11 | fu3 | and they cant confirm my questions about what "is supposed to happen upon disconnect" |
18:02.16 | ManxPower | fu3, It's called Far End Disconnect Supervision. Read the tarrifs to get a USOC. |
18:02.30 | fu3 | I said Far End Disconnect Supervision to them |
18:02.35 | salviadud | wow, those guys are amateurs |
18:02.39 | fu3 | I dont know what the tariffs are, or a USOC. |
18:02.41 | bweschke | fu3: a.k.a. CPC (call progress control0 a.k.a. reversing line polarity momentarily at the end of the call |
18:02.41 | ManxPower | hence my suggestion to read the tarrifs. |
18:02.55 | fu3 | bweschke.. yeah when I said that, they couldnt confirm if it was supposed to happen or not. |
18:02.59 | ManxPower | I've never heard of a real telco in the USA NOT providing FEDS |
18:03.08 | fu3 | I didnt say CPC.. I should mention that. |
18:03.15 | *** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar) |
18:03.21 | bweschke | fu3: http://www.voip-info.org/wiki/index.php?page=Asterisk+Disconnect+Supervision |
18:03.29 | fu3 | ive been over that a million times |
18:03.32 | bweschke | k |
18:03.32 | fu3 | thanks though |
18:03.35 | ManxPower | fu3, tarrifs == legal description of telco services as required by the regulators. USOC is the Universal Service Ordering Code, it's the "part number" |
18:03.40 | bweschke | did you try the lighted keypad thing? |
18:03.42 | fu3 | ahhhh cool! |
18:03.47 | *** part/#asterisk epablo (n=epablo@200.75.139.188) |
18:03.49 | fu3 | no... I dont have one, and cannot find one. |
18:03.54 | bweschke | k |
18:04.01 | fu3 | I had a qwest guy here TODAY |
18:04.04 | bweschke | u have a dc voltmeter? |
18:04.08 | ManxPower | fu3, you're trying to get me to do all the work, right? |
18:04.10 | fu3 | and he promised me that we were using loopstart |
18:04.14 | fu3 | yes, I have a multimeter. |
18:04.20 | fu3 | ManxPower.. No.. I want to do the work! |
18:04.21 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
18:04.23 | fu3 | I want to learn! |
18:04.27 | bweschke | hook it up to the line - see if it dips momentarily when the call ends |
18:04.31 | fu3 | it doesnt |
18:04.32 | ManxPower | http://www.google.com/search?hl=en&q=site%3Awww.qwest.com+usoc&btnG=Google+Search |
18:04.33 | fu3 | already verified that. |
18:04.47 | bweschke | then u aren't getting it from them - but you knew that already |
18:04.52 | fu3 | I do not get a battery drop, a polarity shift, OR a busy signal, or anything. |
18:05.04 | fu3 | I dont know what you mean bweschke |
18:05.09 | fu3 | i DO get my lines from qwest |
18:05.17 | fu3 | ohh |
18:05.18 | bweschke | you aren't getting CPC signaling from them |
18:05.19 | fu3 | duh :) |
18:05.20 | fu3 | yeah. |
18:05.23 | fu3 | thats right |
18:05.36 | fu3 | but they swear their gear is working fine.. |
18:05.38 | fu3 | what do i tell them? |
18:06.23 | bweschke | I'm sure it is - they're not required to provide it on standard analog service - unfortunately |
18:06.23 | fu3 | this isnt |
18:06.23 | fu3 | its a business trunk |
18:06.23 | fu3 | 350 lines |
18:06.23 | fu3 | across a fiber, to a channel bank, to the demarc and then to the stations. |
18:06.23 | bweschke | still.... depends what the prod definition of it is for them |
18:06.32 | fu3 | I see.. |
18:06.35 | fu3 | garrh. |
18:06.36 | bweschke | oh - well that's gotta be a config setting in the channel bank I'd think |
18:06.44 | fu3 | The telco guy is coming back out here in a little while. |
18:07.18 | fu3 | The guy also said that 40VDC is normal for our lines. |
18:07.21 | bweschke | cause I doubt they're actually doing loopstart through the fiber to the channel bank - eg - if it's coming in PRI to the CB - then you're signaling is coming into the CB - you just need the CB to tell the analog stations via CPC signaling |
18:07.21 | fu3 | at least, he suspected. |
18:07.27 | fu3 | my lines idle at 38-41 VDC |
18:07.32 | fu3 | AT the DEMARC! |
18:07.40 | *** join/#asterisk delmar (n=Delmar@203-114-178-231.inspire.net.nz) |
18:08.01 | fu3 | bweschke.. good point. |
18:08.16 | fu3 | all the batteries and everything are all right next to the CB. |
18:08.40 | fu3 | Shouldnt my lines be running at around 48VDC though? |
18:08.54 | fu3 | is a drop of 10 volts going to cause these kinds of problems? |
18:08.59 | ManxPower | fu3, what state are you in? |
18:09.04 | fu3 | Minnesota |
18:09.10 | ManxPower | HEY! you have a channel bank? |
18:09.11 | bweschke | no. not necessarily - I think there's a tolerance for analog line voltage |
18:09.20 | fu3 | Not one I can use for testing |
18:09.39 | fu3 | but yes, there are four CBs. |
18:09.51 | ManxPower | A channel bank totally invalidates anything anyone has told you about CPC |
18:10.02 | fu3 | great :) |
18:10.19 | fu3 | I always liked square 1 |
18:10.20 | ManxPower | and invalidates most of the CPC info on the wiki |
18:10.40 | ManxPower | fu3, you need to start by finding out if you have a CT1 (channelized T-1 voice) or a PRI |
18:10.54 | fu3 | i'm almost certain (but not 100%) that it's a CT1. |
18:11.07 | ManxPower | become %100 |
18:11.14 | justinu | most CB's don't do PRI, no? |
18:11.26 | bweschke | justinu: sure they do - the ones that do flex data/voice do |
18:11.35 | justinu | hmm, i've never had the pleasure of working with one |
18:11.36 | fu3 | well.. the phone company told me it was a CT1 earlier, but I dont trust anything they say anymore. |
18:12.01 | fu3 | brb |
18:12.04 | *** join/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com) |
18:12.40 | ManxPower | Most CT1s seem to provide CPC using CAS. |
18:13.04 | ManxPower | fu3, So you have Telco->line->CB->???? |
18:13.07 | ManxPower | And what is the ???? |
18:13.37 | ManxPower | Adtran 850s can, 750s can't. |
18:13.44 | ManxPower | fu3, what BRAND of channel bank do you have? |
18:15.45 | fu3 | I can find out in a few minutes. |
18:15.47 | clyrrad | how can I make asterisk parse the variable so the include works properly? Do i need Eval or something similair? #include "custom/${ACCOUNTCODE}/main_menu.inc" |
18:15.58 | fu3 | I am telco->fiber->mux->CB->demarc->station |
18:16.24 | bweschke | fu3: DS3 mux? |
18:16.46 | justinu | probably OC3 to DS3 to T1 |
18:16.54 | justinu | i have some facilities delivered like that |
18:16.57 | ManxPower | clyrrad, I don't believe #include supports that |
18:17.23 | clyrrad | ManuxPower, will it work if I just include a variable? instead of a path and variable? Or what is the way around this? |
18:17.26 | ManxPower | clyrrad, since #incude is read ONCE on load and never again |
18:17.35 | bweschke | no. it doesn't. because it needs to process #includes before eval'ing the expressions for variables |
18:17.41 | justinu | fu3: bottom line here is that the CB is before the demarc, so its your telco responsibility to provide that CPC loop current drop |
18:18.01 | clyrrad | So what is the solution if I need this type of functionality? |
18:18.15 | clyrrad | Is there another function or app I can use to accomidate this? |
18:18.17 | ManxPower | clyrrad, Goto(${ACCOUNTCODE},extension,1) |
18:18.28 | clyrrad | but its a File that i need included |
18:18.36 | clyrrad | the file has a bunch of dial plan stuff in it |
18:18.53 | ManxPower | clyrrad, What are you trying to accomplish? |
18:19.16 | clyrrad | I have a custom file, main menu.... for differnt DID's, each DID has its own main menu (ie seperate companies) |
18:19.23 | ManxPower | clyrrad, #include works EXACTLY like a C #include |
18:19.35 | clyrrad | when you dial in, based on the account code it should include the respecive main menu |
18:19.41 | ManxPower | clyrrad, put each company in it's own context, then use Goto based on the DID |
18:19.55 | clyrrad | I have each company in thier own context |
18:20.11 | ManxPower | so what's the problem? |
18:20.13 | clyrrad | I just have the functionaitly of the main menu in seperate files so as not to clutter up my extnesions.conf file |
18:20.31 | ManxPower | clyrrad, Well you'll have to do an #include for each company manually in your dialplan |
18:20.34 | clyrrad | and I need a way to get that file included with out having to hardcode the patch |
18:20.44 | clyrrad | thats what i was getting at.... |
18:20.45 | clyrrad | hrm |
18:20.45 | ManxPower | clyrrad, You can.t |
18:20.46 | clyrrad | that sucks |
18:20.51 | ManxPower | clyrrad, #include works EXACTLY like a C #include |
18:21.03 | djMax | so I upgraded * recently. Now, sip show peers has unknown hosts for everything. Did something change with the handling of sip peers? |
18:21.11 | clyrrad | make sense, thats why I was wondering if there was another way to do it with out hard coding |
18:21.11 | need_sccp_help | Anyone familiar enough with sccp in asterisk that they could help me with the proper sccp.conf for cisco 12SP phones? |
18:21.12 | ManxPower | #include is processed before ANYTHING else. |
18:21.22 | clyrrad | I see.... |
18:21.29 | ManxPower | clyrrad, hardcode it |
18:21.38 | clyrrad | too bad there is not such a function, woudld come in handy |
18:21.55 | clyrrad | thanks for clarification |
18:22.54 | sevard | Does anyone have logs for this channel from now to about 3 weeks ago? |
18:23.45 | iCEBrkr | lol |
18:23.52 | fu3 | justinu.. so I need to specifically ask for CPC from the CB, which SHOULD BE a voltage drop, and I SHOULD be able to see that happen with a multimeter? Correct? |
18:23.56 | iCEBrkr | sevard: Stop being a Nancy Boy. |
18:24.03 | sevard | iCEBrkr: sup bro |
18:24.06 | iCEBrkr | :D |
18:24.15 | sevard | I was just wondering where docelm0 posted his weather script |
18:24.23 | *** join/#asterisk razu_ (n=razu@ip228.cab74.mus.starman.ee) |
18:24.26 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
18:24.28 | iCEBrkr | sevard: I don't think it's 100% ready.. |
18:24.37 | sevard | I suggested that somebody should write a weather script that uses allison's voice in the wx/ directory and he went out and did it |
18:24.41 | fu3 | Now, like I said, my lines idle at around ~39VDC. What kind of drop should I witness? |
18:24.52 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
18:24.58 | iCEBrkr | sevard: I understand that, I tested it a few times for him |
18:25.06 | sevard | iCEBrkr: I know it wasn't but I was under the assumption that he had a project page up |
18:25.14 | iCEBrkr | sevard: I don't think so. |
18:25.18 | justinu | fu3: correct |
18:25.23 | sevard | iCEBrkr: how's she sound |
18:25.38 | iCEBrkr | sevard: Oh, I tested it before it had all the sound files worked out |
18:25.47 | sevard | neat |
18:25.47 | djMax | can anyone point me to how host=dynamic in SIP is "interpreted"? |
18:25.48 | justinu | fu3: basically the channel bank should interrupt the loop current for approximately 500ms |
18:25.56 | justinu | fu3: so it may be difficult to see that on your meter |
18:26.06 | fu3 | ahhhh |
18:26.11 | fu3 | yeah thats pretty short. |
18:26.16 | justinu | the duration is configurable |
18:26.21 | iCEBrkr | sevard: It was pretty rough at the time, he told me he was gonna smooth out the rough edges and release it soon. He wasn't ready last week. So I'm not sure what state/stage it's in |
18:26.25 | justinu | you might try at home with a real POTS line and see what happens |
18:26.44 | sevard | iCEBrkr: cool. thanks for the update. |
18:26.44 | fu3 | I will give it a whirl. |
18:26.48 | iCEBrkr | np |
18:26.55 | fu3 | I certainly appreciate all of everyone's advice. |
18:26.59 | fu3 | thanks! |
18:27.09 | need_sccp_help | Host=dynamic is used to specify that your sip client uses DHCP (changing address) rather that a static address |
18:27.15 | *** part/#asterisk tholo (n=tholo@nat.sigmasoft.com) |
18:27.24 | justinu | sorta |
18:27.31 | justinu | it actually means that you don't know the address of the client |
18:27.48 | justinu | and that it has to register before you can talk to him |
18:28.13 | *** join/#asterisk perlmonky (n=perlmonk@pix.benchmark-systems.com) |
18:28.17 | need_sccp_help | OK, that sounds good, more in depth than my answer |
18:28.27 | djMax | so what's happening is that it's losing that IP for some reason. I've set the registration timeouts on the phones (I think), but at the moment I have no IPs on sip show peers. |
18:28.29 | *** join/#asterisk stse (n=stse@muedsl-82-207-237-090.citykom.de) |
18:28.47 | djMax | but I can make outbound calls just fine, which is strange. |
18:28.55 | justinu | djMax: sounds like a NAT binding issue |
18:29.03 | justinu | djMax: is qualify=yes set for the phones? |
18:29.34 | djMax | in *? no |
18:29.35 | stse | hi! Are here some experts concerning the hint extensions and the notify messages (terminated,busy,ringing)? |
18:29.35 | sevard | iCEBrkr: you a ladies man? :) |
18:29.40 | *** join/#asterisk miller7 (n=none@gige-2.office-nl.irismedia.gr) |
18:29.46 | RoyK | hm |
18:29.47 | justinu | djMax: yeah, try setting that in sip.conf. |
18:29.55 | RoyK | suddenly meetme stopped talking to me |
18:29.56 | RoyK | <PROTECTED> |
18:30.03 | RoyK | "that is not a valid conference number" |
18:30.13 | RoyK | but |
18:30.13 | RoyK | conf => 10,10,10 |
18:30.18 | RoyK | so it should all be fine |
18:30.24 | RoyK | doesn't matter what i try |
18:31.25 | *** join/#asterisk Defraz (n=t0tal@tim.mychoice.cc) |
18:31.26 | bweschke | stse: what is it u need to know? |
18:31.57 | bweschke | RoyK: zaptel/ztdummy still loaded? |
18:32.03 | djMax | so a phone doesn't have to register to make an outbound? |
18:32.20 | need_sccp_help | Not necesarily |
18:32.48 | RoyK | bweschke: lol. doesn't matter... |
18:33.02 | RoyK | bweschke: the audio will suck if ztdummy isn't loaded, but meetme will work |
18:33.13 | RoyK | also, ztdummy _is_ loaded |
18:33.39 | justinu | djMax: not usually |
18:33.42 | *** part/#asterisk need_sccp_help (n=none@198.60.73.230) |
18:33.51 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
18:33.56 | djMax | so is there any way to force register, or just pull the plug? |
18:34.43 | puzzled | evening |
18:35.01 | justinu | djMax: depends on the phone |
18:35.04 | stse | bweschke: I'm subscribe to some channels I wish to monitor, but when someone calls this channel, asterisk sends a notifiy message telling me the channel is terminated instead of ringing. |
18:35.07 | djMax | k. Plug it is. |
18:36.22 | *** join/#asterisk WAudette70 (n=WAudette@c-67-170-156-3.hsd1.or.comcast.net) |
18:37.38 | stse | bweschke: besides this, how is the correct syntax for the hint extension? Sometimes I see "exten => number,hint,SIP/number", sometimes "exten => number1,hint,SIP/number2". |
18:38.03 | *** join/#asterisk my007mssrv (n=my@213.158.171.162) |
18:38.15 | my007mssrv | pleas some one help |
18:38.58 | *** join/#asterisk kkun (n=none@198.60.73.230) |
18:38.58 | [TK]D-Fender | stse : the first |
18:38.58 | sevard | Has anyone tried using Hamachi with SIP ? |
18:38.59 | my007mssrv | genzaptelconf # channel 1, WCTDM, inactive. |
18:39.31 | my007mssrv | when i run genzaptelconf config file have all chanel like this one # channel 1, WCTDM, inactive. |
18:39.58 | bweschke | stse: number is the extension the device is going to subscribe to. (eg - if you want to watch extension 1000, the number == 1000) |
18:40.04 | *** join/#asterisk dpolitech (n=Owner@207.224.48.130) |
18:40.21 | iCEBrkr | sevard: am I a ladies man?? WTF? |
18:40.28 | bweschke | state: the SIP/number SIP/number is the [number] in the sip.conf of the device that extensions 1000 belongs to in the dial plan |
18:40.51 | sevard | iCEBrkr: my gf's birthday is coming up, check this out, you know that once scene in patch adams with all the balloons? I'm going to fill her room full of balloons.. but that's all I got so far. |
18:41.05 | iCEBrkr | lol |
18:41.08 | justinu | damn |
18:41.09 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
18:41.11 | my007mssrv | hello all |
18:41.14 | justinu | be careful... |
18:41.16 | sevard | :) |
18:41.34 | stse | bweschke: thanks, wo what does the example "exten => 200,hint,SIP/201&SIP/202&SIP/203"? |
18:41.36 | justinu | some women get all greedy when you do shit like that for them |
18:42.03 | sevard | I'm going to borrow an aircompresser(-or?) to blow up all the balloons |
18:42.17 | justinu | or a tank of helium |
18:42.18 | [TK]D-Fender | stse : I don't think you can use multiple phones in one hint.... |
18:42.28 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
18:42.29 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
18:42.39 | sevard | tanks of helium are more expensive but a better option, i'll look into it |
18:42.41 | bweschke | stse: that lets you monitor the status of multiple extensions - but the problem is you're probably getting a terminate for the extension that didn't get the call when it rings |
18:42.52 | bweschke | stse: not advised for you to use it that way |
18:42.58 | sevard | justinu: you have any more awesome ideas? :) |
18:43.03 | justinu | heh |
18:43.19 | justinu | there should be a party supply store in your neighborhood |
18:43.22 | justinu | they can arrange it all ;) |
18:43.22 | my007mssrv | pleas someone help me i have error msg say that Notice: Configuration file is /etc/zaptel.conf |
18:43.22 | my007mssrv | line 0: Unable to open master device '/dev/zap/ctl' |
18:43.46 | sevard | justinu: there is but their services would empty my wallet and some |
18:43.53 | justinu | heh |
18:43.57 | sevard | that goes against the greedy policy |
18:44.04 | justinu | try renting a tank from them |
18:44.10 | stse | bweschke: okay, so I have the normal form, e.g. "exten => 401,hint,SIP/401", show hints and sip show subscriptions show thinks that looks correct for me. |
18:44.23 | bweschke | yep |
18:44.26 | bweschke | that looks right |
18:44.35 | sevard | that's what I had in mind. I'd probably be ~30 bucks and would have a better effect than air. |
18:44.51 | djMax | heh. So I hardcoded a host path, but now it's giving me errors because that host is trying to register. |
18:44.53 | *** join/#asterisk CMike (i=daemon@c-544171d5.116-1-64736c10.cust.bredbandsbolaget.se) |
18:44.54 | stse | bweschke: but I still get the terminated notfiy the first time I start asterisk, after this I get nothing anymore. |
18:44.58 | sevard | I was trying to do something awesome but all I could think of is that scene from patch adams, then i thought about the pool of noodles....nah. |
18:45.18 | iCEBrkr | "Excuse me, I need enough balloons to fill a room that's 10x12, can you do that for me? |
18:45.34 | justinu | how tall is the room? :P |
18:45.39 | iCEBrkr | oops |
18:45.42 | iCEBrkr | 10x10x12 |
18:45.44 | justinu | heh |
18:45.52 | [TK]D-Fender | Balloon size desired? |
18:45.56 | bweschke | stse: you will get a terminated sent when you reload asterisk. this is because the hint is going away and then coming back again. the phone should be smart enough to re-subscribe after having received the message but Polycoms and I believe Aastras are known not to. bug 6047 in the tracker deals with this |
18:46.00 | justinu | get the mylar ones |
18:46.08 | iCEBrkr | sevard: you're nutso |
18:46.09 | justinu | after the birthday, you can release them and take down your local power grid |
18:46.14 | sevard | I think it's a 9 foot celing, probably 8x8, but with desks and crap it couldn't take more than 40-60 baloons |
18:46.19 | sevard | balloons |
18:46.29 | sevard | mylar ones! hahahahahaha |
18:46.35 | iCEBrkr | BZzzzzzzzzot |
18:46.38 | stse | bweschke: hm, after I restart asterisk, I always restart the phones to make sure they are correctly subscribed. |
18:46.41 | [TK]D-Fender | bweschke : So Poly's going "sticky" is a bug on THEIR side? Hmmm... |
18:46.55 | sevard | This reminds me of Danny Deckchair |
18:46.57 | djMax | ok, so I see a SIP registration from my phone, with expires set to 3600. At the moment, sip show peers has my ip in it. |
18:47.18 | *** join/#asterisk sch19 (n=sch19@adsl-10-241-101.mia.bellsouth.net) |
18:47.59 | sevard | So, room-full-of-balloons is still the best idea? :) |
18:48.09 | bweschke | TKD-Fender - the termination message has an attribute in the msg to re-subscribe after 60 seconds. the poly's appear not to follow this instruction. We just got access to our tier 2 poly support through our reseller relationship with them and this will be one of the first questions we're going to ask about - because while 6047 kind of gets around the limitation, it shouldn't be "on Asterisk" to provide the workaround |
18:48.36 | stse | bweschke: then I call the test number and get the terminated notify. |
18:49.25 | bweschke | stse: we're going to need likely need a copy of your dialplan and a trace of what's going on to troubleshoot this further |
18:49.34 | *** join/#asterisk bkw_ (n=bkw_@adsl-69-104-16-79.dsl.irvnca.pacbell.net) |
18:50.35 | justinu | djMax: it should stay registered now that you've enabled the qualify (keep alive) |
18:50.51 | justinu | djMax: do you have a time value next to the sip peer? something in ms... |
18:51.11 | *** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com) |
18:51.14 | djMax | yeah, looking a bit better. "OK (83 ms)" |
18:51.30 | justinu | ok, that's good... it means your phone is responding to the keepalives in 83ms |
18:51.31 | djMax | but strangely, no other phones have registered. |
18:51.35 | sevard | Has anyone used Hamachi (or any VPN) with Asterisk? |
18:51.41 | stse | bweschke: with pleasure, how should I get them to you? |
18:51.45 | justinu | sevard: i'd by my woman something a bit more risque.... but that's just me ;) |
18:51.55 | bweschke | stse: bugs.digium.com |
18:51.58 | justinu | s/by/buy/ |
18:52.00 | sevard | justinu: I was thinking about that |
18:52.11 | justinu | i'm not sure how well you know this girl |
18:52.21 | sevard | well enough :) |
18:52.24 | bweschke | sevard: yes - we send calls w/Asterisk through Openvpn tunnels. works ok for the most part |
18:52.31 | justinu | maybe a night at a nice hotel w/ a jacuzzi tub |
18:52.38 | justinu | cover the bed in rose petals |
18:52.42 | justinu | chicks totally flip over thta |
18:52.46 | sevard | bweschke: I have Hamachi, a sort of VPN but I can't get my phone to register. |
18:53.05 | sevard | bweschke: I can do everything else over the VPN though, ssh, telnet, http, you name it. |
18:53.20 | justinu | can you ping the phone |
18:53.45 | sevard | justinu: see that's more of a thing "chicks dig" and that guys usually, i was thinking of weird shit to do, like a bath full of noodles or a room full of balloons :) |
18:53.46 | bweschke | sevard: is this an appliance providing a tunnel or is it software you're running on your PC and you're trying to use a softphone through it? |
18:53.55 | sevard | justinu: the asterisk box can ping the computer that the softphone is on |
18:54.03 | justinu | k |
18:54.10 | stse | bweschke: I don't want to open a bug if I don't know this is a bug. |
18:54.17 | *** join/#asterisk ToTo (n=ToTo@host61-133.pool874.interbusiness.it) |
18:54.37 | sevard | bweschke: Hamachi is a client-client VPN utility that you run on each PC, it sets up an IP for you on the 5. range and you then use it like a normal IP |
18:54.45 | bweschke | stse: I'm a bug marshal - if you assign it to me and it's not - I promise I won't give you neg karma :) |
18:54.50 | Katty | justinu: or sneeze all over it. |
18:55.06 | sevard | bweschke: using Hamachi I can do everything I want minus Asterisk... but it technically should be working. |
18:55.12 | justinu | katty: alergic to roses too? |
18:55.21 | Katty | justinu: thankfully, no |
18:55.44 | sevard | Katty: female input go |
18:55.52 | Katty | sevard: huh? |
18:56.00 | justinu | my mom calls her cat katty |
18:56.08 | justinu | a cat I gave her |
18:56.17 | justinu | i dunno if it has a real name |
18:56.20 | stse | bweschke: I know you are. I read it. ;-) Do I need a acoount? |
18:56.27 | sevard | Katty: Weird/random/awesome things a guy can do for a chick on her birthday, I'm already pumping helium filled ballons into every square of her room |
18:56.27 | bweschke | sevard: technically, yes. The problem with VPN clients though is that the way they generally work is to hack into the IP stack of the machine and generally you need to then make sure that it knows how to handle each protocol/application appropriately .. all the apps above you've cited are TCP and ICMP apps. are there any UDP apps that you know work well through it? |
18:56.37 | djMax | ok, seems better now. Still confuses me why setting an expire time on the phone that is less than the default expire time on * would cause this "no mans land interval" |
18:56.45 | justinu | djMax: NAT |
18:56.54 | djMax | But it's all on the same subnet |
18:56.58 | justinu | oh. |
18:57.06 | sevard | bweschke: Therein lies the problem. I don't have much experience using UDP and wouldln't know how to test UDP connectivity. |
18:57.21 | djMax | I mean, maybe it's something crazy dumb like I read seconds one place and msec another. |
18:57.27 | justinu | typically the problem you describe is caused by a NATing device closing it's binding after not seeing traffic |
18:57.38 | Katty | sevard: oh. |
18:57.47 | justinu | expires are in seconds, iirc |
18:57.50 | bweschke | sevard: well - Asterisk doesn't know how to do any protocols outside of UDP (and it shouldn't because RTP over TCP would likely suck)... |
18:57.52 | djMax | so normally that SIP session/registration is persistent? |
18:58.12 | djMax | i.e. it keeps a conn open? I mean, I can see why * would fail to reach, but not why it wouldn't have the addr in the peer list |
18:58.16 | justinu | djMax: if your phone is on the same LAN as the asterisk machine, you shouldn't need qualify |
18:58.34 | Katty | sevard: anywhere near a beach? |
18:58.37 | sevard | bweschke: Right, but I can't ..telnet.. to a UDP service or something so I wouldn't know if it was an Asterisk configuration issue or a VPN issue. |
18:58.43 | justinu | djMax: if you continue to have issues with registrations dropping off, try running a ping from the * server to the phone |
18:58.56 | Katty | sevard: actually, i'd seriously recommend a treasure hunt. |
18:59.03 | djMax | if that were the problem, wouldn't * still show the IP in the peer list? |
18:59.04 | justinu | lol, that's a great idea |
18:59.07 | sevard | Katty: Yes, Minnesota, land of 10,000 lakes.. except this time of year they have a good 2 foot layer of ice. |
18:59.10 | Katty | sevard: i know it sounds childish, but you could put a little twist on it. |
18:59.24 | Katty | sevard: 5 notes around the house, and then one that says to go to a resturant |
18:59.33 | justinu | djMax: if your phone doesn't re-register in time, asterisk will remove the IP from the peer list |
18:59.41 | Katty | sevard: and then secretly stash something in a purse or a pocket |
18:59.47 | sevard | Katty: Like what? The idea had come across my mind and I put it off to the side because I did that with one of my old girlfriends (who my current is quite jelous of) and because I coudln't think of any neat twists |
18:59.56 | bweschke | sevard: the question you have for the VPN support folks is "does your vpn support SIP over UDP and RTP over UDP"? because DNS is a UDP application, but you can be damn sure they've figured out how to make that work.. but not necessarily true for VoIP apps |
19:00.01 | djMax | yeah, that seems the best explanation, even though I set the timeout lower. |
19:00.01 | justinu | djMax: typically the only reason that might happen is NAT, or a network connectivity issue which stops the register messages from the phone getting to * |
19:00.06 | Katty | sevard: well the little notes could each be a little present. |
19:00.16 | Katty | sevard: note 1, look in bedroom. |
19:00.21 | djMax | I guess now the question is whether qualify=yes is going to somehow make it go away |
19:00.25 | Katty | sevard: and the bed could be covered in flowers. |
19:00.29 | Katty | sevard: and another note on the bed. |
19:00.35 | Katty | sevard: and then note 2 could say look in bathroom |
19:00.40 | Katty | sevard: and you could have the whole spa thing going on |
19:00.44 | sevard | Katty: Heh. |
19:00.45 | Katty | sevard: but it'd be all for later |
19:00.56 | Katty | sevard: by the time you get to note 5, it says go to $resturant |
19:01.01 | *** join/#asterisk sch19 (n=sch19@adsl-10-241-101.mia.bellsouth.net) |
19:01.06 | _Paulo_ | gold is the shortest path to the heart of a woman |
19:01.11 | Katty | _Paulo_: no it's not |
19:01.15 | puzzled | diamonds |
19:01.16 | salviadud | paulo, no way man |
19:01.19 | salviadud | chocolate |
19:01.22 | sevard | That'd be nice... one twist, we're both college students and she's migating the cost of a dorm by staying with realitives... |
19:01.23 | Katty | _Paulo_: there are plenty of girls out there that don't want jewelry |
19:01.25 | salviadud | its the black gold |
19:01.29 | justinu | gold is the shortest path to the heart of the wrong woman |
19:01.46 | salviadud | yeah, i agree, some women like nintendo |
19:01.47 | Katty | sevard: do you live with your parents? |
19:01.49 | tronix | sevard: one way to test udp: # nmap -P0 -sU -p <port> <host or IP> |
19:01.51 | salviadud | i love those women... |
19:01.59 | justinu | i'd rather hang out with a woman who likes video games |
19:02.03 | Katty | salviadud: glad to know i'm loved. |
19:02.09 | sevard | Katty: I live in ground level apartment with my room mate, but my room mate is disabled and never leaves the house. |
19:02.22 | Katty | sevard: you could include him! |
19:02.23 | salviadud | katty, whats your fav nintendo game? |
19:02.26 | Katty | sevard: make him stash a note |
19:02.29 | sevard | Katty: wtf no 3way |
19:02.34 | Katty | salviadud: hmm, though one.........zelda gold or rampage... |
19:02.39 | Katty | salviadud: possibly contra |
19:02.45 | salviadud | contra is badass |
19:02.48 | justinu | zelda: a link to the past |
19:02.50 | Katty | sevard: nonono |
19:02.51 | justinu | but that was SNES |
19:02.54 | asteriskmonkey | rampage and super spike voly ball .. the orginal contra kicks ass :D |
19:02.54 | mut | anyone know anythign about tellab echo can shelvs? |
19:02.58 | Katty | sevard: one note could say See $person |
19:03.02 | mut | how large is a 255d shelf? |
19:03.08 | Katty | sevard: and $person could be secretly stashing a note |
19:03.15 | Katty | sevard: and that note could be the next clue |
19:03.20 | justinu | mut: http://www.voip-info.org/wiki/view/Tellabs+Hardware+Echo+Cancellers |
19:03.22 | Egonis | I have a SIP Phone connected externally, and they cannot hear me, nor transmit any audio, but their phone rings -- I followed a howto found on google, but no luck yet |
19:03.25 | Katty | sevard: it's a birthday party......treasure hunts don't have to be person |
19:03.32 | sevard | OH SNAP speak of the devil that was her on the phone just now, she's bringing me soup for lunch |
19:03.36 | sevard | she's (##(&ing awesome |
19:03.36 | Katty | sevard: but i suppose you could always stash a note..uh..somewhere...person |
19:03.38 | Katty | a; |
19:03.41 | Katty | s/a;/al |
19:03.48 | mut | justinu: um? that says nothin about em |
19:03.52 | justinu | sevard: nothing like the love of a good woman |
19:04.01 | sevard | justinu: this one's a keeper. |
19:04.09 | justinu | mut: are you blind? |
19:04.10 | Katty | sevard: treat her well. |
19:04.20 | justinu | mut: look under the heading "models of shelves" |
19:04.20 | mut | i must be |
19:04.21 | mut | show me |
19:04.34 | sevard | Katty: I've really not been, that's why I want to make this birthday special. |
19:04.43 | mut | i don't see 255D info listed |
19:04.46 | mut | i see 255A there |
19:04.47 | WAudette70 | Katty: Good ideas |
19:04.52 | mut | thats it |
19:04.56 | *** join/#asterisk oogle (n=jart@justin.ctlinc.com) |
19:05.02 | WAudette70 | sevard: Good luck! Go all out on the keepers. |
19:05.05 | sevard | Katty: thanks for the input, very good ideas and I'm going to put them to use. |
19:05.08 | oogle | are there any good services like PasteBin that are read-only and don't delete your code after a day? |
19:05.23 | *** join/#asterisk virterm (n=virterm@shiva.kanatek.com) |
19:05.51 | mut | am i missing something justinu? |
19:06.06 | sevard | WAudette70: thanks bro:) |
19:06.16 | Katty | sevard: excellent (= |
19:06.16 | justinu | i guess it doesn't have the details you're looking for |
19:06.24 | Egonis | exit |
19:06.28 | mut | yea.. like.. ANY |
19:06.32 | justinu | however, I believe the 255 is a 16 slot shelf |
19:06.35 | sevard | bweschke: you still here? |
19:06.38 | mut | k |
19:06.42 | sevard | I did 5060/udp open unknown |
19:06.43 | justinu | w/ removeable alarm and access cards |
19:06.45 | bweschke | sevard: yes |
19:06.45 | sevard | erm |
19:06.53 | oogle | pastebin doesn't seem to like me |
19:06.57 | sevard | I did nmap -P0 -sU -p 5060 <vpn ip> and I got 5060/udp open unknown |
19:06.58 | mut | possible it's a 19" rack mountable shelf? |
19:07.31 | justinu | telco stuff is generally 21" |
19:07.35 | justinu | so I wouldn't bet on 19 |
19:07.39 | bweschke | what about 5061 and 10000-20000 for the RTP or whatever port numbers the softphone wants to use? |
19:07.46 | mut | yea |
19:08.15 | Katty | muttly. |
19:08.16 | bweschke | sevard: try a IAX based softphone |
19:08.21 | WAudette70 | sevard: I took that advise a year ago and she ended up marying me. She still raves about our special dates and she still wants more of them. |
19:08.33 | Katty | muttly is a good name for a dog. |
19:08.47 | justinu | muttly is the name of a dog from a cartoon |
19:08.52 | justinu | smurfs? |
19:08.52 | Katty | oh, is it? |
19:08.57 | justinu | i think it was smurfs |
19:09.05 | Katty | windshield wiper fluid = smurf juice. |
19:09.07 | bweschke | sevard: you'll know straight off if it's an issue with the vpn/firewall or whether it's something else - as iax requires only one udp port/socket to be opened |
19:09.26 | iDunno | muttly is from Dick Dastardly cartoons - such as Wacky Races and Catch The Pigeon |
19:09.29 | tronix | isn't iax 4569? |
19:09.32 | sevard | 5061/udp closed unknown, isn't 5060 used for registration? |
19:09.51 | tronix | oh.. sevard's doing sip right now. gotcha |
19:09.54 | iDunno | the dog in the smurfs was *not* called muttly. |
19:09.57 | Katty | iDunno: oh. |
19:09.58 | bweschke | sevard: yes - on the server... but this is a phone - not a server, right? |
19:10.08 | Katty | iDunno: was this an old thing? |
19:10.24 | sevard | bweschke: I'm nmaping the server |
19:10.27 | iDunno | (and it was actually muttely, but feh ;) |
19:10.28 | bweschke | sevard: some phones receive sip signalling for calls on 5061 - (eg the SPA3000) |
19:10.30 | sevard | bweschke: From the phone |
19:10.35 | iDunno | Katty: well, reasonably, 80s IIRC |
19:10.39 | iDunno | http://www.hotink.com/wacky/dastrdly/ |
19:10.40 | sevard | bweschke: the phone i'm using is an x-lite |
19:10.42 | justinu | that's right.. dastardly and his dog muttly |
19:10.47 | Goral | do i actually put in a ip or leave this like this? bindaddr = 0.0.0.0 |
19:10.57 | Goral | the address i'm using is static |
19:11.06 | salviadud | leave it like that dude |
19:11.11 | *** part/#asterisk WAudette70 (n=WAudette@c-67-170-156-3.hsd1.or.comcast.net) |
19:11.19 | Katty | iDunno: oh, that's not too old. |
19:11.26 | iDunno | rocked :) |
19:11.37 | Goral | so it will bind to any ip? |
19:11.46 | salviadud | exactly |
19:11.46 | iDunno | oh, erm, says 1969 ;) |
19:11.53 | iDunno | so maybe I grew up to reruns of that :) |
19:11.54 | Goral | salviadud ty |
19:12.23 | *** join/#asterisk NotFreak (i=NotFreak@cp12193-e.tilbu1.nb.home.nl) |
19:12.26 | bweschke | sevard: well |
19:12.33 | tronix | I seem to vaguely remember some DIck Dastardly cartoons ran with Banana Splits |
19:12.33 | *** part/#asterisk NotFreak (i=NotFreak@cp12193-e.tilbu1.nb.home.nl) |
19:12.38 | justinu | http://www.hotink.com/wacky/dastrdly/ |
19:12.43 | *** join/#asterisk NotFreak (i=NotFreak@cp12193-e.tilbu1.nb.home.nl) |
19:12.46 | bweschke | sevard: try doing a sip debug and see if the registration ever comes in from that IP - if so - you know it's not the VPN - |
19:13.11 | tronix | Banana Splits was an interesting cartoon. one was a completely psychedelic episode... kids thought it was fun, adults knew the symbolism. ;) |
19:13.30 | fu3 | with loopstart, that battery drop that occurs.. should it drop the line to 0VDC for those 500ms? |
19:13.32 | sevard | bweschke: alright, i'll do that, my gf is here, bbiab :) |
19:13.59 | bweschke | sevard: good call - * isn't your priority right now - or at least it shouldn't be. :) |
19:14.32 | bweschke | i've always got to remind myself of that when my wife and/or kids are looking for me. |
19:14.40 | [av]bani | yay new gxp firmware |
19:14.41 | CMike | If I want to set up a simple prepaid solution using mysql .. what should I use ? I see a lot of different "prepaid-solutions" any suggestion which I should take a look at ? |
19:14.58 | CMike | (ANI based) |
19:15.03 | justinu | bani: any info? |
19:15.13 | [av]bani | ? |
19:15.27 | justinu | on the new firmware |
19:15.29 | [av]bani | ~gxp2000 |
19:15.29 | jbot | rumour has it, gxp2000 is http://www.voip-info.org/wiki/view/GXP-2000 |
19:15.51 | justinu | cool |
19:16.22 | justinu | added support for DHCP option 66! |
19:16.24 | justinu | finally |
19:16.39 | [av]bani | it defaults 0 though :( |
19:16.45 | [av]bani | but meh, better than nothing |
19:17.14 | [av]bani | not that it really matters, i can provision them out of the box :) |
19:23.02 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
19:23.02 | *** topic/#asterisk is Zaptel 1.2.4 Released ... More information available on http://www.asterisk.org |
19:23.11 | djMax | is there some way to tell it never to lose registrations, and if so is that safe? |
19:23.38 | justinu | not really |
19:23.38 | GerbilWrk | if you have an audio file playing, what do you put in to have it ignore key presses? |
19:23.44 | [av]bani | qualify=no |
19:23.50 | justinu | djMax: what happens when you run a continuous ping on the phone? |
19:23.56 | *** join/#asterisk bkw_ (n=bkw_@mc11536d0.tmodns.net) |
19:23.59 | *** join/#asterisk vgster (n=vg@spc1-ledn1-3-0-cust194.seac.broadband.ntl.com) |
19:26.14 | DarthClue | GerbilWrk, how are you playing the file? |
19:26.15 | djMax | no hiccups |
19:26.21 | ManxPower | Anyone here familiar with Nortel? I need information on what to dial from a Nortel (Meridian?) analog line to access call pickup and all-station-page |
19:26.31 | *** join/#asterisk kkun (n=none@198.60.73.171) |
19:26.32 | GerbilWrk | exten => 1,1,Background(thevoice/mailing) |
19:27.04 | djMax | no hiccups |
19:27.07 | DarthClue | use Playback instead of Background |
19:27.10 | ManxPower | djMax, host=1.2.3.4 and don't have the phone register |
19:27.12 | GerbilWrk | ok |
19:27.22 | kkun | Anybody familiar with sccp? |
19:27.26 | djMax | sorry. Wrong up arrow window. So the problem seems to be some timing mismatch in re-registration. |
19:27.42 | ManxPower | kkun, use SIP |
19:27.47 | justinu | djMax: are you sure? |
19:27.48 | kkun | Can't |
19:27.50 | djMax | I set the expire time down to 100 seconds, now I'll see if the problem happens faster or what. |
19:27.59 | kkun | Using 12sp phones, they don't support sip |
19:28.19 | djMax | if this is really working on the polycom, I guess I should see reg messages every 100 seconds... |
19:28.24 | ManxPower | ah. I'm sorry to hear that. SCCP is just miserable |
19:28.43 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
19:28.53 | justinu | djMax: the registrar responding to the register messages can override the client preferred value |
19:28.56 | kkun | I know, any ideas / suggestions on how to setup sccp.conf? |
19:29.10 | djMax | that is controlled by defaultexpiry? |
19:29.15 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
19:29.21 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
19:29.37 | justinu | i think there's a few other params... minexpiry also? |
19:30.10 | djMax | yeah, min, default, max. |
19:30.27 | kkun | manxpower, the few 49xx phones I have are doing sip beautifully, I wish I could get sip firmware for the 12sp |
19:30.39 | Qwell[] | 49xx? |
19:30.42 | Qwell[] | 79xx? |
19:30.49 | kkun | Right, 79xx |
19:30.56 | Qwell[] | I just use sccp...it's pretty nice |
19:30.58 | djMax | interesting, no registrations from the phone. |
19:31.01 | Qwell[] | 12sp+ is well supported I hear |
19:31.02 | djMax | (except for the first one) |
19:31.20 | kkun | qwell, can you help me with sccp.conf for a 12sp? |
19:31.34 | Qwell[] | kkun: should be pretty easy... |
19:31.59 | kkun | I can get the thing to register, and ring, but they won't dial |
19:32.10 | Qwell[] | kkun: need to put them into a context with a dialplan |
19:32.41 | kkun | The context I specify in sccp.conf is the same default context as my sip phones, which work |
19:32.52 | djMax | I guess stopping registration is the simplest fix. Annoying though. Especially because we never had this problem with * 1.0. |
19:33.00 | Qwell[] | kkun: in the lines section? |
19:33.16 | kkun | ? |
19:33.22 | kkun | Line section? |
19:33.29 | Qwell[] | context is in the [lines] section of sccp.conf? |
19:33.57 | kkun | Ok, I though it was in the general section |
19:34.17 | Qwell[] | that could work, but anything in [lines] overrides [general] |
19:34.41 | GerbilWrk | DarthClue, that worked, thanks |
19:35.06 | *** join/#asterisk rtikk (n=rtikk@g220032.upc-g.chello.nl) |
19:35.28 | kkun | I have it in the lines section as well, still no dice. |
19:35.44 | Qwell[] | and you restarted asterisk after making the change? |
19:35.50 | justinu | djMax: what kind of phones? |
19:36.02 | kkun | As always |
19:36.19 | kkun | Let me show you my sccp.conf |
19:36.25 | Qwell[] | pastebin it |
19:36.27 | Qwell[] | ~pb |
19:36.28 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
19:36.47 | Qwell[] | and show me the CLI output when you try to dial |
19:38.23 | kkun | http://pastebin.com/567253 |
19:39.11 | Qwell[] | kkun: Which line/device is it? |
19:39.17 | jbalcomb | konyagayamada |
19:39.22 | Qwell[] | ahh, 130 |
19:39.33 | Qwell[] | okay, near the bottom where you have "line => 130" |
19:39.53 | Qwell[] | either change that to "line => line130" or change the autologin= at the top to "130" instead of "line130" |
19:40.00 | kkun | Cli shows that asterisk is receiving the numbers dialed, but not "dialing" it |
19:40.33 | Qwell[] | because there is no active line... |
19:40.41 | Qwell[] | at least...not a proper one |
19:40.56 | Qwell[] | the device shouldn't even be fully loading |
19:41.15 | kkun | Ok, just a second |
19:42.05 | *** join/#asterisk epablo (n=epablo@200.75.139.188) |
19:42.19 | epablo | hi people. |
19:43.25 | kkun | qwell, rebooting |
19:43.33 | Qwell[] | rebooting? |
19:43.36 | Qwell[] | the phone, I hope |
19:44.50 | epablo | i'm having some problems getting a T1 to work.. anyone have 5 min to give me some pointers? Here is the conf i'm using |
19:45.12 | *** join/#asterisk fugitivo (n=ajf@201.255.177.92) |
19:45.18 | jbalcomb | anyone have an archive of the firmware releases for the GXP-2000? |
19:46.43 | kkun | qwell, thanks, that did it, I knew it had to be something stupid, 1 line of code. |
19:47.10 | fu3 | can anyone tell me about the 500ms battery drop used with loopstart signalling? |
19:47.18 | fu3 | to indicate a hangup. |
19:47.50 | fu3 | more specifically, my lines operate at around ~40VDC. This 500ms "drop" -- does that drop it to 0VDC for those 500ms? |
19:49.14 | *** part/#asterisk bkw_ (n=bkw_@mc11536d0.tmodns.net) |
19:51.10 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
19:51.50 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
19:52.15 | sigmounte | hello ! what apps can i use under linux to record my own .gsm files ? |
19:53.05 | hensema | 'evening |
19:53.06 | Nivex | sigmounte: Don't know of anything that writes them natively. I would use something like Audacity to record, save as .WAV and then use sox to convert. |
19:53.26 | sigmounte | thanks you Nivex ! |
19:53.30 | fu3 | i'm surprised * hasnt included gsm recording. |
19:53.55 | epablo | You can do it.. setting up an ivr |
19:53.59 | fu3 | ahh |
19:54.08 | hensema | I've got an ISDN-2 with multiple inbound numbers connected to asterisk; calls are routed to a SIP phone (reception desk). Is it possible to tell on the SIP phone on what number the PBX was called? |
19:54.24 | sigmounte | i'm setting up an ivr by hand configuring my .conf , their is another way to do it ? |
19:54.30 | hensema | so, in effect a reverse-CID: I want _our_ number to be displayed |
19:54.56 | epablo | sigmounte: I think not nativly.. record and convert with sox |
19:55.30 | *** join/#asterisk W8TAH (n=Tim@static-acs-24-239-210-31.zoominternet.net) |
19:55.33 | *** join/#asterisk JoanFrantzisko (n=jfguarda@200.72.212.133) |
19:55.46 | JoanFrantzisko | hello everyone |
19:55.50 | fu3 | HI! |
19:55.58 | JoanFrantzisko | i've a issue with Asterisk behind a firewall |
19:56.04 | fu3 | you'll get over it |
19:56.04 | fu3 | :) |
19:56.07 | JoanFrantzisko | can anybody help me? |
19:56.25 | fu3 | i would but im just starting into asterisk myself.. sorry |
19:56.34 | fu3 | there ARE people here :) |
19:56.36 | ManxPower | JoanFrantzisko, the wiki is your friend |
19:56.36 | Qwell[] | JoanFrantzisko: ONly if yo ask a question |
19:56.38 | ManxPower | ~docs |
19:56.39 | jbot | docs is, like, probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
19:56.44 | Qwell[] | I botched that one... |
19:56.44 | fu3 | ask the question |
19:57.04 | JoanFrantzisko | ok |
19:58.16 | W8TAH | Good Afternoon everyone: I was just introduced to asterisk this afternoon and am beginning the process of investigating it for implementation in our school building -- Is it possible to create a standalone asterisk server for Only internal use, without it being connected to the outside world? |
19:58.42 | epablo | yes.. block it on the network |
19:58.49 | JoanFrantzisko | when i call of the Internet, the receiver can hear me, but i can't hear the receiver |
19:58.58 | Nivex | W8TAH de N8VNR: absolutely. It doesn't care what IPs it's talking to, internal or external. |
19:59.01 | ManxPower | JoanFrantzisko, that is correct and expected. |
19:59.25 | W8TAH | ok -- cool -- what clients are recomeneded to connect to such a server? |
19:59.27 | ManxPower | unless you do your basic homework about running Asterisk behind NAT, as is documented on the mailinglists, Wiki, various books and messages. |
20:00.02 | epablo | W8TAH: you can start with a softphone.. xten-lite is good and free |
20:00.17 | W8TAH | epablo, excellent -- thank you |
20:00.19 | ManxPower | hensema, SetCIDNum(${EXTEN}) |
20:00.34 | W8TAH | Nivex, Cool -- thanks |
20:00.52 | *** join/#asterisk bkw_ (n=bkw_@m788636d0.tmodns.net) |
20:00.57 | W8TAH | epablo, do you have any experience setting it up on Gentoo? -- i know it is in portage |
20:01.03 | ManxPower | ASSUMING it's a PRI |
20:01.04 | JoanFrantzisko | ManxPower: if we are use SIP Protocol, all is OK, but the problem is when we use H-323 |
20:01.22 | JoanFrantzisko | ManxPower: the Firewall is Astaro Security Gateway |
20:01.24 | ManxPower | JoanFrantzisko, I don't believe that you can put Asterisk behind NAT if you are running H323 |
20:01.28 | epablo | W8TAH: Nop.. i'm more of a RH guy,.,.but it should work fine |
20:01.57 | W8TAH | epablo, ok -- |
20:02.21 | djMax | justinu: polycom IP500 phones |
20:02.33 | W8TAH | epablo, 2 final quesitons then if i may, i am having trouble finding specs on the server machine - -I may have missed them, and secondly -- what if any gui does it prefer? (kde, gnome, fluxbox etc) |
20:02.43 | djMax | heh, this is just plain weird. |
20:03.07 | djMax | So I see reg requests from the poly "frequently," but not frequently enough to avoid * kicking it out before it reregisters. |
20:03.24 | djMax | it's almost like one of their clocks is "very" fast |
20:03.25 | JoanFrantzisko | ManxPower: H323 can't run behind a NAT? because Astaro include open h323 kernel module |
20:03.29 | ManxPower | djMax, Dunno. I have over 60 polycoms and have none of these problems. |
20:03.30 | *** join/#asterisk exstatica (i=exstatic@aboutmylife.net) |
20:03.39 | ManxPower | JoanFrantzisko, No, it's because of RTP |
20:04.13 | ManxPower | H323/NAT puts the IP address/port information in the data portion of the voice packet and signalling packets, so normal NAT won't work |
20:04.17 | epablo | W8TAH: to play and make it work you can use almost anything.. a P3 should be fine. there is a astgui.. but you could always use amp (web based interface) I recommend learning to do it by hand the first time |
20:04.27 | fu3 | you could do H323 with PAT. |
20:04.30 | fu3 | I think :) |
20:04.32 | ManxPower | also usually the SIP or H323 proxies in firewalls are for CLIENTS behind NAT, not for SERVERS behind NAT. |
20:04.48 | djMax | ManxPower, I've never had these problems either. Just after * 1.2 |
20:05.09 | W8TAH | epablo, thanks a bunch -- looks like i got a gentoo box to build -- LOL |
20:05.12 | ManxPower | djMax, I've been running 1.2RC for a while and 1.2.4 for a week. |
20:05.13 | JoanFrantzisko | ManxPower: what can i do? |
20:05.16 | ManxPower | no known problems |
20:05.20 | ManxPower | JoanFrantzisko, don't use H323 |
20:05.23 | epablo | W8TAH: Have fun! |
20:05.25 | ManxPower | or use a different IP PBX |
20:05.28 | W8TAH | will do |
20:05.33 | ManxPower | or write support for it in asterisk. |
20:06.23 | djMax | guess I could just update and such |
20:06.28 | cpm | Got one of them thar fancy x100p 'VoIP 1' fxs boxes in today. |
20:06.48 | cpm | I'm sorry to say, it beats the pants off of the IAXy. |
20:06.52 | djMax | is registration (vs static IP) better/worse? |
20:07.03 | ManxPower | djMax, Neither. |
20:07.05 | JoanFrantzisko | ManxPower: this issue with H323 behind a Firewall is caused for a bug in the linux kernel? or Asterisk not support the implementation? |
20:07.11 | Mavvie | is Corydon76 on this channel? |
20:07.27 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
20:07.30 | Katty | Corydon-w: Mavvie |
20:07.36 | ManxPower | JoanFrantzisko, I'm sorry, I cannot help you further. |
20:07.41 | ManxPower | But Asterisk on a public IP |
20:08.07 | Corydon-w | What? |
20:08.25 | Katty | Mavvie: Corydon-w |
20:08.30 | fu3 | ManxPower.. you're the best! |
20:08.30 | Mavvie | Corydon-w: can I discuss the closure of 6565 with you? |
20:08.35 | ManxPower | fu3, why? |
20:08.38 | fu3 | just because. |
20:08.43 | fu3 | you actually try. |
20:08.46 | fu3 | it's refreshing. |
20:08.49 | Corydon-w | Mavvie: sure |
20:09.08 | ManxPower | cpm, it runs IAX2? |
20:09.16 | cpm | Yup. That's all it runs in fact |
20:09.21 | Corydon-w | Mavvie: try the #asterisk-bugs channel |
20:09.21 | cpm | no more SIP |
20:09.34 | justinu | can you put bootrom 3.1 on an IP 301? |
20:09.46 | MstlyHrmls | yup |
20:09.55 | justinu | for some reason it's not picking it up |
20:10.10 | justinu | it grabbed the sip.ld and loaded that, but not bootrom.ld |
20:10.18 | MstlyHrmls | justinu: is it giving an error? |
20:10.21 | justinu | nope |
20:10.37 | MstlyHrmls | justinu: does it say anything in the <mac>-boot.log? |
20:11.01 | justinu | ah yes, it does |
20:11.10 | cpm | don't care for the cli on this box, I mean not at all. |
20:11.14 | jbalcomb | anyone have an archive of the firmware releases for the GXP-2000? |
20:11.24 | *** join/#asterisk PakiPenguin (i=uppal@linuxpakistan/admin/pakipenguin) |
20:11.27 | PakiPenguin | hello everyone |
20:11.35 | epablo | <PROTECTED> |
20:11.43 | PakiPenguin | anyone here uses astguiclient? |
20:11.52 | PakiPenguin | i need some quick help please |
20:11.57 | justinu | MstlyHrmls: http://pastebin.ca/42806 |
20:12.12 | MstlyHrmls | justinu: looking |
20:12.23 | justinu | guess it doesn't like what i'm trying to load on it |
20:12.31 | *** join/#asterisk saftsack (n=saftsack@p54A7DAAD.dip.t-dialin.net) |
20:12.38 | justinu | i used that bootrom.ld file on my ip501/601s without trouble |
20:12.50 | MstlyHrmls | justinu: how big is your bootrom.ld file, and what version of 3.1 is it? |
20:13.06 | saftsack | hi |
20:13.07 | justinu | size is 422925 |
20:13.13 | sigmounte | do the .gsm file have a special format to follow when creating some myself ? |
20:13.46 | MstlyHrmls | justinu: that seems small, let me have a look at mine |
20:14.04 | justinu | not sure of the exact version |
20:14.16 | jbalcomb | epablo: is that 1.0.2.13 and 1.0.2.12? |
20:14.27 | MstlyHrmls | justinu: my copy of 3.1.2 is ~2.5 megs... |
20:14.32 | justinu | interesting. |
20:14.46 | MstlyHrmls | justinu: it looks like you've got an incomplete version somehow |
20:14.48 | justinu | my version might only include the bootloader code for the 501/601 phones |
20:15.00 | justinu | interestingly enough, my copy of rom 3.0.1 is 2.1 meg |
20:15.01 | MstlyHrmls | justinu: that's what I'm thinking |
20:15.39 | saftsack | is there a possibility to log and archive all outgoing faxes from my normal fax machine which is connected to an tdm fxs card? |
20:16.01 | MstlyHrmls | justinu: if you have a hex editor, or a text editor that can handle binary files, you can open up the bootrom.ld file, and the version should be in the first 100 or so bytes of the file |
20:16.08 | ManxPower | saftsack, no |
20:16.22 | justinu | do I have to ask my resller for the correct bootrom, or can I download it somewhere? |
20:16.32 | ManxPower | unless you want asterisk to itercept the call, accept the fax (using rxfax), then retransmit the fax (using txfax) |
20:16.33 | JoanFrantzisko | ManxPower: ok, i got it... i will try with H323 connected directly to the internet, and SIP behind the Firewall... thnxs :) |
20:16.39 | ManxPower | and really, that's just asking for trouble |
20:16.49 | bweschke | justinu: yes - any version of the X01 should have memory enough to support the later bootroms |
20:17.19 | MstlyHrmls | justinu: techncially you have to ask your reseller, but I'm sure if you asked around... |
20:17.37 | justinu | actually, hang on |
20:17.39 | justinu | i might have this somewhere |
20:17.49 | justinu | too many files, too many machines |
20:18.02 | MstlyHrmls | bweschke: actually I've been able to load 3.x on all the Polycoms I have, including 300s and 500s |
20:18.39 | bweschke | MstlyHrmls: that's not supposed to work because it's not supposed to have memory enough to support HTTPS/HTTP provisioning, but i've not tried it myself |
20:19.04 | MstlyHrmls | bweschke: the 300s and 500s won't do the HTTP, but they use the rest of 3.x just fine |
20:19.15 | epablo | jbalcomb: i got GXP2000_Release_1.0.1.9.zip, Release_1.0.2.16.zip |
20:19.20 | bweschke | current bootrom is 3.1.3 - current fw is 1.6.5 |
20:19.23 | [av]bani | tis very silly polycom wont put http on the 501 |
20:19.38 | [TK]D-Fender | bweschke : I though it was more because that once you DID load 3.x that it wouldn't have enough FREE memory to load a downgrade... |
20:19.55 | bweschke | TKD-Fender: yes that's true |
20:20.07 | [av]bani | epablo: 1.0.2._16_ ?? |
20:20.08 | MstlyHrmls | [TK]D-Fender: doesn't matter what you load a 3.x on to, you can't downgrade to 2.x |
20:20.14 | jbalcomb | epablo when did 1.0.2.16 come out? the wiki only has .13 Can you email them to me? PM for address |
20:20.21 | justinu | ok, i have sip 1.6.3 (which is probably fine) |
20:20.28 | justinu | but no bootrom 3.1.3 |
20:20.37 | justinu | anyone wanna help a brutha out? |
20:20.38 | bweschke | from the "WARNING" with 3.1.13: We recommend upgrading to the 3.1.x BootROM ONLY if you intend to use the security features in the SIP 1.6.x application. For all other customers, we recommend that you continue to use the 2.6.2 version of the BootROM. |
20:20.42 | jbalcomb | [av]bani how is you auto-provisioning project coming? |
20:20.51 | [av]bani | jbalcomb: :D |
20:20.58 | [av]bani | coming along nicely, i got the grandstream stuff figured out |
20:21.16 | justinu | bweschke: yeah, i'm aware of that... i like the 3.x roms because of http provisioning |
20:21.25 | epablo | http://www.grandstream.com/BETATEST/ |
20:21.34 | jbalcomb | [av]bani i have my personal phone being configured through TFTP and I think I'm digging it. |
20:21.38 | epablo | There is were I got the new ones |
20:21.47 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
20:21.53 | jbalcomb | epablo: they only have 1.0.2.3 there |
20:22.07 | saftsack | ManxPower, or maybe it is possible to log the audio on the zapchannel where the fax is connected and in the evening when nobody is in the buero sending the audio files as calls to the hylafax, faxgetty (i have a faxmodem)? |
20:22.21 | [av]bani | jbalcomb: mine does via php, and you can do it totally out of the box, zero config. |
20:22.30 | epablo | jbalcomb: let me make sure that zip is for gxp |
20:22.34 | jbalcomb | [av]bani do you have anything you can share with me? |
20:22.38 | [av]bani | :D |
20:22.49 | [av]bani | maybe when i clean it up |
20:23.08 | jbalcomb | [av]bani I have 120 of those phones so I could certainly appreciate it and provide some additional feedback |
20:23.16 | epablo | jbalcomb: You are right it's for other phones |
20:23.26 | [av]bani | i thought you said you were fine with tftp and manual config, and pooh-poohed my script :) |
20:23.30 | jbalcomb | epablo: ok, thats cool. good lookin' out though. |
20:24.09 | jbalcomb | [av]bani i said it wasn't a that big of a hassle given the notion of having to configure every phone |
20:24.59 | jbalcomb | [av]bani i think the biggest joy i can figure on the auto config is matching dept -> username -> ext -> ip -> mac and having different configs |
20:25.19 | jbalcomb | [av]bani for each dept and only updating the firmwares a dept. at a time. |
20:25.34 | MstlyHrmls | [av]bani: actually I have a 501 here on my desk that's using HTTP |
20:25.50 | ManxPower | saftsack, That's too twisted for me to think about. Best of luck. |
20:26.42 | *** join/#asterisk crich1999 (n=crich@port-212-202-0-5.dynamic.qsc.de) |
20:26.47 | [av]bani | jbalcomb: i'm trying to come up with a method of auto-generating extension#'s from mac addresses |
20:26.52 | *** join/#asterisk darby_t (i=darby_t@dkg24.neoplus.adsl.tpnet.pl) |
20:26.53 | [av]bani | jbalcomb: so you dont have to assign extensions either |
20:27.01 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
20:28.10 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
20:28.51 | jbalcomb | [av]bani sounds awesome |
20:29.59 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
20:30.05 | *** join/#asterisk xtrvd (n=j@d209-121-36-44.bchsia.telus.net) |
20:30.41 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
20:33.06 | austinnichols101 | [av]bani: got an initial response from InterNap |
20:33.15 | [av]bani | austinnichols101: and? |
20:33.21 | [av]bani | "we are looking into it"? |
20:33.27 | austinnichols101 | [av]bani |
20:33.35 | austinnichols101 | [av]bani: more or less. |
20:33.35 | [av]bani | austinnichols101 |
20:33.42 | [av]bani | nice empty response |
20:33.53 | [av]bani | they should have been looking into it 2 years ago |
20:34.12 | [av]bani | we'll see if it's all talk |
20:34.25 | austinnichols101 | [av]bani: check separate window for a copy of the message |
20:35.25 | austinnichols101 | [av]bani: the response is from one of my normal support contacts. I also spoke with her and she indicated that she would follow up and get back to me with more info so the message isn't the last word on the subject. |
20:35.43 | ManxPower | Does anyone have a recommended vendor for RJ21X (Amphenol) cables? |
20:35.54 | bweschke | Manxpower: Graybar |
20:35.56 | austinnichols101 | [av]bani: but you're right that the response is weak |
20:36.22 | ManxPower | I was hoping to avoid them. Their website ordering system is horrid |
20:36.34 | ManxPower | actually their online catalob is horrible |
20:37.01 | bweschke | Manxpower: how much do you want to pay? I've got a depot of theirs about 5-10 mins away which is where I go to get them (yes, I agree - their online ordering sucks) :) |
20:37.13 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
20:37.28 | ManxPower | bweschke, I dunno. It has to be here by Feb 28 |
20:37.38 | *** join/#asterisk AlexCTI (n=alex@weston-69.65.87.173.myacc.net) |
20:37.52 | bweschke | what length? cat3 I assume? |
20:38.09 | bweschke | what connectors M/F ? M/M ? F/F ? any specific angles? |
20:38.14 | ManxPower | Really, all I need is an Amphenol gender changer |
20:38.41 | ManxPower | We are replacing a TA850 with a TA750 and they have different genders |
20:39.11 | AlexCTI | Hi Everyone, I need some help to interconnect 2 IAX2, some one can help me? |
20:39.29 | ManxPower | the 850 would have a male cable going into a female port, whereas the 750 needs a female cable going into a male port |
20:40.00 | bweschke | so you need a F/F gender bender? |
20:40.13 | ManxPower | I believe so |
20:40.22 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
20:41.09 | *** join/#asterisk gammacoder (n=chatzill@64-132-192-33.gen.twtelecom.net) |
20:42.19 | hensema | when I enter a digit, asterisk tries to connect me to the extension in the *calling* context, while Background() is run from a macro |
20:42.30 | epablo | Feb 22 15:41:35 ERROR[6538]: chan_zap.c:7017 mkintf: Channel 16 is reserved for D-channel. But i have an T1. What do i have to change ? please |
20:42.37 | hensema | what's up with that? |
20:42.45 | jbalcomb | ManxPower Graybar |
20:42.45 | stoffell | GXP-2000 owners already tried newest build? |
20:42.45 | *** join/#asterisk afrosheen (n=test@txprotoa2.august.net) |
20:42.58 | ManxPower | epablo, the jumpers on the card |
20:43.00 | *** join/#asterisk gambolputty (n=root@64.74.225.135) |
20:43.07 | jbalcomb | stoffell: 1.0.2.13? I've got it on 100+ GXP-2000s |
20:43.21 | afrosheen | ManxPower: Thanks for your help with zap the other day, it's perfect but now I have echo on my sip trunk :p |
20:43.33 | ManxPower | afrosheen, you can't have echo on SIP |
20:43.34 | jbalcomb | epablo: sounds like you have it set up as an E1 rather than a T1 |
20:43.35 | stoffell | jbalcomb, yes, it's released yesterday/today, better then before? |
20:43.37 | epablo | ManxPower: but ztcfg says its ok |
20:43.41 | afrosheen | well, we do, and it's bad |
20:43.43 | ManxPower | epablo, ignore tht |
20:43.49 | *** join/#asterisk ptblank (n=MURDER1@68-169-161-61.lmdaca.adelphia.net) |
20:43.55 | afrosheen | I guess our provider has something messed up on their end |
20:43.57 | epablo | ManxPower: ok |
20:44.43 | bweschke | afrosheen: yes - it's either that or acoustical echo caused by your endpoints - it cannot be traditional echo caused by crossing a hybrid because you do not have any in your architecture |
20:46.23 | afrosheen | is it possible my provider has a hybrid? the calls have to hit the pstn somewhere and it's not here |
20:46.29 | jbalcomb | stoffell: it was released about two weeks ago i thinks actuallu. it is the best so far. having probelms with older hardware versions of the phone though. |
20:46.46 | stoffell | 2 weeks ago? |
20:47.06 | stoffell | u, talking about 1.0.2.13? released 2/21/2006 ... |
20:49.22 | clyrrad | I saw a page a while back with a whole bunch of [apps] defined, call forward, block number etc... I cant find it again, does anyone have the link to that page? |
20:50.48 | djMax | ok, now I know * is whacked out. |
20:51.05 | djMax | I put static IPs in sip.conf, and it STILL has lost those IPs in sip show peers |
20:51.54 | jbalcomb | stoffell: ah, sorry. I'm thinking of 1.0.2.8 |
20:52.10 | *** join/#asterisk ToTo (n=ToTo@host61-133.pool874.interbusiness.it) |
20:52.26 | stoffell | jbalcomb, ok, i know (same issues) that one :( .. but it seems today's a new one on the wiki.. tryin' it tomorrow morning. .should fix those issues |
20:53.26 | justinu | ip301 all upgraded... thx everyone |
20:53.52 | gammacoder | stoffell: I'm running 1.0.2.13 on my gxp2000 - nothing bad to report yet |
20:53.57 | jbalcomb | stoffell: ok, I've got it set up on my phone "Program-- 1.0.2.13 Bootloader-- 1.0.2.3" |
20:54.27 | stoffell | gammacoder, nice to see it fixes the serious issues |
20:54.41 | jbalcomb | stoffell: I'll be releasing it to the IT Dept now. Assuming no additional problem I'll have all 100+ on it next week. |
20:55.09 | stoffell | jbalcomb, same here, but i will first test it for 24hrs on 2 phones at least :) |
20:55.19 | epablo | ManxPower: That was it.. thanks! |
20:57.48 | jbalcomb | stoffell: yeah, our IT Dept is mostly developers so they don't rely on phones so much; makes for a good test bed. |
20:58.19 | *** join/#asterisk Dr-Linux (n=Dreamer@host202-147-168-130.lhr.dancom.net.pk) |
20:58.26 | stoffell | jbalcomb; heh, aaah, okay :-) so will check back here in 24 hrs ;) |
20:58.50 | jbalcomb | stoffell: right on |
20:59.00 | [av]bani | anyone confirm the display corruption is fixed with 1.0.2.13? |
20:59.12 | jbalcomb | [av]bani so i guess GS getting those TFTP settings in for DHCP really helps out your project eh? |
21:00.32 | Dr-Linux | my few clients complaint that when they talked, there voice is too much static. what could be the problem? |
21:00.38 | [av]bani | not really, i figured out workarounds |
21:00.46 | epablo | Well guys i'm leaving.. c'ya |
21:00.49 | *** part/#asterisk epablo (n=epablo@200.75.139.188) |
21:01.02 | afrosheen | Dr-Linux: that'll happen with echo cancellation on and too much tx/rx gain boosting |
21:01.11 | jbalcomb | Dr-Linux: what phone? |
21:02.17 | Dr-Linux | afrosheen: but someone suggest me echo cancellation should be "yes" ? |
21:02.36 | Dr-Linux | jbalcomb: they are using X-Lite softphone |
21:02.39 | afrosheen | Dr-Linux: it should be, correct, but you may have to tame the rx/tx levels if you're using a zaptel interface |
21:02.55 | afrosheen | or in the case of a softphone, tweak the mic sensitivity on the pc |
21:03.06 | *** join/#asterisk WasPhantom (n=neil@203-86-192-98.tasman.net) |
21:03.09 | jbalcomb | Dr-Linux: ok, afrosheen is right on then |
21:03.29 | Dr-Linux | afrosheen: my rx is rx=1.0 and tx is tx=0.0 |
21:03.43 | afrosheen | have your users turn the mic sensitivity down then |
21:04.04 | afrosheen | they're probably your typical shouters that have no concept of mic sensitivity |
21:04.19 | Dr-Linux | hhm.. may be thats mic problem |
21:04.23 | afrosheen | we have a few of those here |
21:04.36 | afrosheen | well it's a problem of a hot mic level plus the echo canceller |
21:04.43 | Dr-Linux | afrosheen: but my hardphone/US users are fine :S |
21:05.09 | afrosheen | welp, it's easy to see the problem now |
21:05.09 | cpm | that's because softphones suck. |
21:05.27 | jbalcomb | stoffell [av]bani is the firmware prefix/postfix part of the directory structure or just the filename? |
21:05.52 | Dr-Linux | hhm... |
21:06.20 | Dr-Linux | afrosheen: actually i can't change the setting bcoz US users are doing fine. |
21:06.32 | Dr-Linux | afrosheen: so what you suggest me? |
21:06.41 | afrosheen | yeah, SO, change the microphone sensitivity on your softphone users |
21:06.47 | [av]bani | jbalcomb: no idea, never tried it |
21:07.21 | stoffell | jbalcomb; prefix? files are called gxp2000.bin etc. |
21:07.36 | Dr-Linux | afrosheen: it can be changed on the user's system? or in the xlite setting? |
21:07.49 | afrosheen | definitely on their systems, maybe in xlite's settings |
21:08.11 | afrosheen | windows lets you set mic levels in the audio control panel, plus you can enable/disable a 20db mic boost |
21:08.47 | Dr-Linux | ok nice |
21:09.04 | afrosheen | I'm glad you mentioned your hard phone users were ok, that really narrowed it down |
21:09.05 | Dr-Linux | afrosheen: what codec should i use for xlite users? |
21:09.16 | afrosheen | bandwidth permitting, ulaw or alaw |
21:09.36 | afrosheen | it does well with sip and has high call quality, handles dtmf inband nicely |
21:09.56 | Dr-Linux | afrosheen: our xlite(pakistan) users have no good bandwidth thts why i'm asking |
21:11.03 | afrosheen | what are they on, dialup? |
21:12.01 | Dr-Linux | afrosheen: no, on DSL, actually we have 4 offices, 3 in pakistan and 1 is in USA, and few of US home users have cisco ip phones |
21:12.09 | Dr-Linux | but all US users are fine .. |
21:12.09 | afrosheen | g729 is also very nice, it costs a little extra but gives impressive call quality for the bandwidth it consumes, plus it trunks very well... |
21:12.23 | Dr-Linux | some pakistan users complaining every day |
21:12.27 | brad_mssw | eh, if you have any amount of packet loss, g729 is aweful |
21:12.40 | brad_mssw | ulaw copes much better with packet loss |
21:12.53 | afrosheen | you could say that about pretty much any codec, since we're talking udp streams anyway |
21:13.09 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
21:13.14 | Dr-Linux | g729 doesn't support xlite |
21:13.22 | afrosheen | the pro version has support for it |
21:13.42 | brad_mssw | afrosheen: true, without PLC any codec isn't great with packetloss, but g729 is one of the worst at coping with it |
21:13.55 | [av]bani | ilbc 4-evar |
21:13.55 | Dr-Linux | afrosheen: but pro is not free |
21:14.06 | kuku5 | Can I have a cisco phone send messages to syslog? |
21:14.17 | afrosheen | well, if it's an international business, I'm assuming you have a few bucks to spend on phones |
21:14.18 | Qwell[] | kuku5: What kind of messages? |
21:14.23 | kuku5 | debug |
21:14.34 | Qwell[] | messages from the phone, or from *? |
21:14.35 | Dr-Linux | ooo what about iLBC, its my pirority codec? |
21:14.51 | afrosheen | ilbc, I'd rather string two soup cans together |
21:14.55 | *** join/#asterisk Winkie (n=urmom@cpc2-stre1-6-0-cust119.bagu.cable.ntl.com) |
21:14.58 | kuku5 | it restarts by itself, or hangs, I need to find out what. When it hangs I get messages like: |
21:15.07 | kuku5 | "Busy here" |
21:15.13 | ManxPower | Anyone here familiar with Nortel? I need information on what to dial from a Nortel (Meridian?) analog line to access call pickup and all-station-page |
21:15.24 | kuku5 | Qwell: Want to get debug from the PHONE |
21:15.31 | Qwell[] | ManxPower: The Feature button + 60.. |
21:15.37 | Dr-Linux | afrosheen: can you tell again your last line in easy english ? :$ |
21:15.40 | ManxPower | Qwell, analog phones don't have a feature button |
21:15.42 | Qwell[] | ManxPower: Let me know if you figure out how to simulate the feature button though :P |
21:15.51 | Qwell[] | I've actually been trying to find that out too |
21:16.07 | ManxPower | Qwell, I found some of the codes at http://www.tek-tips.com/faqs.cfm?fid=5351 |
21:16.20 | Qwell[] | I found all of the codes...but nothing says how to use feature |
21:16.30 | afrosheen | Dr-Linux: oh, you were saying the Xlite Pro isn't free, but I was asking if your company had a budget..you know, spend a few bucks on software |
21:16.32 | Qwell[] | same problem you're having, I assume? |
21:16.48 | ManxPower | Qwell, I have to do a FLASH before dialing the code |
21:17.00 | Dr-Linux | ok |
21:17.01 | Qwell[] | oh? |
21:17.02 | ManxPower | Mostly I'm looking for someone that has done this to confirm |
21:17.09 | ManxPower | BTW, LINK=FLASH |
21:17.12 | Qwell[] | trying to do it through * or something? |
21:17.14 | Dr-Linux | afrosheen: what if we i use SJphone? |
21:17.18 | *** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk) |
21:17.38 | afrosheen | dunno, I've never tried SJphone |
21:17.40 | ManxPower | Qwell, Trying to allow our Asterisk users to do all station page to the nortel extensions and be able to pickup parked calls on the nortel |
21:17.42 | Qwell[] | ManxPower: I'm just a user, so "LINK=FLASH" is foreign to me... |
21:18.07 | afrosheen | regardless, g729 costs a little extra due to licensing..but it has many benefits |
21:18.34 | Qwell[] | ManxPower: Are you saying that if you have a phone, you just do a flash, then dial the feature, or? |
21:18.44 | Dr-Linux | afrosheen: i have g729 codec, but don't have softclient for it :P |
21:18.49 | Qwell[] | because if I could figure it out, it would save me a few headaches... |
21:18.52 | ManxPower | from the URL I pasted: All access codes must be preceeded with a Flash Hook (Link) |
21:18.56 | Qwell[] | ahh |
21:19.09 | Dr-Linux | afrosheen: my both of servers are running fine from last 3 months |
21:19.25 | ManxPower | Qwell, if you have an ATA2 or EATA (converts Norstar phone ports into analog ports) yes, you FLASH+code |
21:19.37 | Qwell[] | ManxPower: I've just got a nortel phone sitting on my desk |
21:19.44 | Dr-Linux | today i got a complain, that one of my user mailbox is full .. |
21:19.47 | Qwell[] | not trying to use any thirdparty stuff |
21:19.47 | ManxPower | Qwell, that's prolly a nortel DIGITAL phone |
21:19.54 | Qwell[] | still no feature button on it ;/ |
21:20.09 | ManxPower | Qwell, can you plug an analog phone into the line? |
21:20.12 | Dr-Linux | so i was trying trying all day googled to findout how to increase the quota .. |
21:20.23 | Qwell[] | umm |
21:20.31 | Qwell[] | doubt it...haven't tried |
21:20.43 | Dr-Linux | afrosheen: then finally i findout :) |
21:20.44 | Qwell[] | it's got a 6 wire going to it though |
21:20.52 | *** join/#asterisk delmar (n=Delmar@203-114-178-231.inspire.net.nz) |
21:21.16 | afrosheen | increase what quota? |
21:21.37 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
21:21.52 | Dr-Linux | afrosheen: user's voice mailbox |
21:22.11 | Qwell[] | flash isn't doing a whole lot, heh |
21:22.28 | Qwell[] | I freaking hate these phones. No freaking mute button on them |
21:22.32 | Dr-Linux | today i got to know, bydefault user can have 100 voicemails in his/her account |
21:22.39 | asteriskmonkey | the citel gateways rule :D |
21:22.47 | Qwell[] | there is a "handsfree mute", but that puts you in speaker, and mutes that mic |
21:22.51 | ManxPower | Dr-Linux, actually 99 |
21:22.53 | asteriskmonkey | Dr-Linux just change the limit to 1000 :D |
21:23.08 | Dr-Linux | if it's increase caller will be listed, "sorry, the users can't accept more messages" |
21:23.25 | ManxPower | Acutally since they start at 0, yes it is 99 |
21:23.31 | Dr-Linux | yep |
21:23.36 | ManxPower | Anyone that has 99 messages in their mailbox is a moron and should be shot. |
21:23.39 | afrosheen | yeah. btw the xlite pro version is called eyebeam, each copy is $30 each for windows..so that's cheaper than a hardphone or an ata adapter |
21:23.42 | Dr-Linux | but actually i didn't know how to change it :) |
21:24.05 | afrosheen | ManxPower: you should bring a shotgun over here |
21:24.05 | ManxPower | Dr-Linux, voicemail.conf.sample is your FRIEND. |
21:24.08 | asteriskmonkey | i would go that far some people like savign them for legal reasons |
21:24.12 | Qwell[] | ManxPower: Got anything else in that bag of tricks of yours, for me? |
21:24.24 | ManxPower | Qwell, get rid of the piece of crap phone |
21:24.37 | ManxPower | You obviouly have a digital set and I can't help with that |
21:24.40 | Qwell[] | ManxPower: indeed |
21:24.49 | Qwell[] | we're gonna try to get a PRI off the AT&T in the other building |
21:24.53 | afrosheen | asteriskmonkey: that's why our server emails the vm to each user..they can save them locally rather than on the server |
21:25.17 | Qwell[] | there are only two depts still on this nortel. Mine, and securitybuilding management |
21:25.30 | Dr-Linux | yep, i saw a help on WIKI and i added an option in voicemail.conf maxmsg=500 :) |
21:25.35 | Qwell[] | s/yb/y\/b/ |
21:25.39 | Qwell[] | silly boy |
21:25.40 | Qwell[] | bot |
21:26.03 | afrosheen | s /silly boy/silly bot |
21:26.10 | afrosheen | he's not paying attention today is he |
21:26.21 | Qwell[] | he is...if it's "jbot valid" regex |
21:26.42 | *** join/#asterisk Dandan (i=dandan@ellie.pacanka.com) |
21:27.47 | Dr-Linux | asterisk doesn't have voice recognition app yet? |
21:28.02 | ManxPower | Dr-Linux, if you read the .sample files you'll fine all sorts of cool options |
21:28.11 | Qwell[] | Dr-Linux: voice recognition isn't something somebody "just writes" |
21:28.34 | afrosheen | it'll cost an arm and a leg when it's available, companies pay huge money for that feature |
21:29.38 | Dr-Linux | ManxPower: is there some new voicemail option after 1.2.1 version? |
21:29.56 | asteriskmonkey | you can always try sphynx for vr |
21:30.08 | *** join/#asterisk redondos (n=redondos@190.48.41.94) |
21:30.31 | Dr-Linux | asteriskmonkey: yes i installed that, but that doesn't work for me |
21:30.43 | asteriskmonkey | it required making of custom dictionaries |
21:30.49 | asteriskmonkey | its uver time consuming setting it up |
21:31.40 | Dr-Linux | actually i don't understand what custom is in * |
21:32.20 | Dr-Linux | i saw many sample extensions.conf files, there is wide of use this work "custom" but i don't understand what's this |
21:33.04 | asteriskmonkey | in linuix or unix type at the cli "dict custom" |
21:34.13 | *** join/#asterisk gammacoder (n=chatzill@64-132-192-33.gen.twtelecom.net) |
21:34.26 | Dr-Linux | i'll in office |
21:34.42 | jbalcomb | I *heart* Grandstream. "Allow DHCP Option 66 to override server" is perhaps an unfriendly description of the option? |
21:35.00 | afrosheen | custom means 'different than what you get to begin with', so if you buy a car and change the wheels, bam, you have 'custom wheels' |
21:35.00 | Qwell[] | jbalcomb: option 66 is tftp I believe |
21:35.30 | Qwell[] | which is very, very useful on cisco phones |
21:35.37 | ManxPower | Dr-Linux, no, MAJOR releases |
21:35.41 | Qwell[] | plug a brand new phone in...bam, it works |
21:35.58 | jbalcomb | Qwell[] indeed it is. i see it on the wiki I just can't imagine a developer not thinking to just put "Allow TFTP Server override via DHCP" |
21:36.16 | Dr-Linux | afrosheen: yes that i know, but how it works in * |
21:36.20 | Qwell[] | jbalcomb: any admin should know what 66 is :) |
21:36.24 | jbalcomb | Qwell[] Yeah, seems like between GS and [av]bani the GXP-2000 will be there soon |
21:36.26 | gammacoder | Grandstream's DHCP Option 66 isn't enabled by default - you have to enable it for each phone - kinda defeats the purpose of auto-provisioning |
21:36.49 | ManxPower | gammacoder, Much like the VLAN setting for polycom phones. |
21:36.53 | jbalcomb | Qwell[]: haha.. agreed. ;) too bad everyone isn't an admin. |
21:36.53 | afrosheen | Dr-Linux: well, that's how it works, generally custom files are files that add functions to the normal defaults, and are added via include statements |
21:36.56 | Dr-Linux | anybody is using SPA-2100 ? |
21:37.03 | ManxPower | Dr-Linux, they are great |
21:37.21 | Dr-Linux | afrosheen: great |
21:37.41 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
21:37.46 | afrosheen | so for example, Iloveshrimp.conf will have a line like #include extrashrimp_custom.conf |
21:38.05 | Dr-Linux | ManxPower: i have a problem with spa 2100, i forgot web interface admin password. |
21:38.08 | afrosheen | When Asterisk reads iloveshrimp, it sees it needs to read in the custom file as well |
21:38.14 | ManxPower | Dr-Linux, They make a good bookend |
21:38.28 | ManxPower | you tried the factory reset via the analog port? |
21:38.40 | Dr-Linux | i need to do some modification from web, but i forgot the password |
21:39.12 | Dr-Linux | ManxPower: yes that i know, but i don't wanna reset the fectory |
21:39.17 | Dr-Linux | let me explain my problem |
21:39.36 | jpablo | hey people, I'm looking at cdr records and i see that billsecs is starting to count in zap channels even before the channel is answered, any idea how can start the billsec only if the called party in the zap channel (e1) answers ? |
21:40.03 | Dr-Linux | ManxPower: i got this spa 2100 from a VOIP provider, so it has 2 lines provisioned from them for always ... |
21:41.06 | Dr-Linux | so whenever i need to configure it with my asterisk extension, i simpley turn of provisioned option, and add my asterisk extensions, then it work |
21:41.13 | Dr-Linux | but i forgot the password |
21:41.47 | Dr-Linux | ManxPower: so if reset the fectory i will lost the voip provider things too |
21:41.50 | Dr-Linux | what you say? :S |
21:42.07 | afrosheen | yeah reset it then call your ISP saying lightning struck a tree in your yard and reset your phone |
21:42.29 | afrosheen | s /ISP/Voip provider |
21:43.10 | Dr-Linux | afrosheen: but the these 2 lines are free, if i'll inform them, they will not give me back :P |
21:44.01 | *** join/#asterisk iq (n=iq@71-214-6-43.omah.qwest.net) |
21:45.00 | ManxPower | Dr-Linux, What I say is that I've never lost the password to the SPA-2100 |
21:45.04 | Dr-Linux | so all i need is to login to web interface and disable there provisioned option temprary :) |
21:45.09 | afrosheen | jeez you gotta be the cheapest guy I ever met |
21:45.14 | afrosheen | ;) |
21:45.30 | ManxPower | Dr-Linux, Did the box come preconfigured from the provider? If so, they can lock you out. |
21:46.17 | Dr-Linux | ManxPower: yeah it is, but they didn't lock it out since 3 months |
21:46.37 | Qwell[] | doesn't mean they didn't lock you out |
21:47.18 | [TK]D-Fender | Dr-Linux : Maybe you didn't get the idea earlier : THERE SHOULDN'T BE ANY WAY AROUND THE ADMIN PASSWORD. Thats the POINT of it. You want in? Ask them for the password or flush your settings. DEAL WITH IT. |
21:47.30 | *** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com) |
21:48.00 | Egonis | Dumb... DUMB question: Can I use my Vonage Account w/ Asterisk? They are useless for telling me which protocol it uses, in fact, they said it uses the UDP Protocol |
21:48.15 | Qwell[] | Egonis: idiots. no, you can't |
21:48.25 | Dr-Linux | [TK]D-Fender: i agree sir |
21:48.35 | Qwell[] | It uses SIP, but it's against their TOS to use anything but what they give you |
21:48.45 | afrosheen | vonage != * |
21:48.52 | Dr-Linux | [TK]D-Fender: i'm the one who ever set the admin password :) |
21:49.18 | afrosheen | wait, how do you express not compatible with... !+= ? |
21:49.27 | Dr-Linux | actually we bought this device, if it works with data application. but i doesn't work |
21:49.31 | Qwell[] | Egonis: Get another provider.. |
21:49.46 | ManxPower | Egonis, if you have a Vonage SOFTPHONE account you can. Vonage uses SIP. |
21:50.12 | Qwell[] | it's still against their TOS though, isn't it? |
21:50.17 | ManxPower | NotFreak, there is no unlimited calling on the softphone account. |
21:50.27 | *** join/#asterisk nurfe (n=rgff@h24-207-10-162.dlt.dccnet.com) |
21:50.35 | Egonis | ManxPower: ooh, handy! ty |
21:50.41 | ManxPower | Qwell, maybe, but since the account is not unlimited...... |
21:50.54 | ManxPower | oh, and you have to have a normal account in order to get the softphone account |
21:51.13 | ManxPower | Can you guess where I learned this? |
21:51.17 | ManxPower | FROM THE MAILING LISTS! |
21:51.25 | ManxPower | The ASTERISK mailing lists. |
21:51.36 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
21:51.41 | Egonis | exit |
21:51.58 | afrosheen | looks like Egonis spends a little time with a shell open |
21:52.31 | iq | Hi, I've a TE110P. Do I need rob-bit T1 or ISDN T1? I will prefer rob-bit T1. |
21:52.48 | fu3 | ManxPower.. another fine example of why you're the best. |
21:53.15 | ManxPower | fu3, I'm available for consulting if you throw reasonably large piles of cash at me. |
21:53.47 | ManxPower | iq, You need whatever your line needs. |
21:53.59 | ManxPower | But ALWAYS ISDN PRI is best |
21:54.07 | ManxPower | aka EuroISDN or ISDN110 |
21:54.18 | fu3 | I dont know of many problems that cannot be solved with large amounts of cash. |
21:54.25 | ManxPower | fu3, exactly. |
21:54.26 | fu3 | but thanks :) |
21:54.39 | [TK]D-Fender | fu3 : Or really REALLY, REALLY ... big lasers :D |
21:54.44 | fu3 | hahahaha |
21:54.59 | iq | ManxPower: so the board will auto detect the signaling? |
21:55.16 | ManxPower | iq, no. you have to tell the board what signaling your line requires. |
21:55.22 | ManxPower | But if you can, ALWAYS order PRI |
21:55.26 | *** join/#asterisk cogineo (n=soner@81.215.105.114) |
21:55.49 | ManxPower | If your line requires CAS/robbed bit it's not going to work if you configure it as PRI in Asterisk |
21:56.58 | *** join/#asterisk _0_0_ (n=000000@81-178-251-94.dsl.pipex.com) |
21:58.37 | iq | ManxPower: Do I define it under /etc/zaptel.conf or /etc/asterisk/zapdata.conf ? |
22:00.26 | stoffell | now, i know a bit about Mark Spencer creating *, but anyone knows his age more or less? :) |
22:01.12 | Qwell[] | stoffell: Ask him |
22:01.39 | stoffell | ok Qwell, i'll try it on fosdem next weekend, but i think the tutorial is meant for other questions :p |
22:01.45 | *** join/#asterisk jyukes (n=jameshot@c-69-248-195-94.hsd1.nj.comcast.net) |
22:01.52 | Qwell[] | stoffell: Ask him here |
22:02.30 | Qwell[] | stoffell: really though, I'm pretty sure he's got a bio or two online :) |
22:02.46 | stoffell | hm, ok, will google for 'm Qwell ;) |
22:03.03 | Qwell[] | *cough*stalker*cough* |
22:03.05 | Qwell[] | :P |
22:03.28 | stoffell | lol, you wouldn't believe why i'm asking this.. |
22:03.37 | Qwell[] | probably not |
22:03.40 | Qwell[] | so...do tell |
22:03.51 | *** join/#asterisk jyukes_ (n=jameshot@c-69-248-195-94.hsd1.nj.comcast.net) |
22:03.55 | stoffell | my gf is asking, because i'm dragging her to his speech on fosdem ;) |
22:04.05 | fu3 | hahaha |
22:04.09 | fu3 | what a stalker ;) |
22:04.15 | fu3 | thats what the last six stalkers said |
22:04.28 | stoffell | oh boy, gotta change my nick now.. :p |
22:04.31 | fu3 | haha |
22:04.55 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
22:05.11 | *** join/#asterisk paulhuynh (n=paul@c-68-37-18-82.hsd1.de.comcast.net) |
22:05.13 | fu3 | ok.. even using an oscilloscope, I cannot detect a 500ms battery drop upon hangup. |
22:05.19 | stoffell | oh boy, apr 1977.. tnx Qwell for the hint ;) |
22:05.19 | fu3 | so I doubt thats happening |
22:06.55 | paulhuynh | hello |
22:06.57 | paulhuynh | everyone |
22:08.01 | *** join/#asterisk Morex (n=blah@host86-140-246-153.range86-140.btcentralplus.com) |
22:08.11 | Morex | Hello all |
22:08.31 | Morex | Anyone have any thoughts on how an agent might indicate to the system that he/she is wrapping up a call? |
22:08.40 | Qwell[] | Morex: by hanging up |
22:08.52 | Morex | Won't they get the next call straight away? |
22:08.59 | stoffell | Qwell, i was thinking the same but didn't dare to type it .. :p |
22:09.16 | Qwell[] | Morex: yes. there is a wrapuptime option though for that |
22:09.26 | Morex | What if not every call requires a wrap up? |
22:09.55 | Qwell[] | have a pause button on your phone? |
22:10.03 | Morex | Nope |
22:10.06 | paulhuynh | i'm using asterisk@home and i did an upgrade |
22:10.23 | Qwell[] | Morex: I mean...add one |
22:10.27 | paulhuynh | how can i merge the old cdr from the old server to the new old |
22:10.29 | Morex | LOL |
22:10.33 | Qwell[] | have something that calls PauseQueueMember |
22:10.42 | Morex | From inside AgentLogin? |
22:10.50 | Qwell[] | dunno, I don't use agents |
22:11.08 | Morex | Ah OK |
22:12.17 | Morex | So no obvious solution then... |
22:12.39 | [TK]D-Fender | paulhuynh : You will need to have exported them from MySQL, and can then just dump the record back in I guess so long as the rest of the server's config matches (for sanity's sake) |
22:13.28 | [TK]D-Fender | PauseQueueMember works on staticly defined agents.... |
22:13.36 | Qwell[] | well then |
22:13.44 | Morex | You can get it to work with dynamic agents too |
22:13.54 | Morex | But how do I activate it from inside a call? |
22:14.15 | [TK]D-Fender | Morex : You could do a featuremap to it with features.conf I suppose |
22:14.37 | [TK]D-Fender | or use another line appearance to do it and put the current call on hold. |
22:15.01 | *** join/#asterisk Tamarisk (n=adrian@user-2899.lns1-c8.dsl.pol.co.uk) |
22:15.11 | _0_0_ | I'm having some issues converting my dialplan to AEL: |
22:15.11 | _0_0_ | <PROTECTED> |
22:15.13 | Qwell[] | or set DND |
22:16.14 | [TK]D-Fender | DND is not a good idea though. The queue will still try to call you and in certain strategies will cause a dead-end effect for future callers.... |
22:16.32 | [TK]D-Fender | At the minimum it will waste an agent-cycle on you. |
22:16.32 | *** join/#asterisk gomez_ (n=gomez@se400.pppoe03-571.bih.net.ba) |
22:16.42 | gomez_ | hello |
22:16.51 | Morex | I was thinking of putting every call into a long wrapup, then have them exit from it some how |
22:16.55 | gomez_ | is there anyone with gpx-2000? |
22:17.03 | stoffell | gomez_, what you want to know? |
22:17.13 | gomez_ | well, i have problem |
22:17.18 | gomez_ | with my phone |
22:17.21 | stoffell | shoot |
22:17.33 | gomez_ | i have registered extension |
22:17.39 | gomez_ | everything |
22:17.47 | *** join/#asterisk DeadZen (n=DeadZen@adsl-2-119-219.mia.bellsouth.net) |
22:17.50 | [TK]D-Fender | Morex : give them a big enough wrapup to handle "average" and let them make the choice to go on "pause" |
22:17.52 | gomez_ | is set up |
22:18.00 | gomez_ | but phone is still |
22:18.15 | gomez_ | i meen user on line 1 is still Not registered |
22:18.36 | DeadZen | is there anyway to make asterisk call you? with like just a audio message |
22:18.46 | stoffell | gomez_, try asterisk -r -> sip show peers |
22:19.41 | Tamarisk | Are there any basic training courses for * available in the UK, does anyone know? |
22:19.54 | _0_0_ | DeadZen: stick a .call file in /var/spool/asterisk/outgoing/ IIRC. |
22:20.12 | DeadZen | really neat |
22:20.37 | mikefoo | Tamarisk: training book is available via the www |
22:20.43 | Dr-Linux | anybody is using Wakeup call? |
22:20.44 | mikefoo | 'world' wide web :) |
22:20.58 | *** join/#asterisk zeedo (n=zeedo@80-192-53-14.stb.ubr04.glen.blueyonder.co.uk) |
22:21.00 | Tamarisk | Is that the Oreilly book on line? |
22:21.19 | Dr-Linux | ~thebook |
22:21.21 | jbot | methinks thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
22:21.21 | mikefoo | thats the recommended one |
22:21.46 | DeadZen | _0_0_: is there docs on .call files? |
22:22.12 | Qwell[] | DeadZen: Yes |
22:22.18 | DeadZen | asteriskdocs.org ? |
22:22.23 | Tamarisk | I have downloaded that one, unfortunatly still find it hard going trying to work through it |
22:23.07 | paulhuynh | what is the channel for amp |
22:23.18 | Tamarisk | I must also ask, when someone sends a ,essage to me, the line is in red, is that done just by enetring my user name at the beginnig |
22:23.25 | Tamarisk | message |
22:23.32 | DeadZen | Qwell: where? |
22:23.33 | zeedo | Tamarisk: yeh it highlights on your username |
22:23.38 | Qwell[] | ~docs |
22:23.39 | jbot | hmm... docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
22:23.47 | Qwell[] | DeadZen: Pick one... |
22:24.01 | DeadZen | hopefully its in the pdf |
22:24.10 | Tamarisk | zeedo, so this should be highlighted to you, yes? |
22:24.16 | zeedo | Tamarisk: yep |
22:24.23 | Tamarisk | I can learn |
22:24.28 | zeedo | Tamarisk: it's configurable in your client though, so you can switch it off |
22:24.37 | paulhuynh | anyone here use sixtel or cvc? |
22:24.41 | _0_0_ | try here: www.voip-info.org/wiki-Asterisk+auto-dial+out |
22:24.47 | mikefoo | pauldy: I was looking at sixtel |
22:24.58 | mikefoo | going to signup this weekend probably. |
22:25.24 | Tamarisk | To be honest I hate IRC, but it does span continents with great speed, but sometimes my typing abilities and manors can and do affend |
22:25.36 | Tamarisk | offend! |
22:25.53 | DeadZen | manners! |
22:26.25 | mikefoo | I need a realiable inbound voip provider, in the US, anyone have a recommendation? |
22:26.30 | mikefoo | was looking at asterlink |
22:26.35 | Tamarisk | there you go English was never a good subject for me, I need to enable spellcheck |
22:26.37 | Qwell[] | mikefoo: asterlink is good |
22:26.39 | paulhuynh | sixtel is good service |
22:26.46 | Qwell[] | right file? |
22:26.48 | paulhuynh | but not very good when it come to support |
22:26.48 | bweschke | mikefoo: asterlink is good |
22:27.09 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
22:27.28 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
22:28.19 | mikefoo | bweschke: any idea if I can have simultaneous inbound calls? |
22:28.31 | mikefoo | I would need unlimited to some extent. |
22:29.14 | mikefoo | paulhuynh: Ahh i will be using them for outbound, and atleast I can have some redundency with outbound, inbound is a whole other story. |
22:29.15 | *** join/#asterisk Mce (n=Mce@2416444hfc89.tampabay.res.rr.com) |
22:29.15 | Dr-Linux | after 35 days i did "reload" today |
22:30.04 | Dr-Linux | i afraid to do "reload" if may be it nver come back! |
22:30.17 | stoffell | Dr-Linux , hehe :) |
22:30.24 | mikefoo | lol |
22:30.36 | Dr-Linux | lol |
22:30.46 | Dr-Linux | reload is not neccesary, right? |
22:31.00 | stoffell | Dr-Linux; only if you change something |
22:31.19 | mikefoo | reload just re-reads config files |
22:31.24 | Dr-Linux | stoffell: well, if i change then also i do not reload :P |
22:31.48 | stoffell | Dr-Linux; well, that's a good way to insure stability, ..... if it never goes down.. :) |
22:31.54 | Dr-Linux | i only reload specific file |
22:32.34 | stoffell | Dr-Linux you must have a solid dialpattern if you didn't reload for 35 days :) |
22:32.41 | Mce | Hi, I have an odd question: Is it possible to use asterisk not as a PBX, but only as the business end for a commercial wardialer app that claims it can talk to asterisk, or would you need to completely replace your phone system with asterisk to do that? |
22:32.57 | Qwell[] | wardialer app? |
22:33.03 | Mce | iwar specifically |
22:33.11 | mog_work | yeah you can use asterisk to bulk call people |
22:33.12 | Qwell[] | for telemarketting purposes? |
22:33.13 | Dr-Linux | stoffell: why? what do you mean ? |
22:33.14 | paulhuynh | MCE |
22:33.17 | Qwell[] | if so, get out |
22:33.20 | Mce | no |
22:33.20 | paulhuynh | PM me we will talk |
22:33.33 | Mce | not for that |
22:33.36 | stoffell | Dr-Linux; i mean, you have a "great" dialplan if you don't often change it :) |
22:33.37 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
22:34.11 | Dr-Linux | stoffell: well, i reload but i do not use "reload" command |
22:34.46 | Dr-Linux | stoffell: if just use "extension reload" or "sip reload" anyone if needed |
22:34.55 | Dr-Linux | i do not bother entire reload |
22:35.18 | stoffell | Dr-Linux; okay, indeed a good practice, .. entire reload is useless in that case |
22:35.21 | Dr-Linux | bcoz when reloading...... during that time .. my heart also reloads :P |
22:35.39 | stoffell | :) |
22:35.43 | Morex | Solution to queue issue turns out to be set ackcall=always |
22:35.57 | Morex | This allows agents to press # before accepting calls even in AgentLogin |
22:36.06 | Morex | It's in the source code but undocumented elsewhere. |
22:36.13 | Dr-Linux | stoffell: somethime i think, i should remove all other extra config files :S |
22:36.31 | Qwell[] | Dr-Linux: Just noload the other modules |
22:37.22 | Dr-Linux | Qwell[]: great |
22:38.04 | [TK]D-Fender | stoffell : No, its a question of not asking for more than you've got :) |
22:38.06 | Dr-Linux | in which case asterisk process gets auto killed, and it needs start again? :S |
22:40.45 | *** join/#asterisk LeonardoCabelo (n=leonardo@unaffiliated/leonardocabelo) |
22:40.53 | DeadZen | asterisk is the coolest thing since sliced babies |
22:41.12 | FuriousGeorge | sure, if your into that sort of thing |
22:41.18 | DeadZen | lol im j/k |
22:41.21 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-92-216.cybersurf.com) |
22:41.23 | DeadZen | i think i got it to out dial me |
22:41.26 | DeadZen | im trying to leave a msg now |
22:42.02 | FuriousGeorge | make sure you remember to leave yourself your number |
22:42.05 | DeadZen | it says my Status is OK (127 ms) |
22:42.15 | DeadZen | but i can't make my sjphone ring yet |
22:42.15 | DeadZen | heh |
22:42.22 | Dr-Linux | i wish to have bandwidth over here .. aww |
22:42.37 | LeonardoCabelo | Somebody know how to configure the G668 Soyo HardPhone in the asterisk |
22:42.48 | DeadZen | hehe |
22:42.50 | DeadZen | hard phone |
22:42.55 | Dr-Linux | DeadZen: you are using SJphone? |
22:42.59 | DeadZen | yes |
22:43.11 | DeadZen | tried using a sip component for delphi to make my own |
22:43.11 | FuriousGeorge | anyone know why it is that when there is an unavailable CID from my pots provider asterisk assigns "Asterisk" to the CID? |
22:43.18 | FuriousGeorge | anyway to replace that with an unknown |
22:43.18 | DeadZen | but it says my status is unavailable so i can only dial out |
22:43.20 | DeadZen | boo |
22:43.26 | Dr-Linux | DeadZen: what codecs SJphone supports ? |
22:43.36 | FuriousGeorge | another thing that annoys me. it appends the domain to the cid. my users could care less about my * server's domain |
22:43.41 | *** part/#asterisk Mce (n=Mce@2416444hfc89.tampabay.res.rr.com) |
22:43.51 | DeadZen | well gsm ulaw for sure |
22:44.01 | DeadZen | resti dunno.. i suppose alaw |
22:44.05 | ManxPower | I always force the CLID in sip.conf. Never trust the phone |
22:44.19 | ManxPower | or the user for that matter |
22:44.20 | afrosheen | Dr-Linux: yeah now that I think about it, you should use gsm |
22:44.22 | Dr-Linux | DeadZen: does it support g729 ? |
22:44.32 | DeadZen | i dunno |
22:44.34 | FuriousGeorge | ManxPower: CLID? did you mean CID or were you not talking to me? |
22:44.35 | ManxPower | free softphones do not support G729 |
22:44.40 | DeadZen | i think they mostly do 726 |
22:44.47 | ManxPower | FuriousGeorge, Calling Line ID aka Caller ID |
22:44.55 | afrosheen | the quality kinda stinks but it's free, and anyone used to talking on a cellphone won't notice the quality |
22:44.55 | DeadZen | why do you need 729 |
22:45.07 | DeadZen | quality superior? |
22:45.12 | FuriousGeorge | ManxPower: i didnt know you could set that in sip.conf, i guess i stick it in the general section? |
22:45.26 | Dr-Linux | afrosheen: yep gsm eats 13 kb, but with network header etc i reachs to 30 kb |
22:45.36 | FuriousGeorge | anyone ever hear of externhost not to be confused with sternip? |
22:45.37 | afrosheen | I advised him earlier to try it, it does well with limited bandwidth, trunks well, etc. |
22:45.42 | Dr-Linux | DeadZen: we don't have fuckin bandwidth here |
22:45.44 | ManxPower | callerid=Murphy, Tonda <9857683227> |
22:45.47 | afrosheen | lol |
22:45.55 | FuriousGeorge | ManxPower: im talking about for incoming |
22:45.55 | ManxPower | in each sip [whateverpeerfrienduser] section |
22:45.55 | DeadZen | i dunno sjphone has shitty quality |
22:45.56 | afrosheen | funny when foreigners curse :) |
22:46.05 | Tamarisk | An idea! but possibly it would be un-workable, Whould it be possible to run an asterisk voice conference system for assitance, like Skype of freeworld, would bandwidth cripple it? |
22:46.06 | DeadZen | almost unusable |
22:46.09 | ManxPower | FuriousGeorge, incoming from an ITSP or something, not incoming from a phone? |
22:46.09 | [TK]D-Fender | FuriousGeorge : Yes, I know people who use EXTERNHOST, and others that use EXTERNIP. What of it? |
22:46.14 | FuriousGeorge | whenever its unknown i get Asterisk@10.0.0.10 |
22:46.26 | Dr-Linux | afrosheen:: but spa hardphone doesn't work with GSM |
22:46.37 | FuriousGeorge | [TK]D-Fender: wondering what the difference is. im trying to get * not to cache my *.dynu.com dynamic ip'ed boxen |
22:46.37 | mikefoo | Anyone care to buy a 7960? |
22:46.37 | afrosheen | oh man that blows |
22:46.38 | DeadZen | and thats the day Curious George became |
22:46.40 | DeadZen | Furious George! |
22:46.54 | ManxPower | The only low bandwidth codec most phones support is G.729 |
22:46.58 | trixter | what that the day when curiours george was no longer curious? |
22:46.59 | FuriousGeorge | ManxPower: yeah, unknown cid from PSTN come in as Asterisk@10.0.0.10 |
22:47.01 | FuriousGeorge | or something |
22:47.12 | ManxPower | FuriousGeorge, Ah. No idea. I don't use SIP for PSTN access. |
22:47.13 | [TK]D-Fender | FuriousGeorge : Use higher rate of EXTERNREFRESH to keep it udated |
22:47.16 | DeadZen | oh thats dick cheney |
22:47.18 | afrosheen | FuriousGeorge: you can set a system default CID can't you |
22:47.22 | FuriousGeorge | one rig even just randonly displays one of my zap CIDs on unknwn |
22:47.22 | Qwell[] | mikefoo: I'll give you like $100 |
22:47.33 | Qwell[] | assuming it works |
22:47.48 | ManxPower | it would be pretty trivial to do a match on that callerid and reset it to something you want. |
22:47.54 | FuriousGeorge | afrosheen: i would think so, i know i can put it in the dialplan logic, but i'd think there was an option |
22:48.00 | Dr-Linux | but iLBC CPU hungry .. but i have a strong server with daul processor |
22:48.15 | ManxPower | FuriousGeorge, I'll bet you could change it in chan_sip.c |
22:48.20 | FuriousGeorge | [TK]D-Fender: externrefresh where? in sip.conf? cuz my problem is with my iax register=> to my dynu.com boxes |
22:48.25 | Qwell[] | guess not |
22:48.36 | DeadZen | anyone ever heard of astatech ? |
22:48.41 | [TK]D-Fender | FuriousGeorge : Not sure about IAX.CONF. |
22:48.49 | DeadZen | they make a .net and delphi sip component |
22:48.59 | Qwell[] | delphi? wtf? |
22:49.20 | DeadZen | yah delphi.. you know that thing before .net |
22:49.21 | DeadZen | heh |
22:49.24 | FuriousGeorge | [TK]D-Fender: yeah, there seem to be all these work arounds for sip, but my problem is with my dynamic ip's, iax, and * caching the ip |
22:49.28 | Qwell[] | that thing before like...1990? |
22:49.39 | mikefoo | Qwell: ofcourse it works, its sccp, as of now, are you able to flash it to anyother protocol if needed? |
22:49.39 | DeadZen | nah they kept it up till bout 2005 |
22:49.49 | DeadZen | they're selling it though |
22:49.50 | Qwell[] | mikefoo: of course. I'd use sccp though |
22:49.56 | DeadZen | and borlands doing life cycle management |
22:50.03 | [TK]D-Fender | FuriousGeorge : Maybe if you set it up in parallel in SIP.CONF it might take it... |
22:50.40 | mikefoo | Qwell: ok I will get back to you in a day or so. |
22:50.43 | rayvd | I'm a sturgeon. |
22:50.54 | Qwell[] | mikefoo: alright, I'll be around |
22:50.55 | FuriousGeorge | [TK]D-Fender: meaning that i can try to register via sip in every direction as i am now, and that may help my iax registered stay good on ip change |
22:50.56 | FuriousGeorge | ? |
22:50.58 | DeadZen | hahah |
22:51.02 | DeadZen | that was soo cool! |
22:51.08 | DeadZen | I just called myself and said You sound cute! |
22:51.18 | DeadZen | haha i just made my day |
22:51.34 | DeadZen | now to make a goofy dial plan and play tricks on my buddies |
22:51.38 | [TK]D-Fender | FuriousGeorge : No, just try setting the externhost & externrefres in SIP.CONF and that may keep it from caching in iax.conf. |
22:51.59 | FuriousGeorge | [TK]D-Fender: ill let you know if that worked next time my ip's change |
22:52.10 | [TK]D-Fender | DeadZen : I havea special exten that changes my outbound callerid to 867-5309 :) |
22:52.12 | ManxPower | DeadZen, must be a telecom newbie |
22:52.18 | FuriousGeorge | is there any way to not allow my sip clients to display the domain on the CID? |
22:52.35 | DeadZen | thats cool fender |
22:52.35 | ManxPower | FuriousGeorge, preprocess the calls |
22:52.41 | DeadZen | yah I'm new to telecom |
22:52.49 | DeadZen | but I'm a good app programmer |
22:52.51 | ManxPower | I've found that you really have to preprocess calls |
22:52.52 | DeadZen | gonna try to mix em |
22:53.03 | FuriousGeorge | ManxPower: you mean answer() and check the cid by hand, as it were? |
22:53.09 | ManxPower | why answer? |
22:53.42 | ManxPower | FuriousGeorge, to do things like prepend a 9 so people can call back using the phone features, add the local area code if it's not there, etc. |
22:53.44 | FuriousGeorge | well, not necessarily requiring an answer, but basically set the cid myself |
22:54.02 | FuriousGeorge | np |
22:54.05 | FuriousGeorge | ill work on that too |
22:54.47 | ManxPower | I recommend sending all the incoming calls to a context that just has a generic pattern match, do whatever preprocessing you need (callerid, group, etc), then Goto(therealcontext,${EXTEN},1) |
22:55.01 | FuriousGeorge | DeadZen: set your cid to all your friends mom's / girlfriends and call them all day till they complain to your provider |
22:55.16 | ManxPower | FuriousGeorge, for example in the USA all callerid should be 10 digits long. If it's not then fix it up |
22:55.16 | FuriousGeorge | ManxPower: makes sense |
22:55.24 | [TK]D-Fender | ManxPower : I'm nearing the point where I'd rather use timeout rules to pass the # so taht my users can use 7-10-11 digit dialing and 4 digit extens, but would have to wait for it to be issued. they could pre-dial then lift the receiver or hit "send" to speed it up, but I think it'd be for the best. |
22:55.47 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
22:56.10 | Qwell[] | just 9+ them |
22:56.26 | Qwell[] | 9+7, 9+11 |
22:56.31 | ManxPower | [TK]D-Fender, I disagree. My dialplan is basically, 4-digit extensions, local calls are 9+number, toll calls are 9+1+AC+number, international is 9+011+country code+city code+number |
22:56.54 | ManxPower | with a few exceptions like 911 should not REQUIRE 9-911 |
22:57.11 | Qwell[] | BUT...9-911 SHOULD be valid |
22:57.13 | Qwell[] | as should 19911 |
22:57.17 | ManxPower | Qwell, of course |
22:57.18 | Qwell[] | erm, 91911 |
22:57.24 | [TK]D-Fender | Qwell : I do 9x.T right now. Buy in order to take advantage of the "callback from issed calls" and so on I think it'd be easier to switch. it would speed up outside, and only ahve a minor impact on internal. |
22:57.25 | Qwell[] | or something |
22:57.33 | ManxPower | well 91911 should not be valid |
22:57.36 | ManxPower | 911 and 9911 |
22:57.44 | Qwell[] | 91911 makes sense though |
22:57.52 | ManxPower | [TK]D-Fender, users hate to wait. |
22:57.58 | Qwell[] | it won't be a valid exten anyhow, so... |
22:58.01 | ManxPower | My users complain ALL THE TIME about that. |
22:58.22 | [TK]D-Fender | ManxPower : true... but they'd balance that against the ability to sue the missed calls list to INSTANTLY dial DL #'s and save a LOT. |
22:58.25 | FuriousGeorge | why use 9 at all anymore? |
22:58.30 | ManxPower | yes, but 9 + 1 + 916 + number could be misdialed as 91911 |
22:58.31 | [TK]D-Fender | ManxPower : Its a trade-off for sure... |
22:58.47 | Qwell[] | ManxPower: yes, true, but as could 916-xxxx |
22:58.49 | ManxPower | FuriousGeorge, use 9 so users don't have to wait for Digittimeout before calls to extensions are processed |
22:59.03 | ManxPower | And I have to have a long digittimeout or users will yell. |
22:59.06 | Qwell[] | assuming 916 is a valid prefix |
22:59.09 | [TK]D-Fender | FuriousGeorge : if you do and you interal # are like 5xxx, you could have it accept a 5xxx # the instant the 4th digit is entered and make intenal calls convenient. |
22:59.19 | Qwell[] | or, no |
22:59.25 | Qwell[] | any areacode that starts with 6 |
22:59.31 | Qwell[] | so, 9-1-626-xxx-xxxx |
22:59.32 | DeadZen | can e911 trace the call |
22:59.41 | Qwell[] | ManxPower The above could just as easily be misdialed |
22:59.48 | FuriousGeorge | that makes sense. i didnt think about it b/c we are moving to sipphones |
22:59.54 | FuriousGeorge | which you gotta send anyway |
23:00.06 | *** join/#asterisk Kizmet (n=Kizmet@freematrix/sponsor/kizmet) |
23:00.38 | ManxPower | My users would form a mob, come after me with torches and be screaming "BURN THE GEEK" if I made themn press "send" when they are done dialing |
23:00.39 | Qwell[] | 91911 kinda makes sense to me. Users know that in order to dial long distance, they have to do 91- |
23:00.44 | FuriousGeorge | woot, forensic files is on |
23:00.52 | ManxPower | yeah, but 911 is not a toll call |
23:00.55 | Qwell[] | so, in a panic, I could realistically see 91911 be dialed |
23:01.10 | DeadZen | whats the number for 911? |
23:01.16 | Qwell[] | or just 1911 |
23:01.20 | trixter | the real number is 912 |
23:01.32 | ManxPower | Really, I should add a Wait(1) for all 911 calls, so users can panic and hangup when they realize what they did. |
23:01.34 | Qwell[] | ManxPower: I'm just more paranoid I guess. :) |
23:01.35 | DeadZen | wasnt that off the simpsons |
23:01.36 | ManxPower | I think 916 is Toronto |
23:01.42 | DeadZen | when he became a mason |
23:01.46 | FuriousGeorge | cant you see yourself somehow beeing sued too b/c someone didnt know to dial 91 before the 911? |
23:01.50 | trixter | no a stonecutter |
23:01.52 | trixter | they are different |
23:01.56 | DeadZen | ahh |
23:01.59 | DeadZen | sup trixter |
23:02.00 | Qwell[] | FuriousGeorge: No, because 911 and 9911 would also work |
23:02.05 | trixter | not much |
23:02.06 | FuriousGeorge | ah |
23:02.17 | DeadZen | im trying to put sip into my app |
23:02.18 | DeadZen | hehe |
23:02.18 | Kizmet | I have a 'You are now being connected to emergentcy services.' |
23:02.18 | rayvd | 916 area code? |
23:02.20 | Kizmet | :) |
23:02.27 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
23:03.01 | FuriousGeorge | ManxPower: just thought of something, users also get the domain appended to the cid when they call eachother, so i would have to basically set the cid for every extension manually to get it to not do that? |
23:04.57 | FuriousGeorge | i just dont see why showing the domain is the deault behavior. i know where i'm at no one cares |
23:05.16 | FuriousGeorge | * wasnt even dopmain aware till like 6 months ago, right? |
23:06.10 | *** join/#asterisk Nukemizer (n=Nuke@67.137.28.165) |
23:06.59 | Kizmet | how would i do a dial plan match for 911 and 000 in the same line ? |
23:07.07 | Kizmet | i was thinking like [000|911] |
23:07.09 | Kizmet | or somthing |
23:07.18 | Qwell[] | Kizmet: You wouldn't |
23:07.21 | [TK]D-Fender | Kizmet : Just easier to make one Goto the other. |
23:07.39 | Kizmet | :( |
23:07.46 | Kizmet | G.H.E.Y! |
23:07.50 | Dr-Linux | what's 911 in asterisk? do i need to define this extension manually or its some default option? |
23:08.08 | FuriousGeorge | default :) |
23:08.21 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.235) |
23:08.56 | Dr-Linux | FuriousGeorge: how come defualt? :S |
23:09.17 | [TK]D-Fender | Dr-Linux : Ummm, what kind of question is that?! You decide EVERYTHING yourself. You should kno better... |
23:09.23 | FuriousGeorge | and set your zipcode in asterisk.conf and the number for the local pizza place is stored in ${TAKEOUT} |
23:09.34 | Kizmet | <PROTECTED> |
23:10.00 | [TK]D-Fender | .. and did you know the word "gullable" isn't in the dictionary? ;) |
23:10.02 | FuriousGeorge | i the extension 000 or s? |
23:10.08 | FuriousGeorge | lemme go look |
23:10.10 | [TK]D-Fender | Kizmet : thats because you did it BACKWARDS. |
23:10.16 | Kizmet | i did :o |
23:10.17 | FuriousGeorge | on dictionary.com |
23:10.26 | [TK]D-Fender | "s" is not a priority..... |
23:10.45 | [TK]D-Fender | FuriousGeorge : Guess you already fell for it... CHUMP :) |
23:10.51 | FuriousGeorge | LIAR |
23:10.58 | FuriousGeorge | (gets me every time) |
23:11.01 | Dr-Linux | gullable? :S |
23:11.53 | Kizmet | <PROTECTED> |
23:12.08 | Dr-Linux | [TK]D-Fender: actually there are manythings in voicemail... and i saw 911 a lot of time, so i thought its something default. |
23:12.19 | [TK]D-Fender | Kizmet : Show us the WHOLE dialplan not just 1 stupid line from it.... |
23:12.28 | Nivex | I really ought to put a 911 trap in my dialplan that does Playback(no-911-1) |
23:12.38 | Dr-Linux | actually this 911 is new here in pak |
23:12.44 | *** join/#asterisk Seldon1975 (n=someone@toronto-HSE-ppp4239697.sympatico.ca) |
23:12.56 | Kizmet | [TK]D-Fender, If i was to do that it would span about 48 pages on pastebin -_- |
23:13.14 | [TK]D-Fender | Kizmet : How about eveything in that CONTEXT then? |
23:13.16 | Seldon1975 | sorry if this is an RTFM (I've tried to find it) but is there a command that can you tell what * version you are running from the * console? |
23:13.32 | Nivex | Seldon1975: show version |
23:13.33 | Kizmet | heh that is everything in that context |
23:13.33 | [TK]D-Fender | Seldon1975 : "show version" |
23:13.34 | Kizmet | heh |
23:13.39 | Seldon1975 | thanks |
23:13.41 | [TK]D-Fender | Kizmet : just do it. |
23:14.14 | Kizmet | [TK]D-Fender, why should i The only part of the AEL map that you need to be concerned about is the 911 and 000 extensions ;) |
23:14.27 | Seldon1975 | can it tell you the zaptel version? |
23:15.08 | Seldon1975 | Kizmet, D-Fender; can the * console tell you the zaptel version? |
23:15.09 | [TK]D-Fender | Kizmet : It's not that I don't trust you... its just that I DON'T :) I see people hold back parts they don't think are relevent and it nails them EVERY time. |
23:15.15 | [TK]D-Fender | Seldon1975 : no clue. |
23:15.22 | Seldon1975 | i mean Nivex :{ |
23:15.28 | [TK]D-Fender | Kizmet : Fine start with those 2. |
23:15.36 | Seldon1975 | D-Fender is there ay way you know of to check the Zaptel version |
23:15.45 | ManxPower | Kizmet, what verison of Asterisk? |
23:15.49 | Kizmet | [TK]D-Fender, lol the other pages are simply mappings to different states in america, australia, uk, etc etc. |
23:15.51 | Kizmet | 1.2.4 |
23:16.01 | [TK]D-Fender | Kizmet : Point being if you come in here needing help, don't think that everything else is fine :) |
23:16.01 | *** join/#asterisk gorauskas (n=gorauska@66-224-20-131.atgi.net) |
23:16.07 | ManxPower | Kizmet, and that's in extensions.ael? |
23:16.09 | [TK]D-Fender | Kizmet : So pastebin away... |
23:16.14 | Kizmet | ManxPower, yes. |
23:16.22 | DeadZen | jesus |
23:16.32 | DeadZen | vaxvoip charges 5500 for their sip ocx |
23:16.44 | DeadZen | 20 simultaneious calls 1500.. . 100.... is 5500 |
23:16.47 | ManxPower | VAXen were always expensive |
23:16.48 | *** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar) |
23:16.56 | DeadZen | what the hells wrong with them |
23:17.00 | Kizmet | http://pastebin.com/567632 |
23:17.22 | ManxPower | DeadZen, thats pretty standard pricing for telecom |
23:17.34 | DeadZen | seriously? |
23:17.37 | ManxPower | Nortel wants like $8,000 for a PRI card and the software for their stuff |
23:17.44 | ManxPower | DeadZen, Yeah. |
23:17.52 | DeadZen | damn |
23:17.56 | DeadZen | i picked a good business to get into thenm |
23:17.57 | [TK]D-Fender | Kizmet : Your goto is no good. |
23:18.11 | Kizmet | [TK]D-Fender, post a revision maybe :) |
23:18.21 | ManxPower | DeadZen, that's for a ONE PORT PRI card |
23:18.30 | [TK]D-Fender | Kizmet : You are trying to jump to a CONTEXT named "000" and there is no "s" as an exten in this case. |
23:18.43 | Kizmet | whoops :S |
23:18.50 | ManxPower | [TK]D-Fender, is the context required for that? |
23:19.11 | ManxPower | Must be an AEL thing, since you only need a context in a Goto if it's jumping out of the current context. |
23:19.14 | [TK]D-Fender | ManxPower : Certainly not. the target is WITHIN the context. |
23:19.40 | ManxPower | <PROTECTED> |
23:20.01 | [TK]D-Fender | Yup... kinda blatent isn't it :) |
23:20.07 | ManxPower | 1 arg = priority, 2 args = exten and priority, 3 args = context, exten, and priority |
23:20.17 | [TK]D-Fender | They keep doing that.... hiding it in the BIG print :) |
23:20.40 | ManxPower | so his goto is going to extension 000, priority 1 in the current context |
23:21.25 | [TK]D-Fender | ManxPower : actually in AEL does priority even exist? If only by a "label" if present? |
23:21.43 | ManxPower | [TK]D-Fender, I have no idea. AEL is still too new for me to trust it enough to use it. |
23:22.23 | ManxPower | but I do admit that I totally lust for AEL |
23:23.21 | [TK]D-Fender | ManxPower : And even if it WERE stable... who cares?? it doesn't offer anything NEW, it just shortens up the code a little and introduces all new ways to make debugging a ^&%#$ PITA :/ |
23:23.21 | ManxPower | I suppose I should write my dialplan in res_perl, if it's been ported to 1.2x |
23:24.25 | [TK]D-Fender | ManxPower : I haven't touched the "res" stuff yet. Do you mean to basically have all extens call Perl scripts everywhere? |
23:24.41 | [TK]D-Fender | And be treated like "light"-AGI? |
23:25.44 | ManxPower | [TK]D-Fender, THIS is why: http://pastebin.ca/42838 |
23:26.18 | ManxPower | [TK]D-Fender, think of res_perl as AEL, but as Perl instead of some custom bastard of a pseudo language |
23:26.37 | ManxPower | i.e. the perl is resident and not forked each time |
23:27.29 | *** join/#asterisk mattems (n=pronmatt@cust2229.vic01.dataco.com.au) |
23:27.37 | mattems | hey all |
23:27.59 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
23:28.03 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
23:28.14 | [TK]D-Fender | ManxPower : But thats just 1 uber macro :) big deal... mine was almost that big, but because of 1.2 far cleaner looking ;) |
23:28.32 | ManxPower | [TK]D-Fender, that's why I want some kind of language. |
23:28.47 | ManxPower | There's also res_js, but at least I know a bit of Perl |
23:28.55 | *** join/#asterisk Mhohman (n=jvevjon@67.189.92.71) |
23:29.02 | mattems | we are setting up an asterisk system and just having some issues with selecting a card |
23:29.28 | ManxPower | mattems, What specific issues? |
23:29.31 | [TK]D-Fender | mattems : Halmark has never done me wrong.... |
23:30.06 | ManxPower | [TK]D-Fender, also I still have many 1.0.x servers |
23:30.27 | [TK]D-Fender | ManxPower : I just upgraded mine... you can too! We can make it like an intervention! |
23:30.40 | ManxPower | I've already upgraded 4 of my servers |
23:30.47 | ManxPower | I still have at least 4 more to do |
23:30.53 | mattems | we want to have two lines in and 2 lines out and one of the lines will be a duet line |
23:31.00 | mattems | can asterisk pick up the duet? |
23:31.04 | [TK]D-Fender | last Friday I upgraded my A104d firmware, Wanpipe, and full * from 1.0.9. |
23:31.11 | ManxPower | mattems, no |
23:31.23 | ManxPower | but if you define "duet" we might have suggestions for a work around. |
23:31.30 | mattems | ok |
23:31.33 | ManxPower | also what kind of lines? lines to analog phones or lines to the telco? |
23:32.09 | mattems | duet: two numbers running on the same line, if people call from one number it will ring differently |
23:32.40 | ManxPower | mattems, You might be able to do it with Asterisk. In the USA we call it "RingMaster" or "Distinctive Ring" |
23:33.04 | ManxPower | There's supposed to be support for that in zaptel, but I've never know anyone that has used that feature. |
23:33.11 | *** join/#asterisk loick (n=loick@APuteaux-151-1-82-111.w86-205.abo.wanadoo.fr) |
23:33.14 | mattems | and the lines we have are the ones from the telco |
23:33.27 | mattems | 2 from the telco and 1 voip number |
23:33.35 | mattems | can we hook em all up |
23:33.38 | mattems | ?? |
23:33.43 | mattems | and with what card |
23:33.44 | mattems | ? |
23:33.54 | ManxPower | the TDM400P card will provide up to 4 ports |
23:34.06 | mattems | ill have a look now |
23:34.09 | ManxPower | but you will not be happy with plugging your VoIP adapter into an Asterisk card |
23:35.05 | mattems | pretty much we just want to be able to dial out on the voip line to save costs |
23:35.13 | mattems | we are a small org. |
23:35.38 | DeadZen | there's this one sip component i found $299 |
23:35.48 | DeadZen | but the sites down and the emails down and the demo works great.. |
23:35.53 | mattems | lol |
23:35.54 | ManxPower | DeadZen, what are you specifically looking for? |
23:35.56 | [TK]D-Fender | mattems : You wouldn't even NEED *. An SPA-3000 could do that for you. |
23:35.58 | DeadZen | got i hate murphy and her stupid laws |
23:36.32 | *** join/#asterisk Soul (n=Soul@87-196-41-22.net.novis.pt) |
23:36.49 | DeadZen | ManxPower: its a component i need so I can introduce sip incoming/outgoing voip to my app |
23:36.53 | *** part/#asterisk cogineo (n=soner@81.215.105.114) |
23:37.21 | ManxPower | Why not just use EAGI in Asterisk? |
23:37.30 | DeadZen | im not familiar with it |
23:37.42 | *** join/#asterisk WAudette (n=WAudette@c-67-170-156-3.hsd1.or.comcast.net) |
23:37.56 | mattems | [TK]D-Fender: i dont think the SPA-3000 will do the trick |
23:37.58 | ManxPower | It's like regular AGI, but better! (better = you have access to incomning audio and can generate outgoing audio |
23:37.59 | DeadZen | so far i found $799 with royalties... or $5500 with a max user limit and something perfect for $299-$399 but the site and author are awol |
23:38.29 | DeadZen | yah this one i could get up and running right now though |
23:38.33 | *** join/#asterisk DirtyD (n=Miranda@ool-44c24fb7.dyn.optonline.net) |
23:38.55 | DeadZen | i even hexed the demo so it uses my servers addres by default hehe |
23:38.58 | DeadZen | so i know it'll work fine |
23:39.05 | DeadZen | audio quality is even better then sjphone |
23:39.05 | WAudette | mattems: Have you seen the new SPA-9000? |
23:39.06 | DirtyD | How can I connect Asterisk to a Avaya Definity extension? |
23:39.16 | [TK]D-Fender | mattems : Why not? If you have a simple VoIP provider it'd do just fine... |
23:39.21 | *** join/#asterisk websae (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
23:39.41 | [TK]D-Fender | mattems : I've used all kinds fo SPA models for even direct PBX bridging... |
23:39.45 | DirtyD | Is there any type of converter that I can use to convert a digital Definity jack to something Asterisk can work with? |
23:40.13 | WAudette | DirtyD: That's a good question. Been wondering the same thing myself. |
23:40.28 | [TK]D-Fender | DirtyD : Intel makes a low density one, but Citel's are a great deal. |
23:40.34 | DeadZen | http://cti-research.com/files/siptest.rar that's what i want |
23:40.49 | DeadZen | but not the demo i want to buy the product but the companies gone haha |
23:41.08 | ManxPower | DirtyD, even if there was, those sorts of things never support CPC and that's bad, |
23:41.16 | ManxPower | imagine 24 hour long voicemails |
23:41.29 | mattems | [TK]D-Fender: here is a scenario -> we have 6 people in office all need to be able to dial out from their phone through a selected number and we all need to be able to accept incoming calls from the two incoming sales / support lines |
23:41.30 | ManxPower | all but the first 30 seconds of which is dialtone |
23:42.09 | *** part/#asterisk Tamarisk (n=adrian@user-2899.lns1-c8.dsl.pol.co.uk) |
23:42.21 | [TK]D-Fender | mattems : So you are looking at * as being your full PBX then? Not just a front end to 'extra' features? |
23:42.54 | mattems | yes |
23:43.05 | mattems | :) |
23:43.14 | DeadZen | are dual phones like |
23:43.20 | DeadZen | before their time and crappy? |
23:43.44 | [TK]D-Fender | mattems : Oh.. in that case yeah.. you want * :) |
23:43.58 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
23:44.07 | Dr-Linux | ManxPower: you just said something about AGI and EAGI, can you tell me what can i use communicate with external API , like socket based |
23:44.09 | Dr-Linux | ? |
23:44.26 | ManxPower | Dr-Linux, I don't understand |
23:45.17 | DirtyD | D-Femder: I only need one to handle 1 extension. |
23:45.46 | DeadZen | dirtyd... he was talking to me about something else |
23:46.07 | Dr-Linux | ManxPower: i mean there is an API script at other non-asterisk server. i wanna call it from dialplan on socket based communication ... |
23:46.10 | Qwell[] | ouch...Sun is about to get a reaming from somebody at my work |
23:46.13 | DeadZen | brb |
23:46.15 | Qwell[] | poor Sun tech |
23:46.21 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
23:46.25 | DeadZen | Sun sux |
23:46.43 | DeadZen | I know I used to work on their 32 processor systems |
23:46.43 | Qwell[] | a RAID array died, which holds a ton of stuff that we use to connect to outside vendors... |
23:46.44 | DeadZen | had two of em |
23:46.50 | DeadZen | slow as crap.. everything craps out on it |
23:46.54 | Qwell[] | which means people aren't getting paid from crdeit card services... |
23:47.02 | DeadZen | nice |
23:47.15 | Qwell[] | millions of transactions per day |
23:47.18 | ManxPower | Dr-Linux, AGI is like CGI for Asterisk |
23:47.21 | Qwell[] | that poor tech :( |
23:47.27 | mattems | [TK]D-Fender: so i cant plug my voip line into asterisk? |
23:47.30 | DeadZen | cgi for asterisk? |
23:47.37 | DeadZen | you can pass audio through cgi? |
23:47.37 | ManxPower | Is RAID REALLY more reliable than non-RAID? |
23:47.44 | [av]bani | http://www.jalopnik.com/cars/news/more-on-the-enzo-incident-on-pch-with-photos-156121.php |
23:47.45 | Qwell[] | ManxPower: not really |
23:47.47 | ManxPower | DeadZen, EAGI supports passing audio |
23:47.52 | DeadZen | yes it is... |
23:47.53 | Qwell[] | but, a few drives died at once |
23:48.03 | DeadZen | I got two raid 5 sets |
23:48.08 | DeadZen | works great... |
23:48.11 | Qwell[] | "As of 13:08 Technical support determined there are several hard drives that comprise a larger disk array in Tempe that has lost power. The vendor, Sun is on site and the disk array is being rebooted." |
23:48.13 | [av]bani | i use raid1... |
23:48.19 | DeadZen | raid 5 is real raid |
23:48.26 | DeadZen | the others are cheap labor intensive options |
23:48.32 | Dr-Linux | ManxPower: yeah, i know but it only interact with the file which located at /var/lib/asterisk/agi-bin/ at same box. but if this file is on other non-asterisk server? |
23:48.34 | CunningPike | Greets - is there a way to read SIP responses like 'SIP/2.0 486 Busy Here' in the dialplan? |
23:48.40 | DeadZen | i bust a drive i put a new one in and the others rebuild it |
23:48.46 | DeadZen | but it all comes down to your raid controller |
23:48.48 | DeadZen | software raid is a joke |
23:48.59 | DeadZen | and you need a reputable provider.. I like adaptec.. they make good shit |
23:49.04 | CunningPike | I'm trying to differentiate between DND, phone not registered, call rejected etc for voicemail purposes |
23:49.28 | DeadZen | my 2400a's still chug... good throughput... i feel safe putting 350 gigs of stuff i cant lose on it |
23:49.38 | MooingLemur | CunningPike: I think some of that information is in ${HANGUPCAUSE} |
23:49.57 | CunningPike | Hmm - doesn't that only cover ZAP? |
23:49.57 | MooingLemur | that's for Zap channels I think though |
23:50.21 | CunningPike | I've tried ${DIALSTATUS}, but it returns 'BUSY' for everything |
23:50.41 | CunningPike | ${CAUSECODE} is blank :( |
23:50.43 | Qwell[] | "Card Services vendors cannot pickup files for <censored> products (sev 3). Also, there are up to 600 correspondent banks that are affected because of a file needing to be send that contains card holder activity (sev 4). |
23:51.03 | DeadZen | i dont like sun |
23:51.12 | DeadZen | i like freebsd, gentoo and redhat fc4 |
23:51.15 | DeadZen | in that order |
23:51.22 | Qwell[] | and no ACH...man... |
23:51.28 | DeadZen | ouch |
23:51.29 | DeadZen | brb |
23:51.31 | Qwell[] | this is going to seriously affect business |
23:51.38 | CunningPike | And, I don't think sipGetHeader will work, because it's not a header |
23:51.50 | DeadZen | you guys need to look into failovers |
23:51.51 | Qwell[] | So, if you guys don't get your checks direct deposited on time...now you know why :P |
23:52.00 | Qwell[] | DeadZen: it has failover |
23:52.03 | Dr-Linux | ManxPower: the question i asked is not possible in to asterisk? |
23:52.23 | DeadZen | and theres still millions of transactions lost? |
23:52.25 | MooingLemur | CunningPike: according to http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+DIALSTATUS "Note: In order to obtain useful DIALSTATUS information when dialing a peer you will need to have qualify=yes in that peer's definition (e.g. in sip.conf or iax.conf). " |
23:52.34 | Qwell[] | DeadZen: Not lost...just not able to be gotten right now |
23:52.37 | websae | Qwell: what do you do? |
23:52.38 | CunningPike | Yup - done that :D |
23:52.41 | MooingLemur | heh |
23:52.48 | Qwell[] | websae: programmer at a bank |
23:52.48 | MooingLemur | I'm out of ideas :) |
23:52.54 | CunningPike | Me too :) |
23:52.54 | DeadZen | im sure you got a handle of it |
23:53.01 | websae | which bank eeeks |
23:53.06 | Qwell[] | websae: Can't say :) |
23:53.14 | DeadZen | I wrote an online bank before |
23:53.16 | websae | Proprietary information, understandable |
23:53.17 | Qwell[] | I'd get pretty reamed myself, right now...heh |
23:53.34 | websae | would I know the bank here in Milwaukee, WI? |
23:53.41 | websae | by the way anyone here from Wisconsin? |
23:53.42 | Qwell[] | websae: We're a major US bank |
23:53.47 | Qwell[] | so...I'd hope so |
23:54.08 | websae | qwell: fantastic |
23:54.29 | websae | qwell: good thing you mostly likely have redundancy on a "Major US Bank" network ;) |
23:54.37 | Qwell[] | oh, tons |
23:54.54 | DeadZen | doesn't sound like it helped much ;-) |
23:54.56 | Qwell[] | not sure why it isn't working...it should have moved to a different server |
23:55.00 | Qwell[] | DeadZen yeah...no clue why |
23:55.11 | DeadZen | maybe it was an inside job |
23:55.11 | DeadZen | heh |
23:55.17 | Qwell[] | nah |
23:55.17 | DeadZen | like from office space |
23:55.26 | websae | i have yet to meet an asterisk user from "wisconsin" |
23:55.27 | websae | hahaha |
23:55.37 | Qwell[] | websae: Kris Kielhofner |
23:55.43 | Qwell[] | of astlinux fame |
23:55.52 | websae | ohh yes |
23:55.56 | websae | Lake Geneva |
23:56.00 | Qwell[] | yeah |
23:56.00 | Morex | Got another queues question |
23:56.02 | websae | is he ever on here? |
23:56.05 | Morex | How would you put a caller on hold |
23:56.11 | Morex | Then get him/her back again? |
23:56.13 | Qwell[] | websae: I'm told he doesn't IRC much, if at all |
23:56.25 | Qwell[] | Morex: That's usually a function of the phone itself |
23:56.28 | websae | ahh---do you run astlinux? |
23:56.37 | Qwell[] | websae: Nope...I never have, actually |
23:56.42 | Morex | Qwell: Not in this case |
23:56.53 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
23:57.49 | websae | www.kriscompanies.com |
23:57.52 | websae | i wish i had his job :) |
23:58.44 | Qwell[] | Morex: What type of phone? |
23:59.06 | Morex | Just a regular one |
23:59.08 | Morex | Not using SIP |
23:59.14 | Morex | Or VOIP for that matter |
23:59.17 | Qwell[] | how is it connecting to *? |
23:59.17 | Morex | Legacy PSTN stuff |
23:59.23 | DeadZen | damnit im bummed |
23:59.30 | Morex | Through an Avaya Index and an E1 PRI |
23:59.32 | DeadZen | my software almost had full voip |
23:59.45 | DeadZen | and i was gonna code the web app to auto configure the client |
23:59.47 | DeadZen | boo fucking hoo |
23:59.49 | Morex | So DTMF only |