irclog2html for #asterisk on 20060222

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
00:01.11kippiwhen I do *CLI> iax2 show
00:01.12kippiNo such command 'iax2 show' (type 'help' for help)
00:01.24robin_szkippi:
00:01.35robin_szthe reason is that there is no such command iax2 show
00:01.41benjkyou need to tell it *what* to show
00:01.48benjkiax2 show peers
00:01.51benjkfor example
00:02.03robin_sztry help iax2 to see what you can type
00:02.23kippiah ok
00:02.47kippiany ideas why i am getting this error?
00:02.48kippiFeb 22 00:01:50 NOTICE[9082]: chan_iax2.c:3918 register_verify: No registration for peer '1001' (from 10.6.10.149)
00:03.14robin_szcould be any number of reasons
00:03.19*** join/#asterisk sergeus|w (n=sergeus@ippe-245.ippe.ru)
00:03.43robin_szpaster yer iax2.conf some place (less the passwords) .. maybe someone can see
00:05.58kippihttp://pastebin.ca/42712
00:07.33clyrradis it possible to define custom variables for a SIP user in sip.conf, that can be referenced from extensions.conf?
00:09.28clyrradI 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)
00:12.28DorphalsigHello
00:13.08DorphalsigI 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
00:13.25kippirobin_sz: any ideas?
00:13.40Dorphalsigcan anybody give me a hand?
00:14.28*** join/#asterisk dlublink (n=dlublink@modemcable114.38-201-24.mc.videotron.ca)
00:14.50dlublinkhow do I disable "Native Bridge"s in asterisk
00:14.56dlublinkeverytime that message appears I lose all audio
00:15.23*** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)
00:15.28Dorphalsignative bridge is a "flash" where users redirect
00:15.40Dorphalsigthe call to another extension
00:15.54file[laptop]no, it isn't
00:16.01delmarhey 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.
00:16.04delmarany ideas?
00:16.18file[laptop]dlublink: what's the two protocols? SIP?
00:17.45MoutaPTdelmar NAT?
00:17.58delmarnevermind. ill copy the entire working config dir from the working box. if its still broken ill blow away the debian install.
00:18.06delmarMoutaPT, nope. local LAN.
00:18.40delmarMoutaPT, 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....
00:19.10delmarSo 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...
00:19.20MoutaPTdelmar i'm on the phone brb
00:19.39delmarso 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.
00:19.40delmarok
00:19.50delmarim gonna go mess with stuff anyway. bbl
00:19.58DorphalsigHi!
00:20.11DorphalsigI need to have a PHP script trigger a dialing process
00:20.17Dorphalsigis there any API I can use?
00:20.20DorphalsigI tried PHPAGI
00:20.39Dorphalsigbut it seems its only to be used as an AGI in the dialplan :$
00:24.31glm2kDorphalsig: i think what you are looking for is a .call file
00:25.05glm2kDorphalsig: mine for example is a simple bash script i plugged into nagios
00:25.30glm2kDorphalsig: so a PHP routine should be easy to do
00:29.58brookshireDorphalsig: /var/spool/asterisk/outgoing
00:31.01mzoanyone 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)
00:32.50glm2kmzo: i don't remember it being that long.
00:33.45*** join/#asterisk yoll (n=BoraScrp@85.108.204.58)
00:34.38mzoi 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)
00:39.26glm2kmzo: i just checked my archives. it was connected on the same day
00:39.39mzogot any ideas what I'm doing wrong?
00:40.33Abydos313is your firewall blocking?
00:40.59mzoum, 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
00:41.00glm2kiax? last i read it punches through firewalls.
00:41.16mzoiax, the firewall slayer (+5 to demons)
00:41.19*** part/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net)
00:41.22glm2klol
00:41.38glm2kyou must be a pally in your free time
00:41.44mzoangband addict :p
00:41.54mzonow if i could find a way to asterisk to play angband while I'm there :P
00:41.55mzo...
00:42.05glm2khehe
00:43.08glm2kiirc, fwd will allow you to register using SIP or IAX2 (or both? *shrugs*)
00:43.22glm2kcan you register thru SIP?
00:43.55mzoi didn't try sip, they said to do iax2 because it was better?
00:44.03glm2ki 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.
00:44.25Abydos313Nice!
00:44.42[hC]Considering the little to no documentation on XML file formats it wants em all in, heh.
00:44.45Abydos313how 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)
00:44.52glm2kin 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
00:45.12Abydos313how much is that in USD
00:45.29[hC]450-600 range
00:45.37[hC]probably low to mid 5's
00:45.39Abydos313big bucks for one phone
00:45.43[hC]yep
00:45.47Abydos313it must do alot
00:45.47mzocheaper than a DHD
00:45.47[hC]color touch screen with camera support
00:45.49glm2kmzo: 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.
00:45.59mzoglm2k, so what else to do? :P
00:46.06mzoglm2k i did that with the iax2 option. and that was a day ago
00:46.09Abydos313sweet, what kind of camera support? only cisco or any
00:46.14[hC]Cisco only i believe.
00:46.20Abydos313how 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..
00:46.41glm2kmzo: run a debug? or ethereal?
00:46.44Abydos313worth asking to see what they are going for
00:47.05Abydos313so are they worth the money?
00:47.34mzodebug jussays the same thing it's sadi
00:47.41Abydos313that's same price and new dell with flatscreen :))
00:48.45glm2kmzo: rejected due to what?
00:48.52glm2kmzo: it could be a codec issue.
00:50.07mzohttp://pastebin.com/566022 that's what i get.
00:54.37mzoany ideas? :P
00:55.03mzohttp://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)
00:56.48glm2ki can't find what cause code 29 stands for :(
00:57.33websaeany 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)
00:57.41mzobleh, is that bad? :P
00:58.06*** join/#asterisk Gir19 (i=Gir@67.189.110.174)
00:58.36glm2kmzo: nah, just have to find what it is trying to say.
00:59.25MoutaPTG723 is available  free for ASterisk?
00:59.45websaeany suggestions anyone?
00:59.56glm2kdid you double check your password?
01:00.00mzoyes
01:00.07websaeanyone know why my asterisk server can't see my sip phones trying to connect?
01:00.09mzoif i change it on the site it should be right?
01:00.10websaeany suggestions?
01:00.15websaei just compiled asterisk---
01:00.24glm2kwebmind: that on a lan?
01:00.29glm2ker, sorry.
01:00.33glm2kwebsae: that on a lan?
01:00.43websaeat my colo
01:01.25*** part/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net)
01:01.38glm2kmzo: no reason for you to change it. it could be that your number has not been provisioned yet. fwd was down yesterday i think.
01:01.55mzoI did http://asteriskathome.sourceforge.net/handbook/index.html and the instructions for fwd, and no joy
01:01.59mzoit's been like this since sunday or so
01:02.37websaei don't know what's up with it
01:02.41websaewhy nothing will connect
01:02.58websaei can't see anything in the sip debug for registration packets or anything
01:03.31glm2kwebsae: 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)
01:04.06websaeshouldn't be an issue, registers with my other colo fine
01:04.24websaeit's odd because i see nothing at all
01:05.44glm2kwebsae: "nothing at all" sounds like a firewall.
01:05.52glm2kwebsae: specialyl with sip
01:06.30Gir19websae: have you also checked to make sure the ip address hasn't been changed as well.
01:06.56websaeyep
01:07.06websaethe ip addresses are static
01:07.19websaei can't see anything, this quite odd
01:07.22websae*this is
01:07.31brockj49464Asterisk does NOT do TOS on SIP correct?  I have to use iptables to tag the packets correct?
01:07.57justinuit doesn't do TOS on RTP packets
01:08.00justinubut does on SIP
01:08.04websaeany other suggestions
01:08.05justinuiirc
01:08.13websaeit's not a firewall issue
01:08.36websaecause i should be able to at least see the packets hitting the server
01:08.37websaebut nothing is
01:08.59Gir19sounds to me more of a routing issue than a firewall issue.
01:09.16websaerouting issue
01:09.21websaeon the server end?
01:09.54brockj49464what 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)
01:10.18Gir19no, 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.
01:10.26websaewhat kind of routing issue could it be
01:10.59GoRKanyone 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?
01:11.10websaei have an asterisk server at a COLO that i just compiled asterisk on
01:11.24GoRKnot "real PTT" like cell phones or the like
01:11.28websaei tried setting up a atest account and setting up my sip phone to connect
01:11.44websaebut it doesn't register
01:11.52websaei don't see anything trying to register in the CLI
01:12.44websaei then tried with my softphone---still same thing
01:12.54Gir19websae: how is the phone trying to connect? IE: phone > router > internet > router > *
01:12.55websaei don't see anything trying to register at all
01:13.15websaewell 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)
01:13.31GoRKwebsae: i assume you have enabled debug with 'sip debug' ?
01:16.10websaecorrect
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01:16.16websaeand set verbose to 10000000000000000000000000
01:16.16websaelol and nothing
01:16.17Gir19mmm, netsplit
01:16.17delmargreat
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01:16.19*** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)
01:16.19delmar[7012]
01:16.19delmartype=friend                     ; either "friend" (peer+user), "peer" or "user"
01:16.19delmarregexten=7012
01:16.19delmarcontext=testing
01:16.19delmarsecret=7012
01:16.20delmarcallerid="Testphone" <7012>
01:16.20delmarhost=dynamic            ; we have a static but private IP address
01:16.20GoRKwebsae: 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
01:16.20delmarnat=no                          ; there is not NAT between phone and Asterisk
01:16.20glm2kholy macaroni, that's some split
01:16.20delmarcanreinvite=no        ; allow RTP voice traffic to bypass Asterisk
01:16.20delmardtmfmode=info                   ; either RFC2833 or INFO for the BudgeTone
01:16.20delmar;mailbox=7012@local  ; mailbox 1234 in voicemail context "default"
01:16.20delmardisallow=all                    ; need to disallow=all before we can use allow=
01:16.20delmarallow=ulaw
01:16.20delmarallow=g729                     ; Pass-thru only unless g729 license obtained
01:16.20justinu~pb
01:16.20jbot[pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
01:16.20delmarqualify=50
01:17.08delmardefaultip=10.20.1.248
01:17.08delmargrr
01:17.08delmarsorry. hit the wrong key guys
01:17.09GoRKargh ever heard of a pastebot?
01:17.09websaeUA----what's that?
01:17.09justinusupposedly there's some net backbone problem
01:17.09GoRKuser agent -- your softphone or hardphone or whatever is trying to register
01:17.09websaethey are all setup right
01:17.09websaejust is frustrating i guess, not knowing what's blocking this
01:17.09Gir19websae: what is the physical configuration?
01:17.09websaehow can i tell if that port is open?
01:17.09GoRKwell you can tell for sure with netcat
01:17.09websaehow does that work?
01:17.09GoRKstop asterisk on the box, set netcat as a listener on port 5060/udp and then use netcat somewhere else to send it some data
01:17.09delmari 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.
01:17.09websaenetcat is that on fedora core
01:17.09websaecan you give me an example GoRK
01:17.10websaeof what to do?
01:17.10GoRKnetcat is available for everything; whether or not it's installed by default .. can't help you there
01:17.10justinuyum install netcat
01:17.50*** join/#asterisk slePP (n=slepp@S0106000f663692da.ed.shawcable.net)
01:17.55GoRKwebsae: on the server: nc -l -p 5060 -u
01:18.21GoRKwebsae: on the client echo test | nc -u -p 5060 your.server.name
01:18.41mitchelocdoes anyone here know if the sipura devices have an API to them?
01:19.05GoRKwebsae: 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
01:19.21GoRKwebsae: if traffic flows you will see "test" echoed to the remote terminal
01:19.27websaeok
01:19.29websaelet me try this
01:20.02Gir19mitcheloc: from what I can tell about sipura's is that you can only access them via http.
01:20.33mitchelocGir19: 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)
01:21.55GoRKwebsae: 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
01:22.25websaeGorK: that didn't work on the client side
01:22.29websaethat command you told me to do
01:22.43GoRKsorry what was the error
01:22.54websaejust brought up the nc commands
01:22.57GoRKim not on a machine with netcat right now
01:22.57GoRKoh
01:23.10GoRKhang on
01:23.13websaeok
01:23.24Gir19mitcheloc: so you mean you are trying to initiate a call by a command instead of just picking up a handset and dialing?
01:23.38mitchelocGir19: yes, i tend to get very lazy ;)
01:23.47MoutaPTis it G723 available free to Asterisk?
01:24.07GoRKwebsae: ok sorry the client syntax should be:
01:24.14websae...
01:24.14websaeyep
01:24.15websae?
01:24.20GoRKecho test | nc -u your.server.here 5060
01:24.38GoRKthe -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)
01:25.15websaehrm nothing on the server side
01:25.24Gir19mitcheloc: 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?
01:25.53GoRKwebsae: 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
01:26.01mitchelocGir19: yes, i just tell it to call another extension, so first it calls my handset, i pickup, then it rings another extension
01:26.04websaehrm
01:26.06websaenothing at all
01:27.03websaeGoRK: could try sending a packet
01:27.06GoRKwebsae: do this on the server: netstat -l -p -n | grep 5060
01:27.07mzodoes 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)
01:27.21Gir19mitcheloc: 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.
01:27.25GoRKwebsae: that will tell you if anything else is listening on udp/5060
01:27.58GoRKif you run it while netcat is listening you should see nc in the list
01:27.58websaewent to the next line
01:28.20GoRKok so it's not another app on the machine... i'd ask your telco if port 5060/udp is open
01:28.20mitchelocGir19: 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...?
01:28.28GoRKerr isp/colo whatever
01:28.32GoRKnot the telco
01:28.37GoRKunless maybe they are one in the same
01:28.47websaeso it's their side
01:28.54websaedarnit
01:29.06websaehey well thanks for your help GoRK :)!
01:31.33*** join/#asterisk Chai_Sangeen (n=Chai@c-24-61-4-191.hsd1.ma.comcast.net)
01:31.40Chai_Sangeenhello everyone
01:31.42Gir19mitcheloc: 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.
01:32.03mitchelocGir19: i was hoping to bypass asterisk if possible
01:33.35*** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka)
01:33.44Gir19mitcheloc: 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.
01:34.30mitchelocahh good point
01:34.55mitchelocperhaps I should e-mail and ask them to expand their firmware ;)
01:34.57Gir19mitcheloc: 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)
01:36.40mitchelochow would someone go about contacting a developer or project manager (i guess whoever makes decisions at sipura?)?
01:36.56robin_szoh, bucking follocks :(
01:37.05file[laptop]mitcheloc: sacrifice a goat, or pig
01:37.10Gir19lol
01:37.15riddleboxis it possible to start an agi script without calling into asterisk first?
01:37.27mitchelocfile: i would seriously do that because i'm actually serious about this though
01:37.30robin_sz2 hours spent messing around setting up presentation numbers etc, and my provider has sett it all to "withheld number" .. grr
01:37.35GoRKfile: better than getting in touch with someone like that at cisco. you have to sacrifice a human baby
01:38.02mitchelocwell what are the chances someone in here can put me in contact with one of those people? ;)
01:38.05websaefile: how much to port an 800 number?
01:38.13file[laptop]websae: $25
01:38.22file[laptop]plus $1.95/mth standard fee
01:38.28file[laptop]and another $25 if you want it NOW NOW NOW FAST FAST FAST
01:38.33robin_szand your first-born
01:38.34Gir19GoRK: Sipura is an affiliate of Cisco. lol
01:38.42websaewhat's the incoming rate on that?
01:38.43mishehuanybody 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.
01:38.51file[laptop]2 cents per minute, 6/6
01:39.02*** part/#asterisk freat (n=freat@h-72-244-84-43.chcgilgm.covad.net)
01:39.04file[laptop]for US48
01:39.07file[laptop]inbound...
01:39.21websaealright, thanks much
01:39.23websaewhat's the link
01:39.27mitchelocDo you think if I flew out to their offices they would speak with me?
01:39.36Gir19mitcheloc: the only way I know is to use there webcomments / questions email addy. sipura-webmaster@cisco.com
01:39.37file[laptop]for porting? there is no link really, you just email me....
01:39.42file[laptop]it's a manual procedure thing
01:39.46websaeah
01:39.51file[laptop]I talk nice to certain people and they get it done for me quickly
01:39.56file[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)
01:40.22mitchelocsipura's offices are in san jose mm...thats not too far from orange county ;)
01:40.32mitcheloclol ya'll probably tired of me rambling about this
01:40.38tuxinator_linuxmitcheloc: you in OC?
01:40.45mitchelocyes sir
01:41.00tuxinator_linuxa bunch of us are
01:41.01livindedif i use the regular source can i use "make update" to update to newer versions or dos thta only work with the cvs builds?
01:41.41mitcheloctuxinator, are there any groups or anything for asterisk around here?
01:42.10tuxinator_linuxLet's start on
01:42.15tuxinator_linuxone
01:42.26*** join/#asterisk SwK (n=Silik0nJ@12-219-147-107.client.mchsi.com)
01:42.33Gir19I am still being annoyed by my fax and poping noise on my asterisk server. I am starting to run out of ideas.
01:42.40mitchelocthat sounds good to me, but i only know you, and one or two other people that i just slowly started introducing to asterisk
01:42.56SwKwho's doing that asterikast thing?
01:43.13*** join/#asterisk asteriskmonkey (n=phil@69.158.149.183)
01:43.20asteriskmonkeyhey
01:43.21file[laptop]SwK: I dunno but Russell and I were making fun of it earlier today :D
01:43.21Gir19I've got a whole team I direct for asterisk here in Portland, OR
01:43.36mitchelocGir19: mind if i pm you?
01:43.37SwKfile[laptop] someone needs to explain to him but TDM means
01:43.38tuxinator_linuxasterikast?
01:43.41asteriskmonkeyanyone know much about ztmonitor ? and where volume levels should be for good echo cancellation?
01:43.51Gir19mitcheloc: go for it.
01:43.53file[laptop]SwK: too technical!
01:44.23Gir19asteriskmonkey: they should be as close to center as possible.
01:44.23SwKasteriskmonkey there is no specific gain level ... it all depends on the switch you are connected to
01:44.23file[laptop]unfortunately the server is overloaded so I can't download the episode
01:44.48SwKfile[laptop] they have a pretty fast torrent
01:45.01file[laptop]I'm too lazy though
01:45.02tuxinator_linuxSwK: post it
01:45.10tuxinator_linuxSwK: the torret link
01:45.30file[laptop]SwK: thankies
01:45.34SwKthe torrent link is on their website
01:45.36asteriskmonkeySwK 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?
01:45.55SwKasteriskmonkey: have you talked to your carrier?
01:46.19SwKasteriskmonkey: you could be over-driving
01:46.36SwKecho is one of those PITA things to get rid of
01:46.46asteriskmonkeySwK: carried says its not them .. over driving? mmm i dont think so my rx and tx are in the - range
01:47.05file[laptop]I have a can of Echobgone(tm)... I'll sell it to you!
01:47.21SwKasteriskmonkey: I have 1 box with a T1 from qwest and -9 of pad on the RX and -6 on the TX side
01:47.28xachenwhat you guys making fun of?
01:47.41GoRKi 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
01:47.43X-Robfile[laptop], geez, I've been looking for EchoBgone(tm) for ages!
01:47.48file[laptop]I shouldn't download this here...
01:47.51X-RobIs that the aerosol or the roll-on?
01:47.51GoRKi have an ata186 that echos no matter what i do
01:47.54file[laptop]I should download it on the Mini
01:48.06SwKxachen people doing howto videos and not knowing proper terminology
01:48.10file[laptop]although it's working...
01:48.11xachenoh
01:48.14xachenAsterikast or whatever?
01:48.17file[laptop]yeah
01:48.18SwKteah
01:48.25xachenI heard something bout that
01:48.27asteriskmonkeyswk : im at -4rx and -8tx
01:48.28xachenWhat did he confuse?
01:48.44justinuasteriskmonkey: what kind of phones?
01:48.59SwKi forget what he said but when he defined TDM is wasnt time domain mux'ind
01:49.03asteriskmonkeyive gone throught, iaxys, polycoms, atcoms.. everyting :P
01:49.21file[laptop]SwK: did he?
01:49.21justinuasteriskmonkey: :(
01:49.21SwKit was some king of time derived multiplication ro something
01:49.22file[laptop]O.o
01:49.30SwKyeah
01:49.37xachenhaha
01:49.38file[laptop]time division multiplexing foreva!
01:49.38asteriskmonkeyits not all the time so its 100% of the time though so its anoying
01:49.50asteriskmonkeyive request a 1004hz test so i can fine tune
01:50.25justinuplease clarify: (17:49:47) asteriskmonkey: its not all the time so its 100% of the time though so its anoying
01:50.35asteriskmonkeyecho is not all the time
01:50.42SwKasteriskmonkey: 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
01:50.43asteriskmonkeyand it is usually only on 1 end at a time
01:50.48justinuok
01:51.04justinuequally on far end vs. near end?
01:51.05asteriskmonkeySwK: yep checked all that :P
01:51.11Chai_SangeenI 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
01:51.31SwKwell turn down the tx 3 more DB
01:51.50asteriskmonkeyjustinu: yes , only happens usually on orginator .. so if i call i get echo if someone calls me i dont but they do
01:52.01SwKasteriskmonekey: is this a local PRI or a dedicated LD circuit?
01:52.20asteriskmonkeyits 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)
01:52.47lunaphytetonight is dumb question night, right?
01:52.58justinuk
01:53.45lunaphytewhy would i want more than 1 context?
01:53.50asteriskmonkeyits been 8months and spent a shit load on the te406 in hopes of echo cancelling love with no luck :(
01:54.00asteriskmonkeyand digium guys around?
01:54.01Gir19nothing is a dumb question unless you have not Read the FM
01:54.01justinuwow
01:54.07SwKso then get an outboard echo can
01:54.08justinuasteriskmonkey: sounds like a nightmare
01:54.33*** join/#asterisk froguz (n=froguz@224-139-222-201.adsl.terra.cl)
01:54.35lunaphytei guess it's less of a 'how do i do this' and more of a 'what are some applications' question...
01:54.35Corydon76-homeasteriskmonkey: what kind of circuit are you trying to echo cancel?
01:54.39mitchelocfile: may i pm you a quick question?
01:54.40asteriskmonkeyjustinu: yep :P 20k in bills and still an echo syste
01:54.41justinuyou guys know that rx/txgain parameters are not dB, right?
01:54.44Chai_SangeenThe only problem is that I have to manully change the spa3k IP if the spa3k ip changes, to place an outgoing call
01:54.51asteriskmonkeyCorydon76-home: PRI NI2
01:55.01justinuasteriskmonkey: i'm worried I might be in a similar situation :(
01:55.05Corydon76-homelunaphyte: do you want people who call into your system to be able to make long distance calls out?
01:55.27Corydon76-homeasteriskmonkey: odd, I rarely have echo problems with a PRI
01:55.49Corydon76-homeasteriskmonkey: Are you hearing echo or is the other side hearing echo?
01:55.53Gir19lunaphyte: you would want to have more than one context to help seperate networks or help keep track of things and better documentation built in.
01:55.57asteriskmonkeyboth
01:56.13Corydon76-homeasteriskmonkey: are you using echotraining?
01:56.14lunaphyteCorydon-w: i think yes, but not just anyone.
01:56.15SwKI think he's probably oer driving since he's in a colo facility probably with little more then a corss connect from the carrier
01:56.30asteriskmonkeyyes
01:56.35Corydon76-homelunaphyte: Now you see why there's a need for multiple contexts
01:56.38asteriskmonkeywell my bars arnt high at all in ztmonitor
01:56.46SwKand by his statement he's tried 'everything else'
01:56.54lunaphyteGir19: ah.  i see.
01:57.12lunaphyteCorydon76-home: thanks for the example.
01:57.18justinuwhat are typical rx/txgain parameters for PRIs from a ILEC CO?
01:57.21justinunot inside a colo facility
01:57.46SwKjustinu: it varies from switch to switch and connection to connection
01:57.59justinuis there a typical range at all?
01:58.08SwKi've set gains everywhere from -12 to +6
01:58.19justinuwhat do you use to tweak it? milliwatt generator?
01:58.28SwKmy years
01:58.30Corydon76-homejustinu: trial and error
01:58.37froguzsomebody know were can i buy an Ambient MD3200 modem?
01:58.37SwKerr ears
01:58.41Corydon76-homejustinu: you adjust each until it sounds right
01:58.42justinutrial and error sounds awfully primitve
01:58.49SwKfroguz ebay
01:58.53asteriskmonkeywhat is the numberic value suppost to be on call im using zt monitor and everything looks fine
01:58.58Corydon76-homejustinu: There really isn't a better way
01:59.00justinuhow do i know what sounds right? compare to a POTS call?
01:59.01asteriskmonkeythe bars dont go to the end
01:59.18Corydon76-homeasteriskmonkey: you adjust it until it "sounds right"
01:59.39justinuhow do you know when it "sounds right"?
01:59.41SwKztmonitor its not a very goot tool for adjusting gains... it just shows you that you are getting something
01:59.49asteriskmonkeythanks ive been doing that for 8 months without success
01:59.51Corydon76-homeSomebody can do all the engineering in the world, but in the end, if it doesn't sound right, it's not done.
01:59.52Gir19justinu: 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.
01:59.54asteriskmonkeyany logical method
01:59.57SwKjustinu: when you can hear it loud enuff but dont get echo
02:00.02froguzSwK, ebay search: 0 items found for
02:00.02froguzambient md3200
02:00.07justinuk, so start low and go up
02:00.13Corydon76-homeasteriskmonkey: Set your gains to -10 each
02:00.34asteriskmonkeygah ok did that a while ago.. will try again
02:00.35Gir19justinu: I normally go in incraments of +/- 4
02:00.37SwKfroguz: then search for x101 or x100 clone since thats what you are really asking for
02:00.42justinuincrements of 4, ok
02:00.43Corydon76-homeasteriskmonkey: if you still have echo problems at that point, your telco needs to run a line test
02:00.57justinuwhat type of line test? BERT?
02:01.18Corydon76-homejustinu: the telco doesn't usually give us that information
02:01.35justinuit's a t1 so, BERT should be appropriate
02:01.43Gir19justinu: I would try more than one type of line test just to get slightly better readings.
02:01.45SwKthe scale on the gains is db... every 3 db you change it you double or half the power level
02:01.53justinuit's /not/ db tho
02:02.12justinuthe scale is percentage of whatever the amplifier algorithm can do
02:02.17justinu-100 to +100
02:02.19lunaphyteif a phone is in more than one context, what dictates which context is applied?
02:02.36SwKi guess i should rfsc closer
02:02.36Corydon76-homelunaphyte: depends upon the channel type
02:02.44asteriskmonkeyok -10 on each and im getting echo
02:02.48asteriskmonkeydamn it
02:02.50lunaphytelet's say a sip channel
02:03.02Corydon76-homelunaphyte: for SIP, it's whichever one finds a match first
02:03.04Gir19lunaphyte: and the channels configuration
02:03.19lunaphyteso - first in the list kind of thing?
02:03.21Corydon76-homelunaphyte: 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?
02:03.31lunaphyteok
02:04.05Gir19lunaphyte: usualy asterisk will go in priority of which context was listed first for that extension then continue down the line.
02:04.17lunaphytecan 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)
02:05.00Corydon76-homelunaphyte: That's a Goto
02:05.05lunaphyteah
02:05.09Corydon76-homelunaphyte: or an include
02:05.14justinuSwK: unfortunatly it's not even clear from the source
02:05.45Corydon76-homeasteriskmonkey: like I said, schedule a line test with the telco
02:06.00themikester60is there a way in asterisk to set a variable from a file on the hard drive?
02:06.04Corydon76-homeFor such a noisy line, they're likely to find something
02:06.32justinuthe only thing I could see affecting echo on a typical HDSL PRI circuit is latency
02:06.45Gir19Corydon76: that depends on the phone company.
02:06.47justinuslips/frame errors typically will cause clicks or pops
02:07.03asteriskmonkeyno issue using sip provider
02:07.09Corydon76-homeGir19: true
02:07.39Corydon76-homeGir19: we have a few CLEC's that we've found to be grossly incompetant
02:07.47Corydon76-home(but that could be just the local office)
02:08.05Corydon76-home*cough*Birch*cough*
02:08.32Corydon76-home*cough*Xpedius*cough*
02:08.43Corydon76-homePardon 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)
02:10.57KizmetDoes 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
02:11.26KizmetAnd
02:11.41_Sam--the only you can do with the missed calls display is turn it on or off
02:11.46KizmetDoes anyone know of how to add a HINT context in AEL ?
02:11.49*** join/#asterisk firestrm (n=no@S01060013105c1c69.gv.shawcable.net)
02:11.51asteriskmonkeyCorydon76-home: didnt know it was possible to get echo on pri :P
02:11.57Kizmet_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
02:12.42Corydon76-homeasteriskmonkey: 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)
02:12.52Corydon76-homeasteriskmonkey: and yet...
02:12.55Kizmet_Sam--, cheers.
02:12.56*** join/#asterisk doushanes (n=dwright@c-71-194-48-241.hsd1.il.comcast.net)
02:12.58asteriskmonkeySo they do ?
02:13.10Corydon76-homeasteriskmonkey: and not just CLECs, either
02:13.12asteriskmonkeylike 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
02:13.26justinui've had PRIs with echo
02:13.30justinucommon problem for me
02:13.32Corydon76-homeThe ILEC has the same issues, but everybody expects that out of them
02:13.36ManxPowerCorydon-w, our CLEC was bought about 6 months ago.  They started to REALLY SUCK about a week ago.
02:13.45justinusounds like broadwing
02:13.54asteriskmonkeyyes im with mci
02:13.58asteriskmonkeyaka worldcom
02:14.07Corydon76-homeaka Verizon
02:14.13outtoluncoh no, aka hell
02:14.15_Sam--aka verizon business
02:14.17ManxPowerTheir new support group don't even know who we are.  We used to be their 4th largest customer.
02:14.19Kizmet_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
02:14.57asteriskmonkeyKizmet: if you want blf and pagine get allworx stuff
02:15.02[av]banio.o
02:15.07ManxPowerKizmet, your extensive search of the mailing list archives and Wiki did not turn anything up?
02:15.28[av]baniSIPAddHeader(Call-Info: answer-after=0)
02:15.30KizmetManxPower, I had a look on VoIP-Info....
02:15.38ManxPower~mailinglist
02:15.38jbotSearch 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
02:15.41ManxPower~docs
02:15.42jbotdocs 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]baniexten => s,1,SIPAddHeader(Call-Info: answer-after=0)
02:17.34[av]baniexten => s,2,Dial(${ARG2},20)
02:17.35[av]banior so
02:18.03[av]banigxp's will add the header if you punch page on them, though that only works for direct phone to phone
02:18.15[av]baniyou can pick it up from the gxp and then set it for outgoing though
02:25.29asteriskmonkeyHere is my zapata.conf
02:25.31asteriskmonkeyhttp://pastebin.ca/42725
02:25.36asteriskmonkeylooks minty dont it
02:25.48asteriskmonkeythink ill crap my foot up mcis ass shortly
02:30.00asteriskmonkeyjust as long as i can say its not my digium cards ill lay into them full tilt :D
02:30.29justinui'm having the issue with a sangoma card
02:30.34outtolunci must have missed the first part
02:30.52asteriskmonkeyaudiocodes gateway for the win
02:31.24outtoluncfirst thing i seen was
02:31.27outtolunc[18:11] <asteriskmonkey> Corydon76-home: didnt know it was possible to get echo on pri :P
02:31.48asteriskmonkeywell am i right on that? or is that out to lunch
02:31.59outtolunctoo which i replied something to the effect.. 'wonders why telco's have the term 'echo can's
02:32.06justinuthe pri doesn't have anything to do with the echo, it's the endpoints
02:32.15justinuand how much energy you put into the network
02:32.20asteriskmonkeyok
02:32.31asteriskmonkeyso i cant cram it up there ass...
02:32.35mzoWORK dammit
02:32.48justinuthe EC built into some PRI cards helps alleviate echo
02:32.57asteriskmonkeywell i was at -30 and i could hear stuff nor could anyone else
02:32.57justinuyeah, there's a tail and longhaul side
02:33.08justinutail is your pbx to phone loop
02:33.14*** part/#asterisk doushanes (n=dwright@c-71-194-48-241.hsd1.il.comcast.net)
02:33.15justinulong haul is pbx to co loop
02:33.25asteriskmonkeyso its the tail loop then maybe
02:33.26Corydon76-homeouttolunc: actually, I'm aware of echo on PRIs
02:33.35asteriskmonkeyreally good
02:33.40Corydon76-homeouttolunc: I just find it a much bigger problem on analog lines
02:33.48outtoluncusually 'long haul' is used to talk about 'end points that are 500+ miles apart'
02:33.53asteriskmonkeyi though i was goign insance i spent uber loot on a te406 to have 60ms of echo can on each channel
02:33.57outtoluncnot sure what you are talkin about
02:34.12justinu60ms isn't all that much
02:34.26justinusometimes 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)
02:34.48asteriskmonkeyso what change my echo can to 256 for the taps?
02:34.55*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
02:35.31justinuno idea
02:36.03outtoluncCorydon76-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)
02:36.10asteriskmonkeylocal numbers it tends not to happen on anytime a cell or ld (as in 1 city away) calls happens its very apperent
02:36.25lunaphytewhat does s@default mean in "IAX2/guest@misery.digium.com/s@default"
02:36.25justinucell is notorious for increasing latency
02:36.44*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
02:36.46Corydon76-homeouttolunc: that's okay, I'm sure my sarcasm is about to get flamed on the -dev list.
02:36.58outtoluncmost people with pri's never have an issue, but there are some that have 'all the issues'
02:37.18outtoluncut oh <G>
02:37.20justinui've had enough PRIs
02:37.24justinuand i've always had these issues
02:37.41outtoluncright, and you are in the same place you always been
02:38.10*** join/#asterisk |omni| (i=rob@net98.limelyte.net)
02:38.16asteriskmonkeylol
02:38.22asteriskmonkeylocation location location
02:38.38outtoluncrocket science 001
02:38.40KizmetI work for a company that has 4 sets of pri's
02:38.56*** join/#asterisk samueltc (n=sam@Toronto-HSE-ppp3880832.sympatico.ca)
02:38.57outtoluncmove away from the BLAST
02:39.03Kizmetit just happened that the 2 sets that we allocated for internal support lines have shit loads of echo :(
02:39.19samueltcis iax2-trunking is consuming less bandwidth than SIP?
02:39.25Kizmetyet the customer lines have none....
02:39.54asteriskmonkeydepends on the codec
02:39.57outtoluncinternal?
02:40.06Kizmetas in
02:40.08outtoluncwhat cable lengths
02:40.12outtoluncetc etc etc
02:40.16KizmetCall XXX XXX XXX for Support.
02:40.24[hC]anyone here use the open79xx xml directory for cisco phones?
02:40.36samueltcasteriskmonkey: using the same codec indeed..
02:40.41samueltcg729
02:40.41KizmetWe are localted in the BACK of the data centre right near the distrib board.
02:40.59asteriskmonkeythen iax2 has less overhead than sip
02:41.00asteriskmonkeyso there :D
02:41.03samueltcnice
02:41.04justinuactually, i've had PRIs from a bunch of different telcos
02:41.08justinuon a bunch of facilities
02:41.11justinuHDSL, DS3
02:41.14justinufiber
02:41.29samueltcjustinu: any in europe? hehe
02:41.31justinui've got 2 DS3s of PRI
02:41.33justinunope
02:41.41justinubut they have always been in the city of LA
02:41.47justinuor suburbs of
02:41.50mzoanyone 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
02:42.01Kizmetouttolunc, 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
02:42.16outtolunc[18:41] <justinu> actually, i've had PRIs from a bunch of different telcos
02:42.21mzopics plz
02:42.24*** join/#asterisk websae (i=icechat5@CPE-24-167-204-30.wi.res.rr.com)
02:42.27mzohot picz of sexy rack
02:42.29outtolunc[18:37] <outtolunc> right, and you are in the same place you always been
02:42.42justinuwell, sorta
02:42.48katakefaloshello 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
02:42.48katakefalosexten => 1+_NXXXXXXXXX,2,Wait(1)
02:42.48katakefalosexten => 1+_NXXXXXXXXX,3,DeadAGI(a2billing.php|1)
02:42.48katakefalosexten => 1+_NXXXXXXXXX,4,Hangup
02:42.51Kizmetmzo, no pics working on it tho :)
02:42.54outtolunceither explain yourself better or stop while you are behind
02:43.10justinunot worth it to me :P
02:43.13outtolunck
02:43.25mzoi will trade hot rack pics (sexy ciscos!) for help with fwd!
02:43.44lunaphyteany opinions on ipkall.com ?
02:44.29Kizmetlunaphyte, There incoming calls seem to be of poor quality sometimes and my number got deleted after 2 months (It was under relitivly high use)
02:45.00mzoi need to find something to call finland cheap ;)
02:45.09lunaphyteis it true that you can only talk to them through fwd or similar?
02:45.24Kizmetmzo, a VoIP prov in Finland perhapse :P
02:45.43Kizmetlunaphyte, Nope. You may forward the call to any SIP capable host.
02:45.55websaefirestrm: are you there?
02:46.29lunaphyteah - so i f i configure asterisk to accept a sip connection from them, i can point them towards my server when i sign up?
02:46.48Kizmetlunaphyte, yep
02:47.05mzoKizmet, im trying to find one, to talk to there
02:47.34Kizmetmzo, Yes, Get one from there and call inside the country :) Mostly the cheapest solution :)
02:47.50mzothat's what im looking for, i can't find one iwth good english translation :)
02:48.16Kizmetmzo, heh
02:49.00*** join/#asterisk iaxy (n=iaxy@modemcable236.55-131-66.mc.videotron.ca)
02:49.37iaxyhi everyone
02:49.41outtolunci'm still wondering why kizmet wasn't including the distance to the interchange
02:49.53iaxyHas anyone succesfully used Citels handset gateway?
02:49.53outtolunc(50'-100')
02:50.02Kizmetouttolunc, dist to the interchange im unsure of.
02:50.49outtolunconce again...
02:50.50Kizmetouttolunc, 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)
02:51.16outtolunci'm just a guy that thinks a dialplan is a script
02:51.22outtoluncsilly me
02:51.24jeebusroxorsHey - Is IAX recomended over SIP when registering with a PSTN termination?
02:51.54xachenerm
02:51.57xachenI'd take SIP anyday
02:52.09Kizmetouttolunc, Heh all i do with the dial plan is change the way certain dest's route and update billing stuff.
02:52.18jeebusroxorsxachen; i was reading that iax is better on bandwith and what not
02:52.23xachenyes
02:52.26xachenbut IAX is also more insecure
02:52.28Qwell355
02:52.28outtolunci think i get that, really <G>
02:52.31xachenand can be real buggy
02:52.35jeebusroxorshm
02:52.37Telamonjeebusroxors: 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.
02:52.50xachenI'm just a SIP nerd though
02:53.19jeebusroxorsoh, i defanetly dig SIP, im just concered with bandwith issues as im using my own machine as a server
02:53.32iaxyanyone use citels sip handset gateway, to connect nortel digital phones to asterisk?
02:54.08asteriskmonkeyiaxy : yes
02:54.23asteriskmonkeyhave to setup 4 of them tommorow :D
02:54.58russellbxachen: IAX is more insecure?
02:55.00iaxyAH... I haven't got mine to send a single sip ack....
02:55.01russellbdid you make that up?
02:55.13xachenactually no :)
02:55.18TelamonAnyone 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?
02:55.20xachenmay I reprhase
02:55.31russellbof course :)
02:55.33outtolunctis the season
02:55.37jeebusroxorsheh
02:55.43xachenIAX is insecure but so is SIP
02:55.47xachenI'd just take SIP over IAX
02:55.48iaxyasteriskmonkey; can I msg you
02:55.58xachenthe 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)
02:56.28tainted_how many 729 channels could a DSL line hold?
02:56.32tainted_assuming 128k up
02:56.46outtoluncand no ids that looks for 'bruteforce' you agree with?
02:56.48justinutainted....
02:56.57tainted_yea
02:57.02outtoluncmeaning.. no nadda
02:57.04justinuprobably about 7
02:57.05xachenwell you can give no ID and a password and the server will try to find the username for you
02:57.14outtoluncoh yeah bang the shit out of me.. go for it
02:57.15Telamontainted_: Technically, 14.  In reality, about 10.
02:57.16justinuit's pretty light
02:57.16xachenbut thats easily fixd by adding a [guest] entry
02:57.35tainted_wow that's a lot more than i expected
02:57.48iaxyasteriskmonkey;Do you have any docs on using citel GW with Asterisk?
02:58.06outtoluncsorry :(
02:58.09Telamontainted_: g729 is about 8kBytes/sec per call. + UDP/IP overhead.
02:58.34outtoluncjust 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)
03:00.04websaefirestrm are you out there?
03:00.48[hC]urg.
03:00.49[hC]anyone here use the open79xx xml directory for cisco phones?
03:03.46outtoluncso, 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)
03:06.13outtoluncjoy, i just caused world peace! <G>
03:06.20outtolunchah
03:08.05asteriskmonkeyiaxy: did you buy them from williams?
03:08.25*** join/#asterisk xbmodder_lappy (i=nobody@unaffiliated/xbmodder)
03:08.39xbmodder_lappyWho is the most _reliable_ VoIP provider out therE?
03:09.10tuxinator_linuxxbmodder_lappy: when you find out, let us know
03:09.44glm2ktuxinator_linux: rotfl
03:09.46xbmodder_lappylol
03:09.52outtolunclappy, 'most' of us know that when it comes to 'reliable' it depends on the 'backhoe' factor
03:09.53asteriskmonkeyxbmodder_lappy: that would be me :) but i got echo issues lol
03:10.05glm2kouttolunc: aye, i second that
03:10.59*** join/#asterisk wrench (n=signal@68-118-224-178.dhcp.oxfr.ma.charter.com)
03:11.51wrenchhey all, whats a good incoming-only voip provider for small businesses - trying to set up a custom IVR solution
03:12.05outtoluncseriously, if you plot line issues to power issues, the power issues take the cake
03:12.33outtoluncso line issues boil down to 'dude with backhoe'
03:12.48evilbunywrench: ipkall.com offer free inbound numbers in the US
03:13.10voip470"backhoe factor"?
03:13.24outtoluncprefect example
03:13.27outtoluncer per
03:13.43wrenchevilbuny > for commercial use?
03:13.43outtolunc'reset please?'
03:14.54evilbunywrench: they make money per minute on your call
03:15.04evilbunyfraction of a cent per minute
03:16.51outtolunc... punt
03:17.16asteriskmonkeywhat the hell
03:17.16asteriskmonkeyi called a number and its not in my zap show channels
03:17.33asteriskmonkeyi got the mci test tone number
03:17.35outtolunccalled = passed tense
03:17.41asteriskmonkeyno im connecte
03:17.46asteriskmonkeyconnected currently
03:17.51asteriskmonkeyits a test tone number
03:17.55outtoluncthen say what is really happening
03:18.07outtolunci'm not a f'n mind reader
03:18.11asteriskmonkeysos
03:18.13wrenchhmm
03:18.50outtoluncwhich 'type' of channel is it
03:18.55outtoluncif it is '
03:19.00outtoluncof a type'
03:19.10tengulreHi,all! I m backing....! good morning every one!
03:19.13outtoluncthen did you do '<the type> show channels'
03:19.24outtoluncor is that 'out there'
03:19.54outtoluncsorry, i didn't mean to swear with 'f'n'
03:20.12asteriskmonkeyzap show channels
03:20.15asteriskmonkeyshows nothing
03:20.23outtoluncis it a zap channel
03:20.33asteriskmonkeyits goign out a zap channel
03:20.42asteriskmonkeysip=>asterisk=>zap/pri
03:20.48outtoluncand we would know this HOW
03:21.20outtoluncand you say 'it IS going out'
03:21.34outtoluncwell if it is, then there isn't an issue
03:21.47asteriskmonkeyexcept i cant see what channel its on
03:21.53asteriskmonkeygah
03:22.05asteriskmonkeyim getting sleepy and not explaining myself well anymore
03:22.05outtoluncdid you mean you say that 'you attempted xyz and it failed with THIS error message"
03:22.17asteriskmonkeyno
03:22.31asteriskmonkeyi mean its working currently connected listening to a tone over a zap channel
03:22.40asteriskmonkeyjust show zap channels is not showing me what line its using
03:22.54outtoluncso, you attempted "something", and there is no error message that you want to share with us
03:23.10outtoluncok, ummm goodluck
03:23.55outtoluncit boiled down too
03:23.57outtolunc[19:17] <asteriskmonkey> no im connecte
03:23.57outtolunc[19:17] <asteriskmonkey> connected currently
03:23.58outtolunc[19:17] <asteriskmonkey> its a test tone number
03:24.05asteriskmonkeysos..
03:24.09lunaphyteso 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?
03:24.12outtoluncthat is NOT any form of request
03:24.18asteriskmonkeyi called a number using my sip phone which traveled out a zap channel
03:24.37asteriskmonkeythe phone call is currently in progress and a zap show channels did not render which channel it was using
03:24.42asteriskmonkeythere that makes more sense
03:24.57outtoluncyes it does
03:25.19outtoluncbut it also would make me ask why you didn't do a sip show channels
03:25.24outtolunchmmm
03:25.36asteriskmonkeysip show channels shows stuff but not the zap channel
03:25.51*** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.138)
03:25.54Kernel_corehi all
03:25.54outtoluncobviously
03:26.03asteriskmonkeyi need to know that zap channel to use ztmonitor !
03:26.04outtoluncsip, zap, sip, zap
03:26.08outtolunchmm
03:26.15outtolunchello
03:26.21asteriskmonkeydoh i still get * in my ztmonitor .. i guess that means i have to lower it still
03:26.31Kernel_coreanybody successfully compiled H323 on Asterisk ?!
03:26.32outtolunchow about a 'show channels'
03:26.37outtolunchmmmmmmm?
03:27.27outtoluncobviously he never thought of that one
03:27.51*** join/#asterisk bjohnson (n=bjohnson@i216-58-92-216.cybersurf.com)
03:28.31asteriskmonkeydoh!
03:28.32asteriskmonkey:P
03:28.36outtolunck
03:28.47outtoluncyour welcome
03:29.20asteriskmonkeythanks
03:31.30Goralis there a working asterisk module for webmin?
03:33.24outtoluncgoogle 'asterisk pbx webmin'
03:34.10*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
03:34.14shmaltzhi every1
03:34.36Goralthe 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..
03:35.12outtoluncah
03:35.33*** join/#asterisk chetan (i=freetibe@cpe-24-193-188-21.nyc.res.rr.com)
03:35.37outtoluncwell since i've never used it i'd be one the 'undecided'
03:35.56*** join/#asterisk camonz (n=camonz@200.8.20.210)
03:35.59camonzevening :->
03:36.10outtoluncwhich leads to the 'you are in a group of 'those that need/want' it'
03:36.28outtoluncwhich mind you, is a 'subset'
03:36.36shmaltzGoral, what works or doesn't?
03:37.11Goralwebmin module for asterisk
03:37.25outtolunci'm sure it 'did' at one rev, but may/maynot now... he just doesn't get that
03:37.44shmaltzGoral, I looked at it a while back, from all the ones around, I think 3rd lane has done the best job so far
03:39.32shmaltzhttp://news.yahoo.com/s/pcworld/20060221/tc_pcworld/124781;_ylt=AloXUQMSFUN4Ygm4SqoGyx_67rEF;_ylu=X3oDMTBjMHVqMTQ4BHNlYwN5bnN1YmNhdA--
03:39.44Goralty 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)
03:40.10shmaltzGoral, what are you trying to accomplish?
03:40.22*** join/#asterisk L|NUX (n=linux@202.5.145.56)
03:40.29Goralwell..
03:41.26shmaltzouttolunc, back to the ocean
03:42.00outtoluncoh, so someone that knew there 'were from the ocean' didn't head TOO the ocean
03:42.23Gorali want to put a box togethere that will allow me to use a sip subscription throught my house
03:43.03shmaltzGoral, then you should try Asterisk, it's that wonderfull program that can do that for you ;)
03:43.14Goraleg. 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..
03:43.17outtolunche knows this tho
03:43.25outtoluncbut you stepped up
03:43.32outtoluncyou need to make sure his 'home'
03:43.48outtolunc<PROTECTED>
03:44.42Goralguess its like splitting my sip account all over the house
03:44.56outtoluncumm no, it isn't <G>
03:45.23outtoluncdo you pay more than once
03:45.39Goralwhat on my sip account?
03:45.46outtoluncthats a no
03:46.11outtoluncso it 'IS ONE ACCOUNT'
03:46.17Gorallike i said i'm sort of new
03:46.19Goralyes
03:46.24Goralthis is one account
03:46.29outtoluncthen why the bs?
03:46.56Kizmetouttolunc, I have a IAX2 Trunk into my ITSP :) I have a so called 'Unlimited Local-Loop' connection into them :)
03:47.14outtolunci think they are high (or you are)
03:47.25outtolunctake your pick
03:47.26Kizmetouttolunc, lol
03:47.29*** join/#asterisk websae (i=icechat5@CPE-24-167-204-30.wi.res.rr.com)
03:47.53outtoluncso, your question 'would have been' what?
03:47.57Kizmetouttolunc, I work for the ITSP :)
03:48.27Goralwell 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
03:48.36KizmetGoral, There is no reason why it wont work.
03:48.44outtolunci am just wondering what the hell type of person asks an 'open ended, not finished, question' kinda sorta
03:49.12outtoluncbut hey, shit happens
03:49.23outtolunckinda, sorta
03:49.50Goralnow i have to learn how to set it all up
03:49.56JCC_'cause he's a little new at this
03:50.07outtoluncor did i miss some cool ass question that would have just made my day
03:50.34outtoluncjcc, which is?
03:50.47outtoluncor do i need to get on you also
03:51.16Goralouttolunc ty for your perspective its people like you that make me think
03:51.33Qwellhmm
03:51.40Qwellmy sarcasm detector just exploded
03:51.53outtoluncgoral, i am the type that 'will always help you' i don't ask much
03:52.12outtoluncbut if you ask something silly, you get the same
03:52.19outtolunchello
03:52.19Gorali know but i was serious that was far from sarcasm
03:52.20websaefirestrm: you out there?
03:52.27JCC_not sure what you're talking about...
03:53.17outtoluncJCC: it's ok, it's smooth, gooooo back to sleeeeeeep
03:53.20JCC_I'm under the impression he's starting in the wrong place, maybe voip-info.org is his best bet
03:53.39outtoluncgee
03:53.57outtoluncthat starting point never crossed ANY of our minds
03:54.06outtoluncya think
03:54.19JCC_it did'nt cross his though, apparently.
03:54.27outtoluncgee
03:54.30GoralJCC_ i always jump in head first... no better way that to deal with people that know it the iner workings
03:54.35outtolunche gets it, i think
03:54.54JCC_he does now
03:54.57outtoluncnow when we place him in the same category, does he
03:55.29outtoluncyou were just under the wire, i retract that last
03:55.33Goralhell i'm a noob give me a little slack
03:55.44Goralmy linux is rusty too
03:55.55outtoluncgoral, i've done nothing but help
03:56.04outtolunci've not even been mean to you
03:56.32JCC_check voip-info.org and you'll get 99% of your questions answered there, you'll need to do some reading, lots of it.
03:57.11JCC_and get a decent book on linux, too, then.
03:57.30Goralouttolunc i know and i'm thankful
03:57.45QwellQwell's - Teach Yourself Linux in 24 Weeks, is a good Linux boox
03:57.46Qwellbook
03:58.20JCC_couldn't tell you,I haven't read one in 6 or 7 years
03:58.34QwellI'm joking, of course
03:58.54JCC_:)
03:59.03JCC_I figured
03:59.06Gorali forget how simple it is.. micro$oft is so complex... as in its wording..
03:59.32Goralwell looks like i need a book
04:00.04JCC_OReilly's asterisk book is decent
04:01.07JCC_but voip-info is probably your best starting point, not here.
04:01.33JCC_right, outtolunc?
04:01.40JCC_:)
04:03.31*** join/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net)
04:04.44websaeif 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!
04:05.02russellbPlease don't use this channel for commercial purposes.
04:05.14websaemy apologies
04:05.21websaei dono't mean to advertise
04:05.46websaejust saying dedicated staff out there that keep up a good network for asterisk termination if anyone needs it
04:05.59Mavviehas anybody here ever tried to contact support@digium.com ?
04:06.11websaeyeah, they are pretty good about getting back to you
04:06.16Mavvieoh
04:06.20Mavviefunny :-)
04:06.24websaeat least from my experience
04:06.38Mavviedid you call them or via email?
04:06.40websaei had to email them once or twice with a couple hardware issues
04:06.41trixterI get away with talking about my termination services cause they are free :P
04:06.46trixterdont even have to have an account
04:06.52websaethat's awesome
04:07.00websaewhat type of termination do you have?
04:07.05websaeu.s.--?
04:08.27trixterus and canada tollfree
04:08.42websaedo you monintor calls?
04:08.48trixterhttp://www.trxtel.com/index.php?page=Tollfree_Termination
04:09.03trixterno that would be illegal to monitor calls without informing people, it is also a violation of the privacy policy
04:09.26trixterwe are trying to work a deal for inbound dids free as well
04:09.27websaewhy is the service free?
04:09.41trixterto ensure that many people make calls
04:09.51outtoluncthe service is not free, send the check too....
04:10.30mzotrixter, is t that your service?
04:10.41*** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net)
04:10.43trixteryes
04:11.07russellbwe used to do free tollfree on iaxtel ... one of these days I'm going to fix that server :)
04:11.12outtoluncah
04:11.21trixterheh
04:11.28outtoluncscroll back heaven...
04:11.31*** join/#asterisk rustyb (n=rustyb@68-235-135-252.atlsfl.adelphia.net)
04:11.36trixterwell if people push enough minutes I pay them, totally backwards of the way phone service normally works
04:12.20mzotrixter, 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.
04:13.00russellbyes, you need to add the rest of the GotoIf
04:13.11russellbthat tells it to skip it if the result is 0
04:13.26russellbthe 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...
04:13.53russellbyup
04:14.43*** join/#asterisk Jizzbug (n=derekm@63-254-64-44.ip.mcleodusa.net)
04:17.07trixtermzo: yes it is
04:17.30outtoluncyeah, whatever he said.. just YEAH
04:17.42outtoluncsomething
04:18.11outtoluncotherwise, this channel gets fairly boring
04:18.44outtoluncor doesn't that matter here?\
04:18.47Mavvie-- Launched AGI Script /var/lib/asterisk/agi-bin/check-callerid.pl
04:18.50mzotrixter, i can't takl atm , but i had to ask you questsion about setting up for it.
04:18.52Mavviels: /var/lib/asterisk/agi-bin/check-callerid.pl: No such file or directory
04:18.55Mavviewho do I have to believe?
04:19.19outtoluncso 'check-callerid'
04:19.33outtoluncis the topic
04:19.53outtolunci believe shit happens for a reason
04:20.06russellbiaxtel is now running the latest code from the 1.2 branch ... maybe that will make it happy
04:20.13Mavvieit's more that it doesn't throw an error.
04:21.23mzotrixter, 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?
04:21.26outtoluncumm if you can't see that as an error, you just might have issues <G>
04:21.56Mavvieouttolunc: it wasn't asterisk which did do that ls output
04:22.10outtoluncregardless
04:22.23outtoluncthere is a spoon (err error)
04:22.35outtoluncsomewhere someplace
04:22.41Mavviecould 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)
04:22.48trixterthere is no registration
04:22.58mzohow do i set it up then :P
04:23.04trixtercut and paste
04:23.09outtoluncbut all we get is 'life is normal/usual' i have no clue that anything is wrong
04:23.17trixteryou just dial no accounts, no registration, no tracking nothing
04:23.19outtoluncetc etc etc
04:23.26mzooh, hmm, it bombed, i think i did it wrong then ;)
04:23.56trixterin 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
04:24.14outtoluncgood, then we can add you to the 'he agrees that shit happens party'
04:25.02outtoluncpersonally, 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)
04:26.06outtoluncobviously not that much
04:26.47rustybrussellb Woot! I'm registered w/ IAXTEL again
04:26.55russellbnice!
04:27.05outtoluncgee wally, thats special
04:27.09rustybits been like a year
04:27.55outtolunci once again see why it's fairly hard to take anyone serious here
04:28.17rustybrussellB are you aware of any zap <=> zap 1 way audio problems w/ ver 1.2.4 ?
04:28.29russellbno, i have not heard anything
04:28.52mzowhat is iaxtel?
04:29.00outtoluncgee why would there be any zap to zap audio issues
04:29.33niZonhas anyone here ordered from voxilla?
04:29.50rustybi dont know sip <=> zap and iax <=> zap are ok.
04:30.34rustybsoftware or driver/ hardware issues likely.. i guess it's just my install
04:31.04niZonhmm their canadian toll free is down
04:31.38outtoluncstrange, i'm sure there were some slinear issues that floated into my email
04:32.01outtoluncbut gee how can they be the same
04:32.30Mavvieexecv(script, argv);
04:32.30Mavviefprintf(stdout, "verbose \"Failed to execute '%s': %s\" 2\n", script, strerror(errno));
04:32.43Mavviethat explains why I see it on HEAD and not on the other one.
04:33.02outtoluncthe reason for the double "\" crap is?
04:33.03Mavvielet's see how stat works again
04:33.42mzotrixter, 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)
04:34.35Mavviebecause it's supposed to be one argument.
04:36.00trixterum ...  its like anything else in the context you are in
04:36.05russellbrustyb: do you have iaxtel set up where you could make a test call?
04:36.06trixterit quite literally is cut and paste
04:36.26outtoluncgee
04:36.29Kizmettrixter, wasnt here cuz im using AEL :) but eyh it sould be cut n paste
04:36.30mzolemme go poke around, i'm using the amp, so i probably screwed something up :P
04:36.32outtoluncbut and paste
04:36.35outtoluncer cut
04:36.43trixterok I will add ael stuff tonight :P
04:36.43outtolunchmm
04:36.48mzowhat's AEL?
04:37.01trixtera different dialplan language for asterisk
04:37.08Kizmetmzo, nothing you need to worry about at this stage young padawan.
04:37.22xbmodder_lappymzo, You are a young padawan
04:37.23mzoi have a long time until i reach padawan status :P
04:37.33mzoI'm still trying to get xp to level up to noob.
04:37.41Kizmetlol
04:37.43outtolunci'm still trying to see where the NON-padawan
04:37.49outtolunc's get into this
04:37.57xbmodder_lappyWe do asterisk consulting, would you like to be taken under our wing. We can teach you.
04:38.07Kizmetouttolunc, no idea *sigh*
04:38.08mzothey get an invite to the sekret invite #jedi-asterisk
04:38.12outtoluncreally?
04:38.14mzoxbmodder_lappy, teach me, my master. :P
04:38.17outtoluncgee
04:38.27outtoluncwhere do i sign up?
04:38.33xbmodder_lappyRemember I am only here to show you the door, you're the one who has to walk through it
04:38.34russellbi know someone wants to make an iaxtel test call :)
04:38.47mzorussellb, show me how to configure it :P
04:39.01russellbmzo: ha ... iaxtel.com
04:39.13*** join/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net)
04:39.18Mavvieheh... C should have an -x operator :-)
04:39.37outtolunccome on
04:39.42outtoluncthere has to be more than this
04:40.18mzotake the red pill, the one labelled 'iax'
04:40.23Mavvieor can I safely assume that asterisk is always ran as root?
04:41.08outtoluncumm NO
04:41.27outtoluncsearch the wiki for 'non-root'
04:41.40*** join/#asterisk riddlebox (n=victoria@24-171-11-166.dhcp.stls.mo.charter.com)
04:41.44outtoluncwhat the F are you teaching these guys
04:41.58xbmodder_lappylol
04:42.21outtoluncwhat do i care
04:42.24outtoluncgo for it
04:42.33outtolunchave a wonderful DAY
04:42.41outtoluncparty on
04:42.51Mavvieoh, access() is a good way to overcome this problem.
04:43.10outtoluncshoot mavvie in the head
04:43.18outtoluncfor bs f'n crap
04:43.28Mavvieif that's the best you can come up, keep practising.
04:43.54outtoluncgee wally only another 30 years i might be able to deal with bobo
04:44.05outtoluncplease
04:44.10outtoluncstep up
04:44.14outtolunctake a shot
04:44.21outtoluncif you actually have one
04:44.34outtolunchmmm
04:44.38outtolunci'm waiting
04:44.42outtoluncstill waiting
04:44.46outtoluncyet still
04:44.58outtoluncbit slow aren't ya
04:45.27outtoluncyet still, ....
04:45.36*** join/#asterisk DaGeek215 (n=DaGeek21@c-71-226-252-76.hsd1.pa.comcast.net)
04:46.51outtoluncstrange 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
04:47.30outtoluncanyone want to debate SIT tones durations?
04:47.56outtoluncaww
04:48.29outtolunci 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)
04:49.06mzoi'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)
04:52.56Mavviehttp://bugs.digium.com/view.php?id=6565 <- fixed!
04:53.13mzorussellb, are you the admin for that site? Send me my password :P
04:53.46brookshiremzo: which site?
04:54.26russellbmzo: ha, what username did you use
04:54.31russellbmzo: iaxtel
04:54.33russellberr
04:54.35russellbbrookshire:
04:54.48brookshirehehe..
04:54.54mzoyes iaxy
04:54.56mzoer, iaxtel
04:54.59brookshiresomeone seriously needs to fix iaxtel
04:55.13russellbmzo: what's your username?
04:55.19mzomzo :P
04:55.20brookshirerussellb: fix it now!
04:55.22mzothat's what i signed up as
04:55.32russellbit's not in the db :(
04:55.40mzobleh lemme check
04:55.42russellbor the iaxtel db anyway
04:55.49russellbthat might happen after you confirm it
04:55.55russellband i have no idea how that part works
04:55.58mzoi didn't get the confirm email =(
04:56.03russellbyeahhhhh ...
04:56.07russellbi have no clue.
04:56.09russellbbrookshire: !!!!!!!!
04:56.10russellbFIX IT
04:56.11russellbNOW
04:56.20*** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net)
04:56.26brookshiremaybe we could outsource it ;)
04:56.29russellblol
04:56.41brookshirelol.. make it a topic
04:56.48russellbi'll fix it ...
04:56.50russellb... later
04:56.52brookshire/msg brookshire if you want to fix iaxtel
04:56.54[hC]anyone here using the open79xxdir xml directory?
04:58.16outtolunc.. /or outsource everything and oh... whatever
04:58.24outtoluncgee
04:58.32mzo.
04:58.43mzoplz fix, i want to test :)
04:59.40outtoluncfix what? i've not seen a damn thing that looks like anything close to helpful
05:00.19outtolunc'this looks wrong'
05:00.33outtoluncok, yeah, when you say it that way
05:00.43outtolunc'this looks wrong' <G>
05:01.12outtoluncanyone getting this?
05:01.27outtoluncor will the next person do the same damn thing
05:03.05outtolunci'm sorry, but if you can't define it, i surely can't trace it down
05:03.25outtolunc)tjats
05:03.33outtoluncekk sorry
05:04.02*** join/#asterisk redondos (i=redondos@12-207-132-99.client.mchsi.com)
05:04.32outtoluncbut please, anytime you want to state the 'issue' and the 'version' and the 'circumstances' well who am i to be the DICK
05:04.55outtoluncgod forbid
05:05.07outtoluncanyone?
05:07.42*** join/#asterisk jyukes (n=jameshot@pool-71-244-78-223.atc.east.verizon.net)
05:07.47outtoluncso you would rather i said: plz fix, i want to test :)
05:08.02outtolunchmmm
05:09.26*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
05:11.11katakefalosdoes 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
05:11.11katakefalosexten => _NXXXXXXXXX,2,Wait(1)
05:11.11katakefalosexten => _NXXXXXXXXX,3,Prefix,1
05:11.12katakefalosexten => _1NXXXXXXXXX,4,DeadAGI(a2billing.php|1)
05:11.12katakefalosexten => _1NXXXXXXXXX,5,Hangup
05:11.19katakefalosbut it does not work
05:12.41evilbunyexten => _NXXNXXXXXX,1,Goto(1${EXTEN},1)
05:12.48evilbunyexten => _1NXXNXXXXXX,1,....
05:12.55*** join/#asterisk tengulre (n=tengulre@221.11.5.180)
05:13.08MavviePrefix(1) maybe?
05:13.11outtoluncfirst off
05:13.24outtoluncyou need to keep 'this and that' together
05:13.50evilbunybut i might want my thises and thats seperated in the wash :)
05:13.53outtoluncmeaning..  _NXX = this, and _1NXX = that
05:13.57Mavviekatakefalos: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Prefix
05:14.07Mavvie"If you switch into an extension which has no first step, Asterisk will treat it as though the user dialed an invalid extension."
05:14.32outtoluncso, JUMPING from this,3 to that,4 is not right
05:14.34Mavviepriority 4 is not a first step.
05:14.58outtoluncand if you can't see that, clean your glasses
05:15.39KizmetOk 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.
05:16.09outtolunceasy
05:16.20outtolunclisten closely
05:16.37KizmetOk NP
05:16.47outtolunc'asterisk defaults to the [default] context.. hint f'n hint
05:17.22mzobrb i gotta reboot =(
05:17.26outtoluncmeaning, anyone silly enough to use the [default] context as 'normal procedure is HIGH
05:17.27outtolunc'
05:17.37outtoluncsorry<G>
05:17.57outtoluncOUT OF THEIR EVERLOVIN F"N MINDS
05:18.06outtoluncok?
05:18.32outtoluncis that is "ANY WAY" unclear?
05:19.04rustybrussellb yes i can make an iaxtel test call
05:19.08outtolunci personally HOPE not
05:19.09evilbunyouttolunc: sure it is, to people that don't do english :)
05:19.10rustybwhats your number?
05:19.15katakefalosoutololunc> am new to this and try to unterstand
05:19.34Kizmetouttolunc,  SIP/2.0 404 Not Found
05:19.35russellbrustyb: well, i ended up just calling myself :)
05:19.41outtoluncDO NOT EVER USE [DEFAULT} FOR ANY REASON
05:19.48outtoluncer }
05:19.54outtoluncer whatever
05:20.04rustybmzo hello
05:20.23rustybok i did the same
05:20.23*** join/#asterisk mzo (i=user@ool-435193b3.dyn.optonline.net)
05:20.37rustybone number workes & the other doesn't
05:20.38outtoluncso when they ask you 'knowledgeable people that sit here daily' in the future, you will kindly pass that on, RIGHT
05:21.04outtoluncor will you be the non-responding asses you have shown yourselves to be
05:21.11outtoluncwhich is it?
05:21.24evilbunycan we be fence sitters instead? )
05:21.25evilbuny:)
05:21.38outtoluncif you are, do so
05:21.42outtoluncand STFU
05:21.45evilbunylol
05:21.58Nivexouttolunc: referring to the people who have the potential to help you as "asses" does not strenghten your posititon.
05:22.13outtolunclike i give a flying fuck
05:22.22outtoluncwhen have ANY of you helped me
05:22.23outtoluncever
05:22.24Nivexclearly, you don't.
05:22.32outtoluncever
05:22.40NivexPlease take your harsh language elsewhere.
05:22.42evilbunyouttolunc: what did you need help with?
05:22.43outtolunci didn't studder did i
05:22.45outtoluncever
05:22.48russellbouttolunc: have you filed bug reports?
05:22.52outtolunchaha
05:22.53russellbif you did, I'm sure I've worked on them in some way.
05:22.57russellbso there.
05:23.23Kizmetouttolunc, 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
05:23.24Kizmet<PROTECTED>
05:23.27outtoluncplease if you think i'm without bugs, do a search
05:23.35outtoluncPLEASE
05:23.42russellbi know you have
05:23.51outtoluncyet you still pulled that crap
05:23.55outtoluncwhy?
05:24.05outtoluncmore f'n BS
05:24.06Mavvieouttolunc: they have stuff against bugs at chemists.
05:24.10russellbbecause you said "when have ANY of you helped me"
05:24.14Nivexouttolunc: switch to decaf man.
05:24.17rustybrussellb the 2nd iaxtel no halts my *
05:24.20outtoluncwhen have YOU
05:24.28outtoluncrusselb helped me?
05:24.33outtoluncplease
05:24.41outtoluncwhen?
05:24.58outtoluncmore f
05:25.01outtolunc'n bs
05:25.16russellbdude, chill out :)
05:25.21outtoluncwhy
05:25.27outtoluncyou said you helped me
05:25.31outtoluncso say so
05:25.33Mavviemostly because you're making a fool out of yourself.
05:25.39outtoluncexcuse me
05:25.46mzorussellb, please fix iaxtel :P
05:25.46Nivexouttolunc: You are excused.  You may leave.
05:25.59xtrvdKizmet: exten => i,1,Goto(ael-error,s,1)      ?
05:26.02mzoooh i'm missing drama!
05:26.35Kizmetxtrvd, as i have said i have tried using 'i' and it doesnt work.
05:26.53katakefalosthank you evilbunny! i did exten => _NXXXXXXXXX,1,Goto(1${EXTEN},1)
05:26.53katakefalosexten => _1NXXXXXXXXX,1,Answer
05:26.53katakefalosexten => _1NXXXXXXXXX,2,Wait(1)
05:26.53katakefalosexten => _1NXXXXXXXXX,3,DeadAGI(a2billing.php|1)
05:26.53katakefalosexten => _1NXXXXXXXXX,4,Hangup
05:26.54russellbhttp://bugs.digium.com/view.php?id=4768
05:26.58outtoluncyeah, 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>
05:26.59russellbthere's a patch of yours that I merged
05:27.04katakefalosand it worked!
05:27.09russellb:-p
05:27.22outtoluncif you haven't realised... FU
05:27.24xtrvdKizmet: So is the problem using the 'i', or is the problem passing to the new context?   Can you isolate it a bit further?
05:27.26Nivexouttolunc: You're getting flack because you're acting like you're hyped up on speed.
05:27.32outtolunchave a nice day
05:27.36mzohey, who has speed, and isn't sharing, plzkthx.
05:27.58outtoluncwhy is it i've not seen ANY of these nick's EVER
05:28.17outtoluncnot in the 3+ years i've been on this project
05:28.20outtolunchmmm
05:28.33outtoluncanyone
05:28.44russellbwell, i usually go by drumkilla.
05:28.44outtoluncplease show me some irc logs with you before me
05:28.50outtoluncplease
05:28.55Mavviethree years on this project and only 6 little fixes?
05:28.56mzoi 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.
05:29.08outtolunchaha
05:29.11outtoluncask mark
05:29.27outtoluncmaybe, just MAYBE i was here BEFORE THE F
05:29.28evilbunymzo: it's only mostly broken :)
05:29.39outtolunc"n bug tracker DUMBSHIT
05:29.52mzohaha, i just laughed when he told me, i guess i'm used to it :P
05:30.02outtoluncbut excuse me, that's too HARSH
05:30.18outtoluncor is it
05:30.21MavvieThat's all I can get your history from. (you asked us to look at that ourselves)
05:30.25outtolunci think not
05:30.28Kizmetxtrvd, 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.
05:30.42brookshiremzo: he probably couldn't get it to compile
05:30.48outtolunc<PROTECTED>
05:31.03outtoluncdamn, better watch out
05:31.08mzobrookshire i dunno.  I kinda laughed a bit
05:31.16mzo~amp
05:31.17jbothmm... amp is NOT supported here! people using it should join #amportal
05:31.21outtolunchaha
05:31.47xtrvdKizmet: But does the 'i' extension work at all? If you want to, perhaps loop your current context?
05:31.49outtoluncdid you happen to notice the bug number on that first one?
05:32.04outtolunc195 iirc
05:32.22*** join/#asterisk freat (n=freat@h-72-244-84-43.chcgilgm.covad.net)
05:32.23NivexI really hate it when people think that longevity gives them a license to assholery.
05:32.25outtoluncback then we didn't have to play the games we do now
05:32.30Kizmetxtrvd, Well i guess i could try it a little more than i have however a goto does not work.
05:32.48freathello... anyone know how I could call an agi script when an agent answers a call from the queue?
05:32.51outtoluncno, just doing this shit for 25+ years
05:33.05outtoluncand listening to the same ol lame ass crap
05:33.12outtoluncso yeah
05:33.16outtolunci'm a butthead
05:33.25outtoluncand you are 'lame ass crap'
05:33.28Nivexouttolunc: Admitting you have a problem is the first step toward recovery.
05:33.44justinulol
05:33.47katakefalosLOL
05:34.00mzostop fighing and help me fix fwd :
05:34.05*** join/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net)
05:34.13outtoluncfwd is broken<G>
05:34.14justinuthis channel is usually relatively light on the flame wars
05:34.14evilbunyNivex: lol
05:34.21outtoluncomg life is failing
05:34.33outtoluncplease help this man
05:34.39justinuwhy bother?
05:34.46NivexI am beyond help.
05:34.47outtoluncexactly
05:34.48Nivex:-D
05:34.56justinupeople are generally going to fuck it up one way or another
05:34.59outtoluncbs is bs
05:35.07justinuthey'll find someone to help them do it wrong
05:35.18outtoluncso if you want my help, ask a REAL question
05:35.27*** join/#asterisk Psykick (n=anon@203.167.226.250)
05:35.29xtrvdKizmet: At the moment, I can't seem to think of how/why the goto doesn't work, sorry.
05:35.29Nivexouttolunc: What is the airspeed velocity of a swallow?
05:35.31outtoluncthats all i ask
05:35.33mzoouttolunc, it is?
05:35.34evilbunyouttolunc: what's the meaning of life, the universe and everything?
05:35.34Psykickhi all
05:35.36Mavvieouttolunc: how long will you continue with this?
05:35.48justinuis there a way I can use a sunset T1 to help diagnose/tune my tx/rxgains?
05:35.56outtoluncabout 1mp[h more than you, because i'd have stuffed your silly ass in first
05:36.06justinudo I need a milliwat test number at the CO i'm terminating with?
05:36.14justinuor can I just set up one over the PSTN can call it?
05:36.27outtoluncif you don't 'think so' then maybe you need to talk to others that know me
05:36.46outtoluncjust show up
05:36.48Nivexouttolunc: tsk tsk... I expected more than the same old vitriolic tripe from you
05:36.56outtoluncwhy
05:37.00justinuouttolunc: i could use your help, actually
05:37.01outtolunci'm a marine
05:37.11outtoluncwe don't play f'n games
05:37.18mzoit's the doom guy!
05:37.21justinulol
05:37.23russellbmzo: lol
05:37.23NivexI know a marine.  He's not an ass like you.
05:37.25Psykickjustinu: if I could help you I would
05:37.35mzodude, do that thing with the eyebrows?
05:37.40justinuPsykick: thanks pal ;)
05:37.55Psykickmzo: he probably can't cos they're blocking his view
05:38.03mzoasterisk 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
05:38.15Nivexouttolunc has just earned the honor of being the third person ever placed on my /ignore list
05:38.29outtoluncoh 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 ...
05:38.30outtolunchmm
05:38.32Nivexso, where were we?
05:38.43justinulol
05:38.52mzofwd not accepting my iax registration? :) help!
05:38.56outtoluncplease
05:39.05outtoluncspring von
05:39.08outtoluncsan jose
05:39.10Nivexmzo: That's been going on for quite awhile now.  It made me sad.
05:39.12Psykickmzo: there's plenty of info on the net about registering with FWD
05:39.17mzois it me, or their side?
05:39.34mzoPsykick, i already did everythign, but apparently it's not working atm, and i can't find out if that's true
05:39.37outtoluncshow up or shut up, i'm so f'n tired of this weenie shit
05:39.38Nivexmzo: It's them.  I was registered to them fine for months, then it just all of a sudden died.
05:39.52Psykickmzo: I don't know how old it is but the AAH handbook has that info (if you haven't tried that already)
05:39.55mzoso just be patient?
05:39.58Nivexinterestingly I am registered to them now
05:40.12mzoPsykick, i followed it all verbatim.  saysa n error code 29, and no one knows what it means
05:40.13PsykickNivex: ain't that always the way
05:40.31Psykickmzo: which end?
05:40.37mzolemme log
05:40.41outtoluncit's strange, no other channel i'm in thinks it's bs
05:40.45outtolunconly this one
05:40.54mzojoin #linux and say linux sucks. :P
05:41.14PsykickI don't know if I dare ... but what is bs outtolunc?
05:41.17mzoactually when someone finds your body in a few hundred years... we'll be sure to ask. :P
05:41.51outtoluncthe bs being how noone thinks i'm a marine that is more than willing to 'rip thier head off'
05:42.06outtoluncjust ask file
05:42.25outtolunceven tho he's only been here 2 ISH years
05:42.39PsykickI 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
05:42.40justinui'm willing to pay someone to help me with this pri echo issue
05:43.03outtoluncoh yes, me, poor pitiful me.. why oh why
05:43.19Psykickouttolunc: you just don't want to admit it
05:43.21mzonah
05:43.25outtoluncoh gee wally because it's f'n fun to kick the crap out of weenie
05:43.26outtoluncs
05:43.31outtolunchmm
05:43.33justinuno takers?
05:43.35Psykickjustinu: if its an echo issue it's quite likely that its your telco
05:43.38outtoluncshall i say more
05:43.42evilbunyPsykick: i thought most people needing to prove themselves got penis enhancments :)
05:43.45mzohttp://pastebin.com/566298
05:43.48justinuit's SBC, i'll pay someone to work with them to fix it.
05:44.04Psykickouttolunc: to be honest .... got no time for someone like you
05:44.18justinuif you're confident you can solve it, i'm interested
05:44.19outtoluncyet, instead of asking 'actual' questions, this is now a 'otl is a big meany'
05:44.30mzothat's the fwd iax debug log, i posted it on their forum but no answer yet
05:44.36justinui asked some actual questions... was hoping for your insight...
05:44.37outtoluncso what the fuck does it matter
05:44.49rustybjustinu which echo canceler did u set when compiling zaptel?
05:44.52Psykickouttolunc: there are better things to be done than being obliging to anyone that has the inclination of ' ripping someones head off '
05:45.00justinurustyb: running mg2 right now
05:45.07justinuzaptel-1.2.1
05:45.10Psykickand I can tell U ... that I'd be more than willing to rip yours off for you
05:45.14Psykickjust say the word
05:45.20justinulol
05:45.31Psykickhell I'll even make a nice stew from your remains
05:45.36justinulol
05:45.38outtoluncoh well growing up on a farm in MN i was 'reglected' as a person so the use of arms became everyday
05:45.43justinujeffry dahmer style
05:45.57Psykicknot quite .... NZ Kiwi style
05:45.57outtoluncso please, tell me i'm strange<G>
05:46.05evilbunythere we go, penis enhancements!
05:46.10justinuPsykick: lol, you know who jeffry dahmer is?
05:46.32outtoluncoh gee, i squished a bug
05:46.39outtoluncdrat
05:46.41Psykickouttolunc: 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
05:46.48outtolunchaha
05:46.48*** join/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net)
05:46.50justinuhaha
05:46.55outtoluncyou have no f'n clue
05:47.08xtrvdNivex: Sorry I took so long to ask this, but: Laiden or unlaiden?
05:47.16Psykickbelieve me ... it's in my culture .. and my nature
05:47.18evilbunyPsykick: watching red dragon lately?
05:47.24xtrvdNivex: I completely missed that comment up there.
05:47.24FuriousGeorgehey all
05:47.31justinurustyb: you have any recomendations on which EC to use? are there perhaps improvements in zaptel i should be using?
05:47.35PsykickI take it none of you have seen 'Once Were Warriors'
05:47.47outtoluncso psykick, when have you done ANYTHING outside the norm
05:47.52Nivexxtrvd: I... I don't know that... AAAAAaaahhh!
05:47.55evilbunyPsykick: I have, not a very great movie though
05:47.57xtrvd=)
05:48.05FuriousGeorgeso i made an account to telnet into theAPI expecting to see it spitting out all this stuff, and its just silen
05:48.06evilbunyalthough most hollywood garbage is worst
05:48.26outtoluncmeaning if i call the FBI they have a file on you obviously
05:48.26xtrvdNivex: Thank you good sir, you have made my night.
05:48.26outtoluncright?
05:48.26Nivexxtrvd: Glad to be of service.
05:48.27Psykickevilbuny: that is pretty much how things are/were for us/me when we were growing up
05:48.27justinuPsykick: so NZers like to eat people?
05:48.34evilbunylol
05:48.40outtoluncwell?
05:48.43PsykickNZer's or the maori used to be canibals
05:48.55justinuyou a native?
05:48.56Psykickouttolunc: gee lets see
05:49.00PsykickI am a native
05:49.05justinugotcha
05:49.17*** join/#asterisk Eggplant (n=none@dsl-352.cascadeaccess.com)
05:49.26outtolunci'm a 40+ marine that is tired of the bs from you candy ass mf'ers
05:49.27evilbunyPsykick: auckland airport = sucks, no wifi
05:49.32Psykickouttolunc: I've been in more fights than I can remember ... probably why I can't remember
05:49.34outtolunchow is that
05:49.39rustybjustinu you have a recent zaptel. i was checking to see which EC i'm using. I think its mark2
05:49.54outtolunci have a dd214 do you
05:49.56justinurustyb: k... the default for my version was KB1
05:50.00Psykickouttolunc: killed a guy for giving my g/f a hard time
05:50.04outtolunci doubt it
05:50.07justinuheh
05:50.10outtolunchaha
05:50.11Psykickouttolunc: broke another guys legs
05:50.15Psykickouttolunc: broke another guys shoulder
05:50.21outtolunc<PROTECTED>
05:50.23Psykickouttolunc: tore 3 limbs off another
05:50.24justinudid you stew the guy?
05:50.31outtoluncright now
05:50.33Psykicknot completely off
05:50.37justinuhmm
05:50.42justinuthat's pretty brutal
05:50.42Psykickbut enough now that he can't do jack
05:50.54outtolunchaha
05:51.23Psykickgave one guy such a beating that the hunchback of notre dame would look more appealing
05:51.35justinuyou ever see casino?
05:51.38outtoluncoh yeah stay on the good side of the guy that 'might' have done something because someone/somehwhere said 'somehthing' about his gf
05:51.38PsykickI could go on and on
05:51.49outtoluncgee wally
05:51.53outtoluncFU
05:51.57Psykickouttolunc: I seriously don't care if you give my g/f a hard time
05:51.58justinu"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!"
05:52.00Psykickshe's dead now
05:52.01*** join/#asterisk sitexec (n=sitexec@69.62.163.65)
05:52.03outtoluncplease
05:52.37outtolunci'd already said, i'm showing up at spring von for ANYONE that thinks they are able.. to take a shot
05:52.50outtoluncif this means YOU
05:52.50sitexecjust wondering how to add a local exten, i add mine in extentions.conf and i still get extention not found in local context
05:52.54outtolunc<PROTECTED>
05:52.56justinuhigh noon? right in front of the digium booth?
05:53.05outtoluncfor all the rest of you... kiss my ass
05:53.33outtoluncbut saying i'm NOT gonna do this or not gonna do that
05:53.35outtolunchaha
05:53.36Psykickso ... 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
05:53.39outtoluncPLEASE
05:54.08outtoluncoh no, i must have studdered.. PLEASE
05:54.09PsykickI know myself that what I did was seriously fucked up
05:54.14FuriousGeorgeanyone know of a good source of info to get started with the API?  the wiki assumes I know stuff that i don't
05:54.28Psykickyou know what
05:54.32sitexecfeel free to message me if you can help
05:54.34outtoluncwhat the fuck part of 'i'll fuck you up' did you not understand?
05:54.46xtrvdthe 'up' part
05:54.55outtoluncand i've been saying this for AGES
05:54.55PsykickI seriously hope you do rip someones head off .... mainly ... I hope you rip your own off
05:54.57xtrvdBecause without it... it's kind of weird.
05:55.15Psykickbut if we're all not that lucky ... you'll have a nice b/f to share a small room with
05:55.28outtoluncyet none of you 'hip young 'smart' guys' seem to make it to my front door
05:55.47outtoluncnot to mention the less bugs filed
05:56.02outtoluncoh gee that must really be a bitch eh
05:56.12outtoluncany time
05:56.19outtoluncplease!
05:56.31outtoluncor are you just a weenie <G>
05:56.34Psykickouttolunc: go buy yourself a carrot
05:56.47outtoluncfor what? you gf?
05:56.49Psykickor any other vege you prefer
05:56.59outtoluncor are you a gf?
05:57.08outtolunchmm
05:57.22outtoluncgee wally
05:57.48outtoluncawww
05:57.57sitexecjust wondering how to add a local exten, i add mine in extentions.conf and i still get extention not found in local context
05:58.04outtoluncplease don't go away mad... just f'n go away <G>
05:58.45outtoluncor 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
05:58.54xtrvdsitexec: did you 'reload'?
05:58.58outtolunc'please'
05:59.24sitexecsure didnt, thanks i will give it a try
05:59.31xtrvd=)  No worries
06:00.03NivexFYI: Don't need to reload the whole PBX, just 'extensions reload'
06:00.05xtrvdsitexec: just type 'reload' from the asterisk command prompt
06:00.22xtrvdsitexec: or as Nivex just explained, just 'extensions reload'
06:00.25outtolunc'reload' does a 'base level reload' for asterisk
06:00.32sitexecxtrvd same error, no extention 'blah' in context 'local'
06:00.41FuriousGeorgeanyone know a good reference for the * api
06:01.14outtoluncif you want to reload a specific part do a 'reload the_app_func_name_.so'
06:01.22sitexechmm, i didnt see it get added
06:02.18outtoluncas for the reference of * api, i truely hate to say it ... but the source is the best
06:02.34Psykickouttolunc: genuine question .... ever do any interrogations?
06:02.40*** join/#asterisk katakefalos (i=katakefa@194.214.77.65.in-addr.arpa.ethernext.com)
06:02.59outtoluncyes i have, from both sides
06:03.09FuriousGeorgeouttolunc: thats kinda putting the horse before the carriage for a coding novice, dont you think
06:03.17outtoluncmeaning, i've been, and have done
06:03.24Psykickoh ok
06:03.37Psykickbeen practicing lately?
06:03.44outtoluncfg you are not a novice, even though you 'keep saying so'
06:03.59FuriousGeorgenovice=beginner right?
06:04.15FuriousGeorgeam i getting my spoken languages confused?
06:04.16outtoluncafter 1/5-2 of doing do either you 'get smart or you just 'hang here''
06:04.27mzoi'm too busy breaking my asterisk to talk :P
06:04.39Psykickmzo: sounds ... very .... familiar
06:04.40outtoluncfg how long have you 'been hanging in this channel'?
06:04.53outtoluncit's been almost 2ish years
06:04.53FuriousGeorgei came in hear for the first time a year ago
06:05.08outtoluncso after 'that period of time'
06:05.10FuriousGeorgeand i said coding novice
06:05.14FuriousGeorgenot * novice
06:05.20outtoluncyou are still just 'asking what'?
06:05.22Psykickfg: what language?
06:05.38FuriousGeorgein that time i did not learn C or Java
06:05.54FuriousGeorgePsykick: i dunno, heard python was easy, i knew pascal once
06:06.14outtoluncfg i never said you 'were' a novice, what i'm saying is you 'need not ask a question as if you were one'
06:06.22Psykickhaven't gone the P Y way myself
06:06.26*** join/#asterisk svenna_ (n=svenna@p548D388D.dip0.t-ipconnect.de)
06:06.31outtoluncbut if you want to ask as if
06:06.32Psykick.... done the pascal thing too
06:06.42outtoluncwe can go that route
06:06.53sitexecwhere do i put an exten in extentions.conf to get it to show up =\
06:07.03FuriousGeorgei said i was a coding novice, which my understanding of english leads me to believe it is a step up beginner
06:07.47outtoluncand 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)
06:07.57salviadudheeeeelp
06:08.03Psykicklol
06:08.12FuriousGeorgeouttolunc:  have you been drinking, cuz im struggling to understand what you are getting at?
06:08.16outtoluncso lets talk asterisk <G>
06:08.23*** join/#asterisk Goral (n=needsand@CPE0012172e9c9f-CM014080205433.cpe.net.cable.rogers.com)
06:08.27salviadudFeb 22 00:14:03 NOTICE[20764]: chan_iax2.c:7398 socket_read: Registration of '651692' rejected: 'Registration Refused' from: '192.246.69.186'
06:08.35salviadudwhats that supposed to mean?
06:08.37outtoluncwhat is your 'current issue' with asterisk?
06:09.00mzo192.246.69.186:4569   749414      <Unregistered>             60  Rejected
06:09.00mzo<PROTECTED>
06:09.24FuriousGeorgeouttolunc: 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
06:09.36outtolunc(oh and btw: the best code i'd ever written i was drunk, so whats your f'n point <G>)
06:09.40FuriousGeorgebut documentation on the API appears scarce, to put it nicely
06:09.45*** join/#asterisk Abbas (n=Abbas@203.81.220.90)
06:09.52mzosalviadud, it's broken for me too :P
06:10.04salviadudohhh
06:10.10salviadudi shouldn't worry then...
06:10.11outtoluncso, you are now able to admin an asterisk box (after 1 year +)
06:10.34FuriousGeorgethats more or less accurate
06:10.42FuriousGeorgeinsert small
06:10.45outtoluncand yet you inject yourself into a convo you think one of the parties 'might' be drunk...
06:10.49outtoluncgee wally
06:11.01FuriousGeorgeim pretty convinced now
06:11.22outtolunconly took, what, 6 gee wally's
06:11.31FuriousGeorgelol
06:11.58outtolunci hate to say it, but if you guys are 'our brightest' we are screwed
06:12.45FuriousGeorgedont hate grandpa that shit will kill you
06:13.26outtolunci like my grandpa, he died when i was 6
06:13.41outtoluncthe other was dead before i was born
06:14.00outtoluncthen my bother died when i was 12
06:14.16outtoluncbut you all think you know me
06:14.25outtoluncso who gives a shit
06:14.44FuriousGeorgeyou are a combative old drunk, arent you :)
06:14.56outtoluncso which one of you 'intelligent mf'ers' wants to come say 'hi to me'
06:15.20outtolunci've not changed one iota in the last hour+
06:15.29outtoluncat all
06:15.31*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
06:15.40outtolunchave i been attacked by others
06:15.41outtolunc<PROTECTED>
06:15.41FuriousGeorgeyou're "maintenence boozing"
06:15.48outtoluncin various forms
06:16.01outtoluncbut yet you still want to attack 'just me'
06:16.07outtoluncok, here we go
06:16.11evilbunyFuriousGeorge: give him a mickey fin reciepe :)
06:16.48FuriousGeorgeevilbuny: would that help, him?  i'm not sure what it is, tbh
06:16.55outtoluncfg in the time you have been in this channel i can NOT believe that you have only accumulated the knowledge you have...
06:17.08FuriousGeorgeyeah, its  tough
06:17.13evilbunywhat the air stewedesses give drunk people to put em to sleep on planes :)
06:17.24outtoluncoh damn, i'm drunk, i should be more defensive
06:17.33FuriousGeorgei always say the hardest thing about asterisk is propper linux administation
06:17.38outtoluncfg i think you need to spank yourself
06:17.48FuriousGeorge?
06:17.58outtoluncdamn, thats not drunk that's just idotic
06:18.00evilbunyhe likes it when you do that fg :)
06:18.06outtoluncwell GEE WALLY
06:18.08FuriousGeorgedude, you can tab complete my name
06:18.10evilbunyit turns him on :)
06:18.29*** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com)
06:19.24outtolunci'm so glad i can come here to remind myself not to help you guys
06:19.46FuriousGeorgelol, its probably better that way
06:19.47[hC]bahaha
06:19.57[hC]there needs to be like, a divide here.
06:20.08FuriousGeorgea drunk tank?
06:20.24[hC]asterisk at home help, "What is SIP/Asterisk/VoIP" channel, and an actual asterisk channel for intelligent conversation
06:20.28outtoluncwhich is why when you do a lookup for mods done by FuriousGeorge you'll find MANY
06:20.35FuriousGeorge~amp
06:20.35jbothmm... amp is NOT supported here! people using it should join #amportal
06:20.37outtoluncinsert any of you lamers
06:20.46[hC]one down.
06:21.26Qwell[hC]: heh
06:21.29outtolunci came here to help and if you review i did, till someone thought they could come after me
06:21.30Qwell[hC]: there already is
06:21.34[hC]Sup qwell..
06:21.36FuriousGeorgedude, 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
06:21.42Qwell#amportal, #asterisk, and #asterisk-dev
06:21.47Qwell:p
06:22.05[hC]Oooh look at mr wikipedia over here... haha :P
06:22.28outtoluncyou surely aren't talking about me
06:22.30salviadudgui's are for pussies!
06:22.37[hC]No, qwell.
06:22.43outtolunci've NEVER posted to ANY wiki
06:22.44outtoluncever
06:22.50Qwellouttolunc: shh
06:22.54Qwellnot everything is about you
06:22.56Qwellgo play
06:22.58outtolunck
06:24.23mzoi broke my asterisk, oops :)
06:24.36FuriousGeorgedid it hurt
06:24.41mzoyah
06:24.43mzosomeone called
06:24.47mzoand i got like 3000 pages of scroll
06:24.57*** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com)
06:25.00FuriousGeorgeunfortunate, that
06:26.10mzoyay
06:27.25outtoluncthis gets better right?
06:27.26FuriousGeorgeso 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
06:27.43outtoluncguess not
06:28.00outtoluncafk
06:29.39*** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de)
06:29.44exstaticaanyone seen this?
06:29.44exstaticaStarting Zap/4-1 at internal,s,1 still failed so falling back to context 'default'
06:29.59outtolunchaha
06:30.03exstaticai'm trying to get incoming calls from pstn to work
06:30.18outtolunckarma is a bitch
06:31.02[av]baniFuriousGeorge: you need to login first
06:31.29FuriousGeorgehttp://pastebin.ca/42743
06:31.37FuriousGeorge[av]bani: check it out i tried
06:31.48outtoluncput 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
06:32.41outtoluncand if 'others' tell you it's ok to play in '[default]' land, you are on your own <G>
06:32.43[av]baniFuriousGeorge: its wrong
06:32.50FuriousGeorgeexstatica: put your zap channel in the "incoming calls context", whatever you wanna call it
06:32.53FuriousGeorge[av]bani: duh :)
06:33.06[av]baniAction: login
06:33.12[av]baniUsername: blabla
06:33.17[av]baniSecret: password
06:33.31FuriousGeorgethen it will acknowledge me?  lets see
06:33.32[av]baniEvents: on
06:33.38[av]baniand two CR's
06:34.15FuriousGeorgewow, exactly how i thought it should work, i cant belive i didnt think to try that :)
06:34.28outtoluncgee wally
06:34.31exstaticaFuriousGeorge: what do you mean?
06:34.33*** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu)
06:34.42outtoluncsend me to [default] PLEASE
06:35.13outtoluncsad thing is he probably didn't get that
06:35.59FuriousGeorgeexstatica: in your zapata.conf file you see context=default somewhere?
06:36.26FuriousGeorge[av]bani: are the commands case sensitive
06:36.40FuriousGeorgeAction vs action
06:36.50[av]baniFuriousGeorge: try it and find out!
06:37.03FuriousGeorge[av]bani: maybe i will
06:37.23exstaticaFuriousGeorge: i see context incomming
06:38.08*** join/#asterisk CKGC (i=CKGC@202.8.86.162)
06:38.11FuriousGeorgeyou spelled incoming wrong :)
06:38.15FuriousGeorgewell thats ok
06:38.32FuriousGeorgejust put in extensions.conf
06:38.33outtoluncso ok, you felt you had to say something <G>
06:38.34exstaticaquestion, 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)
06:38.49exstaticathe signaling is the opposite of what the modules is?
06:38.53FuriousGeorgeouttolunc: me and you both, brother
06:39.04outtolunchehe, not sure of that
06:39.15outtolunci'm a bit darker
06:39.51outtoluncremember, my brother died when i was 12
06:40.05FuriousGeorgeanyway, exstatica, every channel below that line is gonna default to incomming context
06:40.33exstaticahttp://pastebin.com/566318
06:40.39exstaticai think i have them mixed up
06:40.42FuriousGeorgeso make sure in extensions.conf you have a [context] called [incomming]
06:40.55exstaticamodule 1 is my fxs card, and 4 is fxo
06:41.33outtoluncso glad i wasn't in that convo
06:42.21outtoluncall that for a 4 port fxs/fxo board gone arye
06:42.36outtoluncbloody f'n hell
06:45.09FuriousGeorgehttp://pastebin.com/566320
06:45.25FuriousGeorgedefine it once, its good for every channel defined below
06:45.31FuriousGeorgeeasy when you think about it
06:46.01outtoluncthat is unless you look at inherited channel vars
06:46.24FuriousGeorgeso [from-pstn] is the context that needs to have an s extension which eventually calls you
06:46.33outtoluncbut i'm sure you were being 'general' kinda like 'default' <G>
06:46.37FuriousGeorge(some people like to answer & wait first)
06:48.05outtolunci want to ask you
06:48.33outtoluncdo you really understand why 'fall thru to [default]' is a bad thing
06:48.41outtolunc?
06:49.15outtoluncif you don't, there is no reason to continue on my part
06:49.17FuriousGeorgeouttolunc: shhhh, busy
06:49.31outtoluncah ok, you are busy.. gotchya
06:49.53dudesBusy getting your internet gf hooked up?
06:50.09FuriousGeorgeexstatica: dont forget to reload extensions
06:50.22outtoluncdudes, hush <G>
06:50.34outtoluncty, but hush <G>
06:51.15outtoluncwell, actaually that isn't fair to you
06:51.26outtoluncsince you have been though this
06:51.38outtoluncyou poor scared being you
06:53.29FuriousGeorgeouttolunc: 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
06:53.45outtoluncstate monitoring
06:54.05FuriousGeorgew/ devstate
06:54.05outtoluncis not state monitoring in any sense in any version
06:54.10FuriousGeorgebristuff
06:54.17outtoluncregardles
06:54.37outtolunct1/pri/bri/whatever
06:54.37FuriousGeorgewhat?
06:55.01FuriousGeorgemake your point man
06:55.03outtolunctrying to get 'channel states' in a 'nice manner' is not happening
06:55.15outtoluncyou can understand that right?
06:55.17FuriousGeorgehow hard can it be?
06:55.34FuriousGeorgeyou write all these mods and yet we cant monitor 5 parking spots?
06:55.41outtoluncthe reason why is that for it to be helpfull, state change happens ALOT
06:55.52outtoluncnonono
06:56.03outtolunc'monitor 5 parking spots' is simple
06:56.08FuriousGeorgethat get used say 50 times a day
06:56.19FuriousGeorgeso you gotta set devstate 10 times for each
06:56.22FuriousGeorgewhere does it get hard?
06:56.27outtoluncmonitor all 500 channels in those 5 parking LOTS is a bitch
06:56.42outtoluncwhich part is hard for you to understand?
06:56.46FuriousGeorgespots not lots
06:56.50outtolunc5 spots
06:56.56outtoluncis like 5 channels
06:57.09outtoluncis like why is there a f'n prob?
06:57.13outtoluncit's 5
06:57.30FuriousGeorgefont size="2">(01:56:27) outtolunc: monitor all 500 channels in those 5 parking LOTS is a bitch <--- who's not understanding
06:57.38outtoluncnot 96, times 2 or 4 or 6
06:58.28FuriousGeorgesip-so and so was transfered to parking spot 1, set devstate.  sip-so and so was transfered out of parking spot 1, set devstate
06:58.36FuriousGeorgei dont see where the disconnect is
06:59.06outtoluncfg, 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)
06:59.43FuriousGeorgemy bad for encouraging you to speak
06:59.57FuriousGeorgewont happen again
07:00.00outtolunchahah
07:00.09outtolunchave a nice f'n day
07:00.19Juggieladies, ladies
07:00.53outtoluncoh btw, please feel free to try and make my feel inferior in ANY WAY SHAPE OF FORM
07:00.58outtolunchah
07:01.24outtoluncer OR (since you didn't catch that)
07:01.31Juggieeveryone give outtolunc a hug
07:01.49FuriousGeorgeno, he reaks of cheap rum and feet
07:02.03outtolunctosses a slurpie (big ugly green sickie thing) at juggie
07:02.08outtolunchehe
07:02.18outtoluncmy luck you'ld catch it <G>
07:02.25*** join/#asterisk Goral (n=needsand@CPE0012172e9c9f-CM014080205433.cpe.net.cable.rogers.com)
07:02.33Goralsorry about that all
07:02.40X-Robouttolunc, please ignore FuriousGeorge, he seems to believe that applcation hints are easy.
07:02.53X-RobEven though I've pointed him at metermaid and various other stuff.
07:02.56outtolunchints, damn, i knew i forgot something
07:03.00jeebusroxorsis there anyway to see if my cdr is getting put into mysql?
07:03.09Goralis there any other module for webmin other than thirdlane?
07:03.20dpryojeebusroxors: Make a call, and look in the table.
07:03.31jeebusroxorsdpryo; it should be any call correct?
07:03.39outtoluncis it me, or does that question seem. out there
07:03.44jeebusroxorsthen what? select * from cdr
07:03.46dpryojeebusroxors: yes, unless you've defined some special cdr-events
07:03.58jeebusroxorsouttolunc; my question? heh
07:04.20FuriousGeorgeX-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)
07:04.22Juggiejeebusroxors, mysql status
07:04.23outtoluncthirdlane did NOT invent, mysql, nor anything that happened between that and the 'advent of thirdlane'
07:04.26Juggiethink the command line is
07:04.41Juggieor it might be cdr mysql status
07:04.42outtoluncso you MIGHT want to look for useful tools, THEY DO EXIST
07:04.43Juggiei forget
07:04.44Juggielook
07:05.13jeebusroxorsi will Juggie - thanks
07:05.14FuriousGeorgeX-Rob: you know im using bristuff patch with this right?
07:05.36X-RobYeah, because you want devstate.
07:05.42jeebusroxorsfyi im trying to get asterisk-stat setup and i dont have any calls listed
07:06.13FuriousGeorgeX-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?
07:06.20FuriousGeorgecant i set a devstate based on that?
07:06.23outtoluncjeebus.. i've not seen an actual question
07:06.34X-RobFuriousGeorge, no, you can't.
07:06.43outtoluncat most, i have 'this installed' and it's 'not working'
07:06.44X-Robyou want application hints. Metermaid gives you that.
07:06.53justnulling2what is the best was to setup vm for smallbusiness that wants to look like largebusiness by having more ext then ppl?
07:06.56X-RobIf you don't want to use trunk, then you can't use metermaid, so you're stuck.
07:07.03jeebusroxorsouttolunc; i was asking if there was a way to check if my CDR was in mysql
07:07.11X-RobAnd, really don't use trunk in a production environment
07:07.13dudesdo a query and see
07:07.18outtoluncwell cdr in mysql is a prob
07:07.34X-RobSo what you want to do can't be done, yet. Give it 6 months
07:07.35outtoluncwhat version you using of everything
07:07.37FuriousGeorgeX-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
07:07.42X-Robyou know, I did say that the very first time you joined and asked that question
07:07.55jeebusroxorsasterisk is 2.4
07:07.59X-Robbecause you can't get info about park and unpark.
07:07.59*** join/#asterisk Gir19 (n=JDepp@67.189.110.174)
07:08.02jeebusroxorssql is .18
07:08.10X-RobWoo.
07:08.13outtoluncmeaning, mysql cdr's has NOT, and will NOT be standard for what, the last 1.75 years
07:08.20X-Robjeebusroxors has a time machine and has asterisk 2.4!
07:08.23FuriousGeorgeX-Rob: i dont remember you saying that, tbh
07:08.30outtoluncbut noone is counting
07:08.32jeebusroxorshehe
07:08.42X-RobDoes it have application hints? FuriousGeorge wants a copy.
07:08.43jeebusroxorsouttolunc; so what- postgres is standard?
07:08.49FuriousGeorgeX-Rob: np, ill use meetme's, itll be a bit harder but ill learn more and thats the point anyway
07:08.58outtoluncso, you have what asterisk, and what asterisk-addons versions?
07:09.08brookshireuse odbc cdr :)
07:09.25outtoluncjeebus, yes, pgsql IS STANDARD nowdays
07:09.31jeebusroxorsbrookshire; ive never played with odbc and am having errors with it heh
07:09.43brookshirehttp://www.voip-info.org/wiki-Asterisk+cdr+odbc
07:09.45brookshire??
07:09.46outtoluncif you aren't aware of that, you HAVE been gone 'awhile'
07:10.13jeebusroxorsouttolunc; im a newb to asterisk heh
07:10.38outtoluncby any chance do you remember the gpl/lgpl mysql fiasco almost 2 years ago
07:10.39jeebusroxorsbrookshire; i read that - im getting errors on connecting to my ds - isql isnt working
07:10.49FuriousGeorgeX-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
07:10.55outtoluncwell, thats how long it's been
07:10.58outtoluncso
07:11.21outtolunceither, read some more, or dare i say, figure something else out
07:11.37brookshirebut.. odbc is the way to go with asterisk
07:11.41outtoluncit's been almost 2 freakin years for petesake
07:12.18brookshirei'm sure it's just a simple configuration misconfiguration
07:12.32jeebusroxorsbrookshire; im sure too - just gotta find out where heh
07:12.34brookshirehe
07:12.35brookshireheh
07:12.57Gir19I have a copy of ppc-asterisk0.1
07:13.06outtoluncsweet
07:13.12outtoluncyou gonna share?
07:13.29Gir19I'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)
07:13.48Gir19It tends to lockup my ppc every couple of hours.
07:14.01outtoluncisn't life great
07:14.30Gir19but I wanted to be able to make iax2 calls from my ppc's wifi.
07:14.38outtolunci get to listen to assholes all night give me shit about asterisk and 'i've given back already' and here is you
07:14.47outtoluncgee wally
07:15.04outtoluncbut i'll say cool and move on
07:15.13outtolunci'm just not it the f'n mood
07:15.41Gir19ah, havin some clients that are not pleased with asterisk?
07:15.57outtolunchaha clients, hell ASTERISK PEOPLE HERE
07:16.08Gir19oh, lol
07:16.11outtoluncpeople i'm trying to HELP
07:16.43outtoluncstrange eh
07:17.07Gir19yeah, I was helpin earlier and some just couldn't believe me or just refused.
07:17.31outtoluncyou are real aren't you? <G>
07:17.41Juggieouttolunc, seinfeld: serenity now
07:17.48outtoluncdamn
07:17.53Juggienothing good will come from venting, some people are just slow they cant help it :)
07:18.03outtoluncoh oh
07:18.13outtolunci got a good one (today even)
07:18.24outtolunclets see if i get this correct
07:18.34Juggiewe want even the people who dont try, to learn
07:18.39Juggiethey will get it eventually
07:18.40Gir19it's all good. I get vented on and sometimes I need to be the one venting, it all balances out eventually.
07:18.49outtoluncactually i think i already said it tonight
07:19.12Juggiewe dont want to turn people away from * just because we dont like their question.
07:19.21outtolunci never have
07:19.25Juggiesometimes just pointing people at voip-info and telling them to go read
07:19.28Juggieis a good answer
07:19.43Gir19I also point them to asteriskguru
07:20.46Gir19the 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.
07:21.09outtoluncthe prob with that site, they want help with '
07:21.18outtoluncthier software'
07:21.27outtoluncbut it's conditional
07:22.07Gir19I'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.
07:22.30outtoluncmeaning, 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)
07:23.01outtoluncso , i'm not too happy to 'be thankful'
07:24.26Gir19anyone know much about how well asterisk runs on an AMD system? I ask cause I am considering switching from Intel chips to AMD.
07:26.09brookshirepretty well :)
07:26.39Gir19any major differences with the 64 chips?
07:26.53outtoluncthe thing that i'd watch for is the threading
07:27.32Juggiewhy would threading be effected?
07:27.33Qwellyeah, AMD processors don't thread at all
07:27.40outtolunci run it on a DC intel with HT disabled just fine, with HT enabled it gets alittle freaky sometimes
07:27.54Juggieouttolunc, i run asterisk on dual EMT64 xeon systems
07:28.06Juggie(2 real, 4 virtual processors total)
07:28.09Juggieand i have no problems
07:28.12outtoluncwhich ones?
07:28.19JuggieXeon 3.2 EMT64
07:28.28outtolunci ran mine with HT enabled for the first 3 weeks
07:28.31X-RobMmmmshiny xeon
07:28.37Juggierunning centos 4.2 64bit
07:28.42Juggie5 gigs of ram
07:28.44brookshireqwell: they can thread better than mac processors.. they only have one thread
07:28.45outtoluncbut there were issue after issue
07:28.45Qwelleww!  centos?!
07:28.46brookshirelol
07:28.47Juggie500gigs of hdd.
07:28.50Qwellbrookshire: haha
07:29.04JuggieQwell, its corporate, centos is close to rhel
07:29.13QwellJuggie: Just messing with you. :)
07:29.14Juggieand also all the tools for the servers eg the HP server tools
07:29.22Juggiework on centos
07:29.26Juggiebecause they were desgned for rhel
07:29.30Qwellcentos is fine, as long as it isn't *@~
07:29.37Juggiehaha
07:29.40Juggiecommon
07:29.42Juggiewhat do you think
07:29.46Qwell:p
07:29.46Juggieyou met me
07:29.50Qwellnewb :D
07:29.50Juggiedo i see like a @home kinda guy
07:30.16Juggieguys who run @ home dont drink like 6 pitchers at a hick bar and still go to the 9am presentation the next day :P
07:30.21Qwellheh
07:30.31QwellYou had cheap American beer. ;]
07:30.38brookshirei bet centos only has one thread
07:30.39Juggie5$ for a pitcher
07:30.43Juggieis good though
07:30.48Juggieeven the good beer was like 6$
07:30.48Qwellbrookshire: centos has like 2 threads
07:30.53Qwell1 is for the kernel
07:30.57brookshireahh.. right..
07:31.11QwellThat's seriously the cheapest beer I've ever seen
07:31.11Juggie[root@TRN-HTTP-SV01 ~]# free
07:31.11Juggietotal used free shared buffers cached
07:31.11JuggieMem: 5015180 4582256 432924 0 55440 4293280
07:31.16Juggieumm
07:31.18outtolunc$5/pitchers.... damn starts having a party
07:31.22brookshirefree -m !
07:31.42Juggie[root@TRN-HTTP-SV01 ~]# free -m
07:31.42Juggietotal used free shared buffers cached
07:31.42JuggieMem: 4897 4474 422 0 54 4192
07:31.48brookshire:)
07:31.53Qwell-m?
07:31.57Juggiemegs
07:31.57brookshirefree naked is sooo over rated
07:32.07Qwellahh...silly free
07:32.09Qwellshould use -h
07:32.09outtolunche could have done -h <G>
07:32.14brookshirehah
07:32.17outtolunchehe
07:32.23Qwell-h doesn't exist on free though
07:32.23brookshireqwell: we should right a patch
07:32.32Qwellbrookshire: we should
07:32.39brookshireand submit it to them
07:32.43Qwellthough...
07:32.44brookshirei wonder what they would say
07:32.46QwellI'd write one
07:33.13brookshirei can just see it now...
07:33.26brookshire"here's for not following standard unix -h"
07:33.32brookshirelol
07:33.36Juggiemy free does -h
07:33.42QwellJuggie: what version?
07:33.51*** part/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net)
07:33.56Qwellprocps version 3.2.5
07:34.09Juggie[root@TRN-HTTP-SV01 ~]# free -V
07:34.09Juggieprocps version 3.2.3
07:34.13Qwellwtf
07:34.14Qwellthose newbs
07:34.16Juggiewell,
07:34.19Juggiei see now
07:34.24Juggieit says invalid option
07:34.25Juggieand prints help
07:34.27Qwellheh
07:34.30brookshirehaha
07:34.35Qwellis it in the manpage?
07:34.40outtoluncare you two done yet <G>
07:35.05Qwellit should totally follow like how du does it
07:35.17X-RobOooooh
07:35.21Qwellwith GNU opts
07:35.29brookshireqwell: and df
07:35.33Juggiejeeze, i havnt updated centos in a while
07:35.34Qwellyeah
07:35.37Juggie39 updates avail
07:35.41X-Rob(all I did was tick 'use multiple burners' in nero, but I still feel proud of myself)
07:35.47Juggiehaha.... anyone watch king of the kill
07:35.51Juggiethe best eppisode is on fox
07:35.52Qwellshould be so easy to do
07:35.58Juggie"thats my purse, i dont know you"
07:36.08Qwellhell...it probably uses the same code for -b, -k, -m, -g
07:36.18Qwellerm, no -g
07:36.26Juggiemine has -g
07:36.33brookshirethey are going to have to add a -t one day
07:36.33Qwellerm...yeah, so does mine
07:36.37Qwellnot in the man page though
07:36.39*** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-21.claranet.co.uk)
07:36.40brookshiresee.. it just keeps grown
07:36.53brookshiregrowing
07:36.57Juggiehmmm
07:37.02Qwell-t is already used in free
07:37.03Juggiehow can i schedule yum to auto update
07:37.05outtoluncsorry, bad 'purse' thing
07:37.16QwellJuggie: cron?
07:37.19outtoluncimagine that <G>
07:37.31outtolunctag every purse in 'a city'
07:37.41outtoluncnow track
07:37.42outtolunc'em
07:37.47Qwellhaha, this rocks
07:37.58outtolunceveryone, everyplace, all the time
07:37.59Qwellyum has been running a "yum remove xorg-x11-libs" on my router for like...
07:38.02Qwell3 hours
07:38.09outtolunceww
07:38.13Juggiei think theres a yum service
07:38.13brookshirejuggie: /etc/init.d/yum start
07:38.14Qwelland it's still calculating deps
07:38.48Qwellthat freaking rocks
07:38.48outtoluncwell rock on, i'm gonna get some rest
07:38.48Qwellchkconfig yum on
07:38.48outtoluncgnight
07:38.48Juggieyah, i just found that, thanks
07:38.53Juggieonly problem is i need to use a http server to get out
07:39.01Juggiei wonder if the proxy env var will exist
07:39.07Qwellshould
07:39.19Juggiei think i'm setting it in .profilerc
07:39.21Qwellif not, you can put it in the /etc/sysconfig/ file it calls
07:39.23Juggiei forget where i put it
07:39.27Juggieyah true
07:39.37brookshireqwell: here is one for you
07:39.39salviadudhey, what does . mean?
07:39.40Qwellit should run a shell though
07:39.44salviadudas in period?
07:39.45Qwellsalviadud: in what context?
07:39.51salviadudfor example...
07:39.53Qwellin unix, it means current dir
07:39.56brookshiremv /etc/rc3.d/K01yum /etc/rc3.d/S01yum
07:39.57salviaduddialing an extension
07:40.06Qwellin Asterisk dialplan, it means "match any number of anything"
07:40.07Juggie64bit linux screams
07:40.07brookshiremv /etc/rc5.d/K01yum /etc/rc5.d/S01yum
07:40.08Juggieits so nice
07:40.09brookshire:D
07:40.16Qwellbrookshire: silly debian ;)
07:40.23salviadudohhhh
07:40.24Juggiebrookshire, i only run in level3, 5 is evil :)
07:40.26salviadudthanx
07:40.30Qwellchmod +x /etc/rc.d/S01yum
07:40.40salviadudi use slackware... no probs here...
07:40.50Juggie5 is xwin right?
07:40.52QwellI prefer `rc-update add service default`
07:40.56QwellJuggie: centos?  yeah
07:41.03Juggieyah, i dont run xwin
07:41.03brookshireyum remove yum
07:41.07X-Robor 'chkconfig yum on'
07:41.12QwellX-Rob: see above
07:41.29FuriousGeorgeive 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
07:41.47brookshirejuggie: can you actually run in any other level?
07:41.59QwellFuriousGeorge: never heard of a registration manager, but you can use externhost
07:42.02Juggiebrookshire, i'm sure you can
07:42.09camonzhi, if i want to send a call to voicemail after it has ringed 10 secs to a extension how would i do that?
07:42.15QwellRH distros have 2, 3, 5
07:42.16camonzwith n+101 priority?
07:42.17brookshireinit 11
07:42.19brookshirelol
07:42.21Qwell(and 0, 1, 6, of course)
07:42.22FuriousGeorgeqwell, thanks ill check into that
07:42.27brookshire2 is single right?
07:42.30brookshire6 is reboot
07:42.31Qwell1 is single
07:42.34Juggielook in ... /etc/inittab
07:42.35brookshire0 is powerdown?
07:42.36Juggiethey are all in there.
07:42.36Qwell0 is startup, 6 is shutdown
07:42.42brookshireoh yeah
07:42.43Qwell2 is no network CLI
07:42.56Juggie# Default runlevel. The runlevels used by RHS are:
07:42.56Juggie# 0 - halt (Do NOT set initdefault to this)
07:42.56Juggie# 1 - Single user mode
07:42.57Juggie# 2 - Multiuser, without NFS (The same as 3, if you do not have networking)
07:42.57Juggie# 3 - Full multiuser mode
07:42.59Juggie# 4 - unused
07:43.00brookshirei knew this at one point
07:43.01Juggie# 5 - X11
07:43.02Qwellyeah
07:43.03Juggie# 6 - reboot (Do NOT set initdefault to this)
07:43.07brookshirebut it's kinda pointless
07:43.10Qwelloh...0 is halt
07:43.15brookshireHAHA
07:43.19Juggiewho cares, use 3 :)
07:43.22brookshirejuggie: # init 6
07:43.37Qwellecho "telinit 6" >> /etc/local.start
07:43.37Qwell:D
07:43.44Juggieif i actually want to reboot a server, can i do init 6
07:43.47brookshireeveryone should use init 69 man!
07:43.49QwellThat's a real bitch to figure out
07:44.11brookshireturn your server into a "love machine"
07:44.21Juggiethe only love machine in the lab is me
07:44.31Juggiei dont need any competition
07:44.53brookshiredepends on who you're loving.. i guess.. lol
07:45.26salviadudis anybody else having trouble with FWD?
07:45.35salviadudim 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'
07:46.14brookshirei guess they don't like you anymore :(
07:46.29salviadudthey never liked me! im a prankster...
07:46.41salviadudi call rehab clinics and lie that im drunk
07:46.47salviadudwith a redneck voice
07:46.53FuriousGeorgeQwell: 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
07:47.18FuriousGeorgei just thought there should be some way to have it not cache the ip, which appears to be what * is doing
07:47.19IronHelixahlll tell yoo whnnn ivh had enufffff....
07:47.37salviadudhahaha, yeah, iron thats the spirit
07:48.03IronHelixhehe
07:48.10salviadudone guy wanted to call an ambulance, and i told him "no son.... im over here *hick* attt. walmart, im getting more booze"
07:48.31FuriousGeorge~externhost
07:48.32salviadudi want to mess around with the monitor function
07:48.41*** join/#asterisk P0L0 (n=n0n3@62-43-65-175.user.ono.com)
07:48.44salviadudtape all my pranks and social engineering hacks
07:49.25salviadudwish me luck... cya guys in the mornin'
07:49.31IronHelixcya
07:49.35IronHelixalso- dont abuse hotlines
07:49.37IronHelixyeah
07:49.38IronHelixthats bad
07:49.43FuriousGeorgethe wiki page is down but it looks like its for sip
07:49.52salviadudhaha, hotlines are funny too
07:50.15*** join/#asterisk vgster (n=vg@host217-45-221-53.in-addr.btopenworld.com)
07:50.31brookshirewhat's a hotline?
07:50.34brookshirehaha.. just kidding
07:50.37IronHelixhehe
07:54.19glm2kbrookshire: what's a batline? hehe
07:54.38brookshirewe need one of those at work
07:54.39*** join/#asterisk pengyong (n=lala@222.188.134.10)
07:55.10IronHelixsetup a red phone on the wall
07:55.21IronHelixatt slimline style if you can find one
07:55.28IronHelixput a removable plastic cover over it
07:55.49IronHelixthen explicitly tell everybody to not ever, ever pick up or use that phone, no matter what
07:55.56brookshirecortelco!
07:56.12FuriousGeorgeexten => s,1,dial(${POTS_OUT}/${COMISSIONER_GORDON})
07:56.18brookshirelol.. yeah.. that would be greaet
07:56.25brookshirei wonder if i can find one of those on ebay
07:57.05IronHelixthen 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
07:57.17glm2klol
07:57.18FuriousGeorgei use a slimline mounted on my front foor to replace the functionality of the doorbell my landlord wont fix
07:57.23IronHelixas well as monitor()ing everything for future amusement
07:57.56FuriousGeorgebatphone mode, of course
07:58.15trixterbut does it ring to commisioner gordon?
07:58.23*** join/#asterisk astar` (n=astar@ANantes-154-1-27-237.w81-53.abo.wanadoo.fr)
07:58.37FuriousGeorgeno, it actually calls the local cluck-u chicken
07:58.52FuriousGeorgewhich really confuses them when they deliver
07:58.53IronHelix(then if you get bored, call the red phone, which will be wired to flash brightly when it rings)...  immediately fire whoever answers it
08:00.26sl16Feb 22 10:05:46 NOTICE[3952]: chan_sip.c:3593 process_sdp: No compatible codecs!
08:00.29sl16how to fix this
08:00.35Qwellsl16: use compatible codecs
08:00.49sl16i have those 2 licenses
08:00.53sl16of g729
08:01.10sl16and the other side also has g729 NetCentrix or sort of
08:02.16sl16i also get: SIP/2.0 488 Not acceptable here , when trying incoming calls
08:02.55Qwellyes...ypu arem
08:03.00Qwell't using the same codecs they are
08:03.06Qwellneed to allow the right ones
08:03.29sl16i am only allowing g729 which is the one my VoIP provider working with
08:03.52sl160/0 encoders/decoders of 2 licensed channels are currently in use
08:03.56sl16the module is loaded
08:04.04sl16i 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)
08:11.34X-Robsl16, use 'allow=all'
08:11.44X-Robyou've got ever codec, let them pick which ever one they want
08:11.48X-Robevery
08:12.38sl16ok, just a second
08:13.55sl16no compatible again
08:14.19sl16the interesting thing i that a phone behind * can do the calls
08:14.19X-Robthen you didn't reload sip, or they are broken.
08:14.29sl16but * can't
08:14.35sl16i've reload it
08:14.45sl16what is "broken" mean ?
08:14.45X-Robsip debug
08:14.48X-Robpost to pastebin
08:14.52X-Robpastebin.ca
08:17.33*** part/#asterisk anandbabu (i=ab@69-12-132-138.dsl.static.sonic.net)
08:17.52camonzi'm having a prob with voicemail conf
08:18.08camonzi'm creating a voicemail account like this : 1002 => 0000,Simon,camonz@localhost,attach=no
08:18.19sl16http://pastebin.ca/42746
08:18.22sl16X-Rob:
08:18.31camonzbut when entering voicemail main and entering the pw the authentication fails
08:19.16X-RobCapabilities: us - 0x8000e (gsm|ulaw|alaw|h263)
08:19.23camonzexcept that when testing the example account 1234 it works allright
08:19.25X-Robtry 'show translations'
08:19.51Abydos313hi everyone
08:20.02{zombie}camonz: did you "reload" asterisk after making changes to voicemail.conf?
08:20.05camonzyep
08:20.16camonzseveral times
08:20.27Abydos313looking for some help on equipment choice
08:20.49camonzdo 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?
08:21.00camonzyep
08:21.15sl16X-Rob: http://pastebin.ca/42747
08:21.17{zombie}asterisk creates the directories itself
08:21.19camonzactually i first had it in other context and then i put it on the default context
08:21.45Juggiecamonz, pastebin.ca your voicemail.conf
08:22.32camonzthanks!, had the same login in the 2 contexts
08:22.49camonzremoved it from the non default context and it worked
08:23.03{zombie}cool
08:23.26Juggiesl16, your problem is simple
08:23.30camonzi also want to dial an extension for 15 secs and then if no answer leave a voicemail
08:23.32Juggieyour phones are set to g729
08:23.40Juggiethats why they can talk via passthrough
08:23.45sl16X-Rob: i've put allow=all at the beginning sip.conf
08:23.53sl16and i have more capabilities
08:23.56Juggiebut asterisk doesnt have a g729 codec
08:24.03Juggieallow=all = bad
08:24.05Juggiedoes wacky things
08:24.18sl16ah. ok
08:24.35Juggiebut your not listning
08:24.42Juggieyour phone is set to g729 only
08:24.47sl16Juggie: the phones are not set to g729 only ...
08:24.52Juggieasterisk does not understand g729 by default
08:25.06camonzhttp://pastebin.ca/42748
08:25.23sl16Juggie: 0/0 encoders/decoders of 2 licensed channels are currently in use
08:25.35camonzis this the way to do it?, i'm using Dial(sip/${EXTEN},15,j)
08:25.53camonzhoping it will fall to n+101
08:26.10Juggiewell, sl16, asterisk is presenting
08:26.22Juggie(gsm|ulaw|alaw|h263)
08:26.37Juggieand the phone is presenting
08:26.38Juggiepeer - audio=0x100 (g729)
08:26.50Juggieno match
08:28.49Juggietry this :)
08:28.50Juggiesip.conf
08:28.52Juggiedisallow=all
08:28.54Juggieallow=ulaw
08:28.57Juggieallow=g729
08:29.02Juggiein [general]
08:29.22Juggiethen try again
08:29.58sl16Juggie: 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)
08:31.46sl16Juggie: http://pastebin.ca/42749
08:32.42k3yi 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
08:32.56Juggiesl16, well now this problem is clearly new
08:33.03Juggienot codec related
08:33.41Juggietheres no match in whatever context is associated with that peer
08:33.47Juggiefor the number its trying to dial
08:33.50Juggiehence the 404
08:34.43Juggietheres probally some other output to the console to tell you exactally what context it was looking in
08:35.38Juggienone the less, its a simple context matching problem
08:36.18Juggiehurry
08:36.22Juggiebecause i'm going to sleep
08:36.31sl16could it work putting all in one context
08:36.33sl16default
08:36.35sl16??
08:36.50sl16Juggie: ok, sorry
08:38.18Juggiesl16, it can but for organization purposes that would not be reccomended
08:38.20Juggienone the less
08:38.27Juggieif its default
08:38.50Juggieyou need to match the incomming number
08:38.53JuggieTo: <sip:4912332@193.68.217.37>
08:39.01Juggieso the incomming number is 4912332
08:39.02*** join/#asterisk jaike (n=a@203.131.137.76)
08:39.21Juggieso you could do exten=>4912332,1,Dial(SIP/somesiphone)
08:39.22JuggieOR
08:39.32Juggieexten=>491233X
08:39.36Juggieand so on
08:39.38Juggiewhatever
08:39.51*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
08:39.53Juggiei would recommend keeping things seperated
08:39.54sl16could i in register say : user:PASS@provder/ext
08:40.02brookshirek3y: email digium support support@digium.com
08:40.11brookshirek3y: sounds like a configuration problem really
08:40.20*** join/#asterisk anandbabu (i=ab@69-12-132-138.dsl.static.sonic.net)
08:40.28k3ybrookshire when my nat session expires, it continues to work
08:40.46Juggiesl16, the register doesnt determine the context in which the call is received
08:40.51Juggiethats determined in the peer configuration
08:41.14Juggielook on the wiki and read the docs on sip.conf
08:41.32Juggieregardless for the purpose of making it work for the moment, add a pattern match into default for the number dialed
08:41.38Juggieyou can learn more later :)
08:41.39sl16it worked
08:41.41sl16:)))))))))))
08:41.44sl16thank youuuuuuuuuuuuu
08:41.59Juggieidealy the best thing is that incomming sip calls, and iax calls and calls from your TDM/zap interface all go into seperate contexts
08:42.00Juggieeg
08:42.02brookshireoh yeah.. iaxy doesn't support dns
08:42.07Juggieyou should have sip-in
08:42.10brookshireso..
08:42.11Juggieand iax-in
08:42.13Juggieand zap-in
08:42.14Juggieand so on
08:42.18Juggieto keep everything seperated
08:42.39Juggiewww.voip-info.org
08:42.45Juggielots of info, now i must sleep, work in 5hrs
08:42.52Juggieand i have to drop my skis to the shop on the way to work
08:42.55Juggieso i have to get up early
08:45.35k3ybrookshire server address is static... iaxy is behind NAT... the NAT ip changes
08:47.20brookshirehmm.. 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)
08:47.48Juggiek3y, set qualify=yes
08:47.51Juggiefor the iax peer
08:48.20Juggieand if you can on the iaxy set the registration time low
08:48.26Juggiei dont know about those devices
08:49.54brookshirei don't think you can set the registration time on the iaxy
08:50.16trixterit all depends on skill level
08:50.20trixterand time
08:50.22Juggiei dont see it in iax.conf either
08:50.35Juggiebut you can set qualify=yes
08:50.35k3yJuggie my iaxy device sends packets from src port 4569
08:50.38k3ythen it works
08:50.51k3ybut whan IP changes, it starts going with src port 1024
08:51.00k3ythat what my NAT (linux, netfilter) does
08:51.08k3yi'm suspecting its the port problem
08:51.12Juggiek3y, the iaxy source port isnt changing
08:51.13anandbabuk3y, i have set qialify=yes now in iax.conf
08:51.17Juggienetfilter is writing
08:51.24k3yJuggie yes, that's what i meant
08:51.24Juggieer, rewriting it
08:51.33jaikeiax2 problems with nat?
08:51.44Juggienat w/ changing ip
08:51.51Juggiewhat kinda isp changes its ip anyways
08:52.02Juggiek3y, did you set qualify=yes
08:52.09brookshirebastard ones :(
08:52.13Juggiewith that * sends contant keep allive packets
08:52.16anandbabuJuggie, will qualifysmoothing = yes also help?
08:52.23Juggieno
08:52.24k3yJuggie dsl
08:52.28jaikeisp with a super paranoid network administrator
08:52.30anandbabuJuggie, i have access to k3y's server. configuring it
08:52.38Juggiejust qualify=yes
08:52.43Juggiethat will have * send packets
08:52.54Juggiewhich has the effect of A) keep nat connection open
08:53.02JuggieB) will notice when the iaxy isnt responding
08:53.21anandbabuk3y, done, restarted asterisk. try now
08:53.43Juggieanandbabu, type 'iax2 show peers'
08:53.49Juggieto see the status of his iaxy
08:54.56kippihey
08:55.04k3ycurrently it is sending with src port 4569 as the udp session expired
08:55.05k3yhangon
08:55.13anandbabuJuggie, bharathi/bharat  59.92.149.246   (D)  255.255.255.255  4569          Unmonitored
08:55.14k3yits come online
08:55.33kippican 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)
08:55.55brookshireis that fwd?
08:56.28brookshirenm... lol
08:56.33Juggieanandbabu, qualify=yes should make it monitored
08:56.47anandbabuJuggie, thanks i have set it now
08:56.58*** join/#asterisk saftsack (n=saftsack@p54A7EBE7.dip.t-dialin.net)
08:57.14saftsackare 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)
09:03.01maruzflexibel 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?
09:05.45jaikemaruz: have you tried removing mp3 ID info?
09:06.32maruzi have no skill on mp3 and i dunno which sw to use to do it; someonehere tried in win to change rate of mp3
09:07.10maruzso i thought that the fastest solution is download by some url an mp3 that is good for asterisk
09:07.43maruzbut in www.asterisk.org there is any mp3 suitable for moh?
09:08.06jaike1.2.4 comes with 3 mp3s for MOH
09:08.07*** join/#asterisk marktt (n=marktt@203.217.18.2)
09:08.23markttring.ring
09:08.34jaikefpm-calm-river.mp3  fpm-sunshine.mp3  fpm-world-mix.mp3
09:09.11maruz-r--r--r-- 1 root src 1939812 2006-02-07 18:35 ./sounds/fpm-calm-river.mp3
09:09.11maruz-r--r--r-- 1 root src 2217563 2006-02-07 18:35 ./sounds/fpm-world-mix.mp3
09:09.11maruz-r--r--r-- 1 root src 2582496 2006-02-07 18:35 ./sounds/fpm-sunshine.mp3
09:09.21maruzare them?
09:09.25jaikeyes
09:09.27brookshiremaruz: http://www.freeplaymusic.com/
09:09.31maruzthe size is right?
09:09.40maruzbronze: let's try :)
09:09.45anandbabumusic from this link worked for mehttp://www.sounddogs.com/catsearch.asp?Type=2
09:09.48jaikeyes
09:10.26anandbabualso after replacing mpg321 with mpg123 in my debian box, all mp3s played well
09:10.59*** join/#asterisk corruptor (n=andrew55@www.tae.ru)
09:11.08jaikemaruz: mpg123 installed?
09:11.40anandbabudebian sym links mpg321 as mpg123 - be aware
09:11.44maruzyes, i removed soft links to mpg321
09:11.54maruzin /etc/aslternatives
09:11.59maruzand so on
09:12.00maruz:)
09:16.43*** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.190)
09:16.48Kernel_corehi all
09:19.42Kernel_coreanybody used asterisk 1.2.4 with h323 module on debian ?
09:21.09justnulling2how 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)
09:23.36x86i'm trying to setup my softphone to connect to my asterisk server, but it doesnt appear to even try to register
09:23.59x86i have sip debug on in my asterisk console, but i'm not seeing anything
09:25.06x86any ideas?
09:25.14x86the asterisk box and the softphone are on the same box
09:25.18ChrisUKsoftware firewall perhaps?
09:25.24ChrisUKoh
09:26.14x86i just flushed the firewall just to make sure
09:27.03dpryoIs there any softphones without stupid sci-fi GUI?
09:28.31x86hmm
09:28.39jaikemaybe the softphone cant register to its own ip
09:28.46x86asterisk automagically acts as a sip proxy right?
09:29.59x86also, i'm trying to IAX2 trunk to freeworlddialup
09:30.05x86my asterisk server is behind a NAT
09:30.07x86is this possible?
09:31.08*** join/#asterisk Sloboda (n=slob@194.42.196.254)
09:31.35SlobodaHi! I'm looking for soft-phone for Linux that supports usb-phones.
09:33.35trixterok
09:33.36camonzmaruz did you compiled ztdummy?
09:33.39trixtergood to know
09:33.44camonzyou can do it without MP3Player
09:33.48maruzcamonz: well question :)
09:34.02maruzcamonz: i am having many troble about that
09:34.11jaikeyup..1.2 has native MOH player
09:34.30camonzmaruz: how so?
09:34.46trixterasterisk-addons format_mp3
09:34.54maruzyes, taht i have installed
09:35.06trixterjust use the native moh then :)
09:35.15trixterits a lot better and more friendly to your system
09:35.21camonzit's easier to use native MoH
09:35.30*** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com)
09:35.32maruzbut i stil miss timing source, i have visdn driver and i have posted on mialing list
09:35.38camonzjust uncomment the # ztdummy line in the makefile
09:35.43camonzfor zaptel
09:35.53camonzremove the #
09:36.00maruzsure? i just do it now
09:36.07jaikemoh needs timing source? though that was for meetme
09:36.10camonzthen make clean && make && make install
09:36.11jaikethought
09:36.29camonzalso for MoH, i just resolved that issue on sunday thanks to [av]bani
09:36.43maruzFeb 20 11:08:13 WARNING[3486] res_musiconhold.c: Unable to open pseudo channel for timing...  Sound may be choppy.
09:36.48vgsteris busydetectmartin much better than regular busydetect?
09:36.59jaikereally? good thing i got ztdummythingny compiled too
09:37.00camonzyep that's because you don't have ztdummy.so module loaded on kernel
09:37.01maruzyes, for moh too ...
09:37.57*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
09:38.07camonzafter you've made and installed zaptel do a modprobe ztdummy
09:38.09jaikewish mixmonitor bugs would be fixed soon too. shell+soxmix really eats up a lot of resources
09:38.17camonzthat should do it
09:38.37camonzoh..., in musiconhold.conf in the default context set mode=files
09:38.51camonzthen reload and you're all set
09:39.02jaikemaruz: using kernel 2.6?
09:39.36maruzjaike yes
09:39.39maruz2.6.15
09:39.49maruzby ftp.kernel.org
09:39.54maruznot that of distro
09:39.54jaikeztdummy should load with no problems
09:40.33maruzMODULES:=zaptel tor2 torisa wcusb wcfxo wctdm wctdm24xxp \
09:40.33maruz<PROTECTED>
09:40.34maruz<PROTECTED>
09:40.36maruzeh eh eh
09:40.39maruzhere it is
09:40.41camonzyep
09:40.43camonzremove the #
09:40.44maruzgreat!
09:41.10*** join/#asterisk jorgito (n=jorge@82.113.32.241)
09:41.23jorgitocant uninstal asterisk , neither upgrade . bleee tfuj shit soft
09:41.34maruzmaybe i can fix flexible rate error, too ;)
09:42.00pcmhey there
09:42.07camonzi had a similar issue, the makefile checks for a shell variable called KVERS
09:42.14pcmi need to sell a couple of tormenta2 quad t1/e1 cards, anyone interested ?
09:42.18camonzwich should be linux26 for 2.6 kernel ver
09:42.32prhKVERS is set by make-kpkg (the debian kernel build system)
09:42.46camonzi don't have that var on my shell, so it defaulted to kernel 2.4
09:42.50camonzi'm running SuSe
09:43.01prhand should be the full kernel version to compile against - ie 2.6.15-rc2-localwibble-rev1 or whatever
09:43.06prhooh
09:43.13camonzshould KVERS get a value?
09:43.49camonzthat'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)
09:45.22jorgitohave proglems with asterisk Unable to connect ro remote asterisk (does /var/run/asterisk.ctl extist?0
09:45.25camonzactually the KVERS var is the result of uname -r wich on my system turns out to be 2.6.11.4-21.9-default
09:45.36camonzjorgito: are you running as root?
09:45.41jorgitoyes
09:45.49jaikejorgito: asterisk is not running
09:45.58camonzyep
09:46.13jaikeasterisk -vc
09:46.13camonzdid you do asterisk -vvvc before going asterisk -r?
09:46.25jorgitoi did asterisk -p
09:47.12*** join/#asterisk froguz (n=froguz@188-142-222-201.adsl.terra.cl)
09:47.14camonzhmm
09:47.18jorgitoAsterisk Dynamic Loader Starting:
09:47.18jorgito<PROTECTED>
09:47.18jorgitoFeb 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)
09:48.25pcmdo ldconfig
09:48.53vgsterdoes anyone know if busydetectmartin is much better than regular busydetect?
09:49.09pcmvgster: it was I believe
09:49.21pcmvgster: 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)
09:50.17vgsterok, i need to comment the original line out but do i need to remove the + from the BUSYDETECT+= #-DBUSYDETECT_MARTIN
09:50.34vgsterin order to use it?
09:50.35maruzcamonz: jaike : Notice: Configuration file is /etc/zaptel.conf
09:50.35maruzline 0: Unable to open master device '/dev/zap/ctl'
09:50.52*** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.190)
09:50.52vgsterhave you made any udev changes?
09:50.56jaikeudev
09:50.57maruzi have no zaptel hw, so i think there no problem out there
09:51.08maruzor not?
09:51.27pcmvgster: comment the outher one and uncomment the 'MARTIN' one by removeing '#'
09:51.30camonzthere shouldn't be, it worked really well for me
09:51.37maruzcamonz: ok :)
09:52.01camonzi'm a newb at * and linux too sorry i cannot be of much help :->
09:53.04pcmanyone need 4xe1 or 4xt1 card ? I need to sell a few
09:53.53vgsterpcm: thanks
09:54.03jaikemaruz: go through README.udev inside your zaptel folder
09:54.17pcmvgster: np
09:54.50Mavvieif it just me, or is the editline in asterisk configured as emacs instead of as vi?
09:54.56vgsternext question, :D how many channels/ how small a cache in order to use CONFIG_CALC_XLAW in zaptel?
09:55.11jaikedidnt need to do that though, since with Fedora you just do 'make config', which creates a zaptel init file
09:55.52vgsterfedora used to take a few seconds to make all the devs on my system
09:57.22jorgitoshit asterisk Makefile is a mess
09:57.44jaikei think it does need some cleanup
09:58.34jorgitono doesnt have uninstall
09:58.53jaikemaruz: if this is becoming too complex just for MOH, just use the mp3 that came with asterisk
09:58.58jaikeang mpg123
09:59.20jaikei on the other hand really needed ztdummy for meetme
09:59.23camonzjust delete everything in var/lib/asterisk , usr/lib/asterisk /etc/asterisk
09:59.32camonzand where you keep the executable
09:59.36trixterapp_conference doesnt require a timer
10:00.11jaikehmm...will read on that
10:00.21vgsterdo 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)
10:00.47camonzvgster: i didn't had to
10:00.51vgsterok
10:00.52jaikevgster: i didnt need to..default conf worked fine
10:00.53camonzjust modprobe ztdummy from bash
10:01.28*** join/#asterisk sch19 (n=sch19@adsl-9-107-161.mia.bellsouth.net)
10:02.15jaiketrixter: i dont think its with stable yet
10:03.55trixterdont think what is?
10:05.02maruzcamonz: yes sir, i did it 2 times, the first time i got error, teh 2nd it got looaded
10:05.06maruzloaded
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)
10:07.30Delvar~ping
10:07.31jbotpong
10:09.33*** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
10:09.45vgsterdoes 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)
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10:17.44*** join/#asterisk h3x (n=h3xor@64.192.116.16)
10:23.53areskigood morning everybody !
10:24.18pcmgood
10:24.24vgsterhello
10:27.49*** join/#asterisk apardo (n=apardo@87.218.45.71)
10:30.27*** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net)
10:30.45jorgitoi 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
10:30.45jorgitoFeb 22 11:24:40 WARNING[26892]: loader.c:499 load_modules: Loading module chan_modem.so failed!
10:33.38pcmchan_modem exports some functions that are NEEDED
10:33.44pcmyou shouldn't do noload for it
10:35.06jorgitowell the problem is that i specified INSTALL_PREFIX?=/usr/local/asterisk
10:35.43jorgitowhen i run /usr/local/asterisk/usr/sbin/asterisk -vc , it took configuration from /etc
10:35.48jorgito/etc/asterisk
10:36.44vgsteri didnt think chan_modem got built with 1.2
10:37.18pcmthen you need to maybe add /usr/local/asterisk
10:37.21pcmto /etc/ld.so.conf
10:37.24pcmand do ldconfig
10:38.18jorgitowell i am not missing lib but configuration file
10:40.36*** join/#asterisk fulgas (n=fulgas@82.102.2.254)
10:47.25markttCan someone confirm that I can what seems plausable with asterisk....
10:48.16markttI 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.
10:48.51markttSo I have a modest little network on a fixed IP where I have NAT running behind a firewall.
10:49.14markttTHere is a Cisco IP phone on this as well as softPhones.
10:49.34markttCan 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)
10:53.31mut<PROTECTED>
10:53.35mutwhat causes that?
10:53.45mutseems randomly all the bchannels restart
10:54.29jaikemarktt: should be possible, if youre able to sort out sip-nat problems
10:59.00camonzi'm gonna go to sleep
10:59.06*** join/#asterisk Vyeperman (n=Vye@ip68-8-174-154.sd.sd.cox.net)
10:59.09camonzcya
10:59.13maruzbye
11:04.18*** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net)
11:11.08FalleFolks, are there any irclogs for this channel on the internet somewhere?
11:11.43tzafrirFalle, yes
11:12.26tzafrirhttp://www.asteriskgeeks.com/
11:12.30Fallethe 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)
11:12.52tzafrirFalle, that's the link from the wiki
11:13.20Falletzafrir: 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)
11:25.39jwu<PROTECTED>
11:25.48iDunnoyes, now infact.
11:25.57iDunnoin 1 second, a second will have passed!
11:26.10jwuoh..
11:26.34jwuidunno... you speak to me?
11:29.11dpryo;P
11:29.12jwuidunno, can you help?
11:37.34*** join/#asterisk Samoied (n=Samoied@201.14.162.82)
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11:48.40RoyK~realtime
11:48.41jbotit has been said that realtime is http://www.voip-info.org/wiki-Asterisk+RealTime
11:48.42RoyK~docs
11:48.43jbotrumour 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
11:48.48RoyK~rtfm
11:48.48jbotit has been said that rtfm is Read The F*cking Manual (TM)
11:48.50RoyKoh
11:48.52RoyKhe quit :P
11:50.57RoyKhmmmmmmm
11:50.58RoyKexten => s,n,Set(DIALOPTS=${IF($[ "${CCMAXSECS}" = "-1" ]?90:90\,S(${CCMAXSECS}))})
11:51.02RoyKthat doesn't work....
11:51.59tuxinator_linuxguess jwu didn't want an answer ;-)
11:53.30*** part/#asterisk maruz (n=maumar@adsl-123-3.38-151.net24.it)
11:54.45RoyKcan someone please help me out with this? the above Set(DIALOPTS.... just sets DIALOPTS to 90
11:54.46RoyK<PROTECTED>
11:54.56RoyKeven though I've escaped the ,
11:57.11thazzajaike: Bugger thats a few hours of recorded phone calls.
11:57.24jaikethats in wav49
12:02.11Modcutswith 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)
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12:24.45enemy^xexten => t,1,Queue(teknisk|tT|||0)
12:24.54enemy^xhow can this actually go further to t,2,....
12:25.03enemy^xwhen the timeout is set to 0?
12:27.57jaike100gigs to go tomorow..am outta here
12:28.01jaikebye all
12:28.05*** part/#asterisk jaike (n=a@203.131.137.76)
12:29.46RoyKenemy^x: #asterisk-no
12:31.09RoyKhmmmmmm
12:31.20RoyKseems 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)
12:36.19remissw00t.. ata-box in da house :D
12:43.10*** join/#asterisk fugitivo (n=ajf@201.255.177.92)
12:43.15fugitivohello
12:49.43saftsackFeb 21 21:56:24 debian kernel: Power alarm on module 2, resetting!
12:49.49saftsackwhat is that?
12:49.51*** join/#asterisk _Paulo_ (n=Paulo@200-168-112-132.dsl.telesp.net.br)
12:50.06fugitivosome problem with your power supply
12:50.51saftsackmy asterisk server said this
12:50.54*** join/#asterisk }btorch{ (n=kvirc@208.63.19.184)
12:51.03saftsackFeb 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 *
12:51.18saftsackthis 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 ?
12:52.25fugitivo}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
12:53.48x86heh, i managed to make the voicemail extension crash asterisk when called :)
12:53.49}btorch{calls to the outside ... any howtos out there ?
12:54.41fugitivo}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 !!!
12:56.23fugitivo}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)
12:57.12fugitivothanks
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
12:57.57fugitivowhat do you get when calling from the pbx to asterisk?
12:57.58*** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net)
12:57.59fugitivobusy tone?
12:58.37luke-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)
12:59.34fugitivo}btorch{: so it never reaches asterisk?
12:59.36luke-jr_* is a pbx
12:59.39}btorch{no
12:59.42luke-jr_;)
12:59.57EgonisI 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
13:00.17luke-jr_Egonis: check the commandline options
13:00.22fugitivoluke-jr_: well, it's hard to talk that way, it's better * and pbx :)
13:00.32Egonisluke-jr_: i.e. asterisk --help?
13:00.37fugitivo}btorch{: what model of siemens?
13:00.39luke-jr_Egonis: sure
13:00.53Egonisluke-jr_: lol, okay.. ty!
13:01.10}btorch{150E officepro
13:01.14luke-jr_so has anyone here used AEL yet?
13:01.25RoyKerm
13:01.37RoyKanyone here tried the S(timeout) flag with dial?
13:03.06fugitivo}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
13:04.51fugitivowhy 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)
13:06.16fugitivothat 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
13:07.29Garak_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
13:07.34EgonisI have built asterisk w/ gtk, but cannot find how to open the gtk console
13:07.37EgonisI am using gentoo
13:07.55}btorch{anyway got go setup a sonicwall now
13:08.14saftsackfugitivo, 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)
13:09.00RoyKbummer
13:09.10RoyKS(timeo) flag surely doesn't work
13:09.53fugitivosaftsack: maybe, didn't see that message in my whole life (11 years of linux)
13:11.39RoyKit's bill gates fault, all of it
13:11.41saftsackok. 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)
13:13.31f7950qs0hi
13:13.34f7950qs0anybody talking
13:13.38f7950qs0yea yea I know it would be best to ask
13:13.44f7950qs0I am using A@H
13:13.56f7950qs0and have configured a trunk a route and an extension successfully
13:14.14fugitivo~amp
13:14.14jbotextra, extra, read all about it, amp is NOT supported here! people using it should join #amportal
13:14.34f7950qs0what is amportal?
13:14.48fugitivothe web interface of a@h
13:15.10f7950qs0oh yea I forgot
13:15.11f7950qs0hehehe
13:16.02stseHi, I need some help with Asterisk 1.2.4 (Bristuff) and Snome phones (320/360).
13:16.33fugitivostse: what's the problem?
13:16.35stseI can't get the LEDs to show the ringing/used state.
13:17.16stsesip show subscriptions shows the subscriptions, show hints shows the correct hints/watchers, but asterisk doesn't send notify messages.
13:17.29TheCopsstse, It was working before and not anymore with the new 1.2 ?
13:17.44TheCopsI that problem here
13:17.49*** join/#asterisk backblue (n=igor@87-196-36-85.net.novis.pt)
13:17.51TheCopsI have that problem here
13:18.11stseTheCops: It never worked with 1.2.x. I didn't have 1.0.x
13:18.14backbluehi, ppl, why use tdm and not iax2 trunk?? which its better?
13:20.51*** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it)
13:21.12Garak_How do I get asterisk to make a call and play a recording when someone answers it
13:21.53remissanswer(), playback(recording)
13:22.04stseIf I understand correctly, Asterisk should send notify messages to the subscriber, but if I trace with ethereal, asterisk doesn't send any messages.
13:22.16Garak_remiss: yea but where, its not an exten...
13:22.30remissGarak_: it is...
13:22.41remissoh.. make a call
13:22.51remissstill.. create a context with it
13:23.14remisshmm.. good question..
13:23.39Garak_their is no onanswer =>
13:23.43*** join/#asterisk Eitch (n=hugo@unaffiliated/eitch)
13:24.43Garak_Has anyone ever implemnted a collect call system with asterisk?
13:25.11f7950qs0what is the best way to learn dialling rules in asterisk?
13:25.24remissLIMIT_CONNECT_FILE=filename
13:25.25remissSpecifies which file to play when call begins
13:25.27tuxinator_linuxHas anyone figured out the appeal to the Olympic game, curling?
13:25.36remissas an option to dial
13:26.21Modcutstuxinator_linux:to entertain old people
13:26.41f7950qs0I have the asterisk book
13:26.56Garak_tuxinator_linux: try playing it, then you will understand
13:26.56fugitivof7950qs0: editing the files and not using web interfaces
13:26.59f7950qs0I need a good reference to asterisk
13:27.23fugitivo~docs
13:27.24jbotwell, 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
13:27.54RoyK~wiki
13:28.07RoyK~wiki asterisk
13:28.59f7950qs0thanks jobt
13:29.01f7950qs0jbot
13:29.28EgonisI'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)
13:34.25Modcutshow 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)
13:35.10fugitivoModcuts: DIALSTATUS
13:35.30Modcutsah i see and how do i set that?
13:35.49fugitivowhen you dial you have the status of the call in ${DIALSTATUS}
13:36.07vgsterEgnois 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)
13:38.20Modcutsfugitivo: 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?
13:39.22fugitivoModcuts: what do you need to do?
13:43.13*** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl)
13:44.26f7950qs0bye all
13:44.35f7950qs0i 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)
13:58.55Modcutsfugitivo: 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.?
14:01.17Garak_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
14:02.03RoyKregister =>
14:02.47Garak_so that keeps a connection open?
14:03.32stseNo one here with a hint for my SNOM/Asterisk/Notify problem?
14:06.04*** join/#asterisk asteriskmonkey (n=phil@69.156.197.242)
14:11.31shmaltzwhy 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)
14:16.31luke-jr_Is there any documentation/howto for moving an Asterisk server to a different system?
14:16.49luke-jr_eg, obviously move the config files and voicemail directories... what else?
14:16.52mitchelocjust copy your config and voicemail/sound files
14:17.06mitchelocmaybe your cdr records (cdr-csv) if you want that
14:17.10luke-jr_ah
14:18.15shmaltzluke-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
14:18.40shmaltzluke-jr, if you did any mass provisioning using http/ftp/tftp then move those as well
14:18.45mitchelocheh what do you need the leases for? i can never find that file when i want to move to another server...
14:19.05mitcheloctftp i've forgotten once or twice ;), don't forget that! i had to manually redo the directory *sigh*
14:19.15shmaltzmithceloc, that way your dhcpd doesnt give out the same ip again ;(
14:19.17luke-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)
14:19.32shmaltzluke-jr, automatic provisioning
14:19.39luke-jr_oh
14:19.48mitchelocit's more like for the infrastructure of your network, your phones specifically
14:20.11luke-jr_Can Asterisk handle IPv6, I wonder
14:20.12mitchelocallthough, shame on us using the asterisk server for dhcp, tftp, AND asterisk...
14:20.26shmaltzluke-jr, I don't think so
14:20.34shmaltzmitcheloc, why?
14:20.50shmaltzI do it all of the time, and apache, webmin as well
14:21.06mitcheloci know, it's just a major point of failure...
14:21.08shmaltzand sometimes mysql for CDR as well
14:21.18shmaltzmitcheloc, what is?
14:21.19mitchelocif the asterisk server dies, so do your workstations (if they lose their leases), etc, etc, theres a lot of reasons
14:21.28mitchelocdhcp on that machine
14:21.44shmaltzmitcheloc, what are you talking about, why would the asteirsk server die quicker than Windowz?
14:21.54mitcheloci never mentioned windows...
14:22.09shmaltzmitcheloc, the other option for dhcpd would be windowz
14:22.36mitchelocno, 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
14:22.49Nugget"the other".  How quaint.
14:23.00shmaltz<PROTECTED>
14:23.11*** join/#asterisk danzig (n=chatzill@130.226.169.177)
14:23.18shmaltzmitcheloc, and that supports vendor specific dhcp options?????????????????
14:23.26Nuggetuptime is irrelevant.  it's *downtime* that matters.
14:23.33mitcheloc06:23:24 up 369 days, 11:36,  1 user,  load average: 0.00, 0.00, 0.00
14:23.52shmaltzNugger, you are right, that box never had downtime without *me* bringing it down
14:24.11mitchelocthats not the point shmaltz, if it goes down, you are screwed
14:24.17mitchelocyou have more work and more services to bring back online
14:24.30asteriskmonkeyshmaltz: your probably getting clicks cause your audio is cliping
14:24.31shmaltzmitcheloc, but where doesn asterisk come into this "if it goes down..."
14:24.37mitchelocspecifically workstations can be affected without dhcp...so your users don't have any computers to use!
14:24.45shmaltzasteriskmonkey, meaning?
14:24.48*** join/#asterisk flujan (n=flujan@internet.nube.com.br)
14:25.05flujanhi all.
14:25.07mitchelocshmaltz, 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)
14:25.16shmaltzmitcheloc, what you say has *nothing* to do with asterisk on the same machine
14:25.17flujanI 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...
14:25.17flujanflujan Can I connect a E1 interface from the analogic server to a E1 interface from the asterisk server?
14:25.27asteriskmonkeyshmaltz: meaning your hitting the audio threshold.. use ztmonitor on the channel you should see it peak when you hear a click
14:25.33Nuggetactually it has everything to do with it.
14:25.35asteriskmonkeythey you can adjust it accordingly
14:25.49mitchelocshmaltz, no i'm not blaming asterisk, i'm just worried about too many services on one machine
14:25.51austinnichols101flujan: yes, with a crossover cable
14:25.59shmaltzasteriskmonkey, how do I adjust it? tx/rxgains in zapata.conf?
14:26.09shmaltzmitcheloc, you are over worried
14:26.18Nuggetor over experienced.
14:26.19asteriskmonkeyyes
14:26.23shmaltzlol
14:26.30mitcheloc;)
14:26.31shmaltzasteriskmonkey, thanks I'll monitor it
14:26.36asteriskmonkeyremember you have to restart asterisk after you make the changes though or load/unload zap modules
14:26.46adaro1I'm having a problem trying to overflow calls from one queue to another
14:26.50mitchelocwell when you have a server failure, and an entire business offline, you'll understand what an additional peace of mind can mean for you
14:26.50asteriskmonkeynp its in /usr/src/zaptel-xx/
14:27.18adaro1Queue1  has only one extension    when that is busy  I want the call to go to Queue2 which has 2 extensions
14:27.41adaro1I'm getting   please try again later  instead of the overflow
14:27.46mitcheloci 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...
14:28.13shmaltzasteriskmonkey, why am I getting this:
14:28.14asteriskmonkeyive had one up for 8months now running asterisk and centos :D
14:28.15shmaltzUnable to open /dev/dsp: No such device or address
14:28.16shmaltzCannot open audio ...
14:28.24adaro1any suggestions on the best setup for this?
14:28.47*** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac)
14:28.48asteriskmonkeyshmaltz: it cant find a device.. if youve recently recompiled zap you have to re modprobe
14:29.12shmaltzasteriskmonkey, this is running right now
14:29.21asteriskmonkeyodd then
14:29.33asteriskmonkeydont know would have to poke around the box
14:29.51asteriskmonkeygoogle for some possible things to look for :)
14:29.56shmaltzasteriskmonkey, it's a bug, I did ztmonitor 9 -vvvvvvvvvvvvvvv
14:30.01shmaltzso it gave me that error
14:30.09shmaltzif I do:
14:30.11shmaltzztmonitor 09 -vv
14:30.12shmaltzits ok
14:30.18adaro1IS it possible to overflow one queue to another
14:30.23*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:31.16shmaltzadaro1, based on what conditions?
14:32.06adaro1shmaltz  -   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
14:33.37shmaltzadaro1, so I guess that if you do:
14:33.39shmaltzexten => dial(extenthatsbusy,1)
14:33.41shmaltzexten => jumphere if busy and use second queue
14:33.42shmaltzexten => if not busy call queue
14:33.44shmaltzthis should sure work
14:34.02sergeus~seen jvu
14:34.17jbotsergeus: i haven't seen 'jvu'
14:34.20adaro1I'll give it a go  -  thanks
14:34.38*** join/#asterisk oej (n=oej@fuckoff012231-1-3.clients.easynet.fr)
14:34.46sergeus~seen jwu
14:34.48jbotjwu <n=jwu@221.221.11.37> was last seen on IRC in channel #asterisk, 3h 5m 36s ago, saying: 'idunno, can you help?'.
14:36.32shmaltzlooks like dells ftp server is down
14:39.26*** join/#asterisk ManxPower (n=ewieling@stirprop-S4-0-0-21.ndcr2.datasync.net)
14:39.53Garak_what dose facility not subscribed mean in idefisk
14:41.03*** join/#asterisk iCEBrkr (n=icebrkr@6532244hfc169.tampabay.res.rr.com)
14:41.08iCEBrkrwerd!
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)
14:43.06BlackthornHey 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?
14:43.48BlackthornThis town wireless is what we run our voip services on and is connected to our * box. Voip cut-offs :(
14:44.08need_sccp_helpAnybody familiar with sccp setup in asterisk for cisco 12sp phones?
14:44.16mitchelocblack, out of curiousity what town?
14:46.32BlackthornMarion VA
14:48.47lunaphyte_what is 'n' in 's,n,Wait,2' ?
14:49.39need_sccp_help"n" means next priority
14:50.24need_sccp_helpInstead of ordering priorities 1,2,3,4, etc.. . You can order then 1n,n,n,n, etc. . .
14:50.54lunaphyte_simply meaning next in the list?
14:50.58need_sccp_helpexactly
14:51.00RoyKyeah
14:51.08greendiseaseluckyduck: 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)
14:51.22need_sccp_helpMakes life easier
14:51.53lunaphyte_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...?
14:52.03need_sccp_helpcorrect
14:52.08lunaphyte_thanks.
14:52.35need_sccp_helpBut remember that the first priority still must be numbered '1'
14:52.45lunaphyte_was just about to ask that.  ;)
14:52.50need_sccp_helpAfter that you can use 'n'
14:52.54*** part/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)
14:54.26lunaphyte_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)
14:59.30backblueanyone works with tdm?
15:00.02asteriskNewb1eI have a question about Asterisk@home with a X100P card for POTS.  Anyone familiar with that setup?
15:00.10Garak_backblue: I've done alittle bit
15:00.20*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
15:00.42backblueGarak_: why use tdm, if you can use a iax2 trunk between 2 asterisk servers?
15:00.44asteriskNewb1eI 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)
15:01.02iCEBrkrasteriskNewb1e:
15:01.03iCEBrkr~amp
15:01.04jbotmethinks amp is NOT supported here! people using it should join #amportal
15:01.12asteriskNewb1ethanks
15:01.35Garak_backblue: TDM is for connecting to a telco or connecting to channel banks
15:02.07asteriskmonkeyshmaltz: ztmonitor is not the same as asterisk :) it has a 2 v limit 1 shows the bar graph 2 shows with numeric indicators
15:02.39backblueGarak_: can we talk a litle in pvt?
15:03.18RoyK~wtf is amp
15:03.26RoyK~wtf  amp
15:03.44GerbilWrkwould 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?
15:03.45shmaltzasteriskmonkey, I know, but giving me that error when I enter -vvvvvvvvvvvvv is a bug
15:03.46asteriskNewb1easterisk management portal
15:04.40*** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net)
15:07.06asteriskmonkeyshmatlz: yes good bug observation :D
15:07.52asteriskmonkeyanyone 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
15:08.17shmaltzasteriskmonkey, what firewall is it?
15:08.23asteriskmonkeysonicwall
15:08.24brad_msswasteriskmonkey: what v of asterisk?
15:08.28fugitivoasteriskmonkey: sounds like a codec problem
15:08.34asteriskmonkey1.2 as of yesterday
15:08.44*** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu)
15:08.44brad_mssw1.2.????
15:08.49asteriskmonkeyah ok :) ill go force ulaw to be safe
15:08.57brad_msswor do you mean as of svn yesterday ?
15:09.06fugitivoif it rings, it should work (with iax)
15:09.10brad_msswyeah, ulaw would be a good choice
15:09.14*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
15:09.22asteriskmonkeybrad_mssw .. i dont do the 1.2.??? i compile by revision i think im like SVN-branch-1.2-r10368
15:09.55asteriskmonkeyi always use ulaw its nice but so damn heavy on bandwidth
15:10.01brad_msswasteriskmonkey: ah, ok ... i usually don't do svn for production stuff
15:10.07tdonahuewhat causes "Stopping retransmission on '568a0d65068cb83838de831b1542f89b@208.51.101.194' of Request 102: Match Found
15:10.08brad_mssw(and I really only do production stuff)
15:10.16tdonahue" 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)
15:12.17asteriskmonkeybrad_mssw: thats branches not trunk branches is stable trunk is not
15:12.38fugitivowell, a release is not necessary "stable" ;)
15:12.54asteriskmonkeyis anything made be the open source considered to be stable :)
15:13.01fugitivoapache
15:13.05fugitivolinux
15:13.20fugitivosendmail
15:13.20asteriskmonkeydude id say they have some serious amount of years on them compared to asterisk
15:13.28asteriskmonkeywell FreeBSD for the win
15:13.35fugitivowell, you asked for opensource stable :)
15:14.22asteriskmonkeylol sorry was asking for grief when i said that :)
15:14.36asteriskmonkeymmm ... odd it dosnt seem to like gsm..
15:15.02fugitivodid it work with ulaw?
15:15.05RoyKasterisk will prolly stabilise one day....
15:15.17asteriskmonkeyswitching to ulaw atm
15:15.20asteriskmonkeyill tell you in a sec
15:15.25RoyKwhy ulaw?
15:15.36asteriskmonkeyits nackedest codec you can use
15:16.07RoyKµlaw is american evilness :)
15:16.27tdonahueno one knows what causes the "Stopping retransmission" message?
15:16.34asteriskmonkeyok so ulaw dosnt work :(
15:16.37asteriskmonkeyok so its not codec
15:16.56fugitivoweird
15:17.05asteriskmonkeyhere is my tree so to say  cubixsoftclient(iax)=>asterisk=>pstn
15:17.18asteriskmonkeyi can call numbers fine just incomming numbers get rapped and dropped
15:17.25RoyKasteriskmonkey: 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)
15:18.14asteriskmonkeyRoyK: there is nothing man in the verbose log that says anything why its ditching the call
15:18.22litecodeI'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
15:18.37asteriskmonkeyits got to be a fireweall thing
15:18.56asteriskmonkeybut i though iax was impervious to firewall .. (this works everywhere else) just not at this office
15:19.14zoneoutHow do I write a GotoIf() which matches against the number, is there a way to do this making use of the dialplan _XXX format?
15:19.42mikefooasteriskmonkey: iax is only firewall friendly because its in/out ports stay the same.
15:19.42asteriskmonkeyzoneout : yes
15:20.13asteriskmonkeymikefoo: so if the ports are no longer 5060 then i can assume the firewall is blocking them
15:20.28asteriskmonkeysorry 4569
15:20.29*** join/#asterisk Delvar (n=irc@host-83-146-53-46.bulldogdsl.com)
15:20.55mikefooiax is on 5060?
15:20.58mikefooahh ok yah..
15:21.03asteriskmonkeyfor some reason my iax connection is on port 47925 on the server.. so does this mean the firewall is blocking
15:21.21mikefooasterisk: is it behind a nat?
15:21.22asteriskmonkeysorry iax is support to be 4569.. i quoted you a sip port
15:21.26asteriskmonkeyno
15:21.31asteriskmonkeyasterisk is on a public ip
15:21.36zoneoutasteriskmonkey: how? :)
15:21.56mikefooSo.. disable firewall and test?
15:22.18asteriskmonkeydude :) ist a corporate firewall there is non of those disable options available to me
15:22.34asteriskmonkeyso ill be happy if its just it dont work for this reason sorta answer :D
15:22.50mikefoohahah
15:23.34asteriskmonkeybut 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
15:24.33asteriskmonkeyah wait
15:24.45asteriskmonkeyi notice there is another iax connection from this location on that port
15:24.46asteriskmonkey:P
15:24.49asteriskmonkeygrr
15:25.06asteriskmonkeyso is this an iax port related problem
15:25.06*** join/#asterisk Dr-Linux (n=Nothing@202.125.141.8)
15:25.27Dr-Linuxi forgot my sipura-2100 amdin password any idea?
15:25.33RoyKmikefoo: 5060 is sip
15:26.06zoneoutHow 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)
15:28.41asteriskmonkeyzoneout: if you google for zapteller youll see a nice example on the voip wiki with the code structure you require
15:29.58mikefooRoyK: yes I know..
15:30.04remisswhat is the best way to call asterisk from an outside script?
15:30.10mikefooi was responding to asteriskmonkey
15:30.23remissor java
15:30.24mikefooHey, can anyone suggest a toll free did provider?
15:30.53iCEBrkrmikefoo: Good luck with that
15:31.00mikefoolol..
15:31.06iCEBrkrmikefoo: But asterlink.com has really cheap 800 DID's
15:31.18stormfrhello, is there any dsp card compatible with asterisk for G729 or G723 ?
15:31.20iCEBrkrmikefoo: $1.95/mo at like .02/min
15:31.25mikefooyeah?  niice
15:31.35zoneoutasteriskmonkey: cheers
15:31.37mikefoohow are the for an inbound provider?
15:31.44ManxPowerAnyone 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)
15:32.00iCEBrkrmikefoo: I haven't used them exclusively yet.  I just got the account.
15:32.03Darwin35DR Linux Call Sipura
15:32.13mikefoocool..
15:32.22ManxPowerzoneout, exten => _XXXX,1,Goto(context,extension,priority)
15:32.29litecodewell, it looks like 1.2.4 is making PaX angry.
15:32.30Darwin35or open it up and find the reset button
15:32.55ManxPowerstormfr, NO.
15:32.57iCEBrkrmikefoo: 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)
15:34.15RoyKManxPower: exten => _X.,1,Goto([[context,]extension,]priority)?
15:34.59ManxPowerRoyK, no, that would match too much.  He only wanted to match exactly 4 digits
15:34.59mikefooiCEBrkr: thanks  :)
15:35.02RoyKManxPower: i just though about the goto syntax.....
15:35.22ManxPowerRoyK, It's easier to tell people to always use all three
15:38.05asteriskmonkeyha thats odd sip with stun dosnt work but sip without stun does behind the sonicwall :P
15:38.10stormfrhello, 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)
15:38.29mikefooasteriskmonkey: oh its a sonicwall?
15:38.46asteriskmonkeyyes
15:39.04mikefoothey have tons of features of packet analysis, which interfers with alot of crap, I would do any xml requests with those turned on.
15:39.21mikefooI couldnt*
15:39.49asteriskmonkeywhacky good to know though :) so anymore issue like that i can point at the sonicwall :D
15:40.13mikefoohah, yeah
15:41.10lunaphyte_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-Fenderlunaphyte : Lookup "DISA" on the WIKI
15:42.09*** join/#asterisk SplasPood (n=jwb@206.252.198.100)
15:42.22dezenthm strange.. when i compile zaptel i get this error  "/bin/sh: line 1: scripts/mod/modpost: No such file or directory"
15:42.29lunaphyte_[TK]D-Fender: thanks.
15:42.31*** join/#asterisk firestrm (n=no@S01060013105c1c69.gv.shawcable.net)
15:42.37dezentanyonek know what could be wrong ?
15:44.57*** join/#asterisk Nebukadneza (n=daddel9@i3ED6E1A6.versanet.de)
15:45.09Nebukadnezahihi
15:45.48asteriskmonkeyDISA the bomb :D
15:46.18asteriskmonkeyi 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)
15:50.16saftsackis it possible to run a pri card and 30 - 60 VOIP telephones on one computer?
15:52.01asteriskmonkeyis it possible to fit 12 people into a van
15:52.06asteriskmonkeydepends on the pc man :D
15:52.20asteriskmonkeyif its a p4 no worries
15:53.36asteriskmonkeyi can get 40 ulaw calls on a linksys 54g router running linux
15:53.57asteriskmonkeyso 60 calls shouldnt be an issue unless you want to do something mega funky
15:54.22saftsack^^
15:54.32saftsackok
15:54.37Delvarin some recent testing on a dell dule xeon 1gb ram i had over 800 channels with no transcodeing
15:54.41asteriskmonkeyanyone going to VON this year in toronto?
15:55.05Delvari got about 300 channels when transcoding
15:55.11asteriskmonkeyyes a dual 3 gig xeons craps at around 100channels tops g729 coding
15:55.23Delvardidnt try g729
15:55.29asteriskmonkeyDelvar: what gsm?
15:55.37Delvaryes
15:55.38saftsackDelvar, wtf? do you have 800 telephones or were that simulated channels?
15:55.43Delvarnooo
15:55.50Delvari had 5 asterisk boxes
15:56.03saftsack:)
15:56.04Delvar1,2 started callls, though 3 to 4 and 5
15:56.06docelm0dezent, YOU BROKE IT!
15:56.21docelm0Delvar, I have 12 in my cluster right now
15:56.23lunaphyte_so if i want ext 7508 to be for disa access, is something like this appropriate? exten => 7508,1,DISA(1234|home) ?
15:56.25Delvarso not a real world test by any means
15:56.46dezentdocelm0: SHIY !!!
15:56.49Delvarcool
15:56.56dezentdocelm0: that sux :/
15:57.11docelm0My real world test is 800 calls so far  :)
15:58.39iCEBrkr1 MILLLLLLLLLLLLION calls! </doctor evil>
15:58.50RoyKnow many concurrent calls?
16:00.16docelm0Im figuring around 5000 top's..
16:00.26docelm03000 w/ transcoding
16:01.19asteriskmonkeydude if you have cards with dsps your pci slots are the limit :D
16:02.53remisssuggestions on how to make asterisk do something from an external application?
16:03.06*** join/#asterisk redondos (n=redondos@190.48.41.62)
16:03.14asteriskmonkeyremiss - AGI's
16:03.15remisse.g. if i want to call santa claus from a java-application..
16:03.28asteriskmonkeyagi agi agi
16:03.34remissasteriskmonkey: isn't that the other way? O_o
16:03.45*** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com)
16:03.55asteriskmonkeysos
16:04.02remisse.g. asterisk calls up the agi-script?
16:04.17asteriskmonkeyjust make a php script or java scrpt that dumps calls into files for the asterisk manager to dial
16:04.24remissahhhhhh
16:04.27remissof course :D
16:04.37docelm0hay iCEBrkr I had 250 calls up last night at once..  its kinda impressive..   :p
16:04.42asteriskmonkeythen in does dumpfiles specify a context in which you want it to execute and have your funckyness there :D
16:05.11remissyou can send me flowers, but i won't arrange them
16:05.15EgonisI 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
16:05.24Egonisit's via SIP
16:05.25*** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com)
16:05.43remissEgonis: if you want ppl to help you, you have to provide more information
16:06.01remisse.g. nat or not
16:06.07*** join/#asterisk salviadud (n=ralfalfa@dsl-201-129-72-144.prod-infinitum.com.mx)
16:06.34mikefooremiss: dude, give me santa's number
16:07.08asteriskmonkeyhere is the most uberist asterisk box http://www.williamsglobal.com/fuseworxv2000.jpg  12terabytes storage space 32gig ram 4 cpu :D
16:07.36asteriskmonkeyhehehe
16:07.39mikefooI have a bone to pick with him from 1993, i got the fak MyBuddy and was mad fun of
16:07.44remissmikefoo: i'm not gonna give you my mommas number :-/
16:07.44Egonisremiss: Which information would be useful?
16:08.28[TK]D-Fenderasteriskmonkey : Holy McFuck.....
16:08.41[TK]D-Fenderasteriskmonkey : that box is just... WRONG....
16:08.53asteriskmonkeythats the beast i be building now :)
16:09.11[TK]D-Fenderasteriskmonkey : And that SuperMicro case is FUGLY.
16:09.22[TK]D-FenderWho needs that much for a telephony server?
16:09.25asteriskmonkeyits for 8 ds3's
16:09.37asteriskmonkeybanks
16:09.50[TK]D-Fenderasteriskmonkey : can it ration off the CPUS to handle it properly?
16:09.56zoneoutin 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??
16:09.56EgonisWhen 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
16:10.00asteriskmonkeyyes
16:10.03[TK]D-FenderDS3 *data* I take it, not voice?
16:10.06Egonisbut I can't hear the person I am calling
16:10.09asteriskmonkeyit can even run in multi jail mode
16:10.12fugitivoTHAT'S "HEAVY"
16:10.19asteriskmonkeyvoice :D
16:10.27[TK]D-Fenderasteriskmonkey : What card?
16:10.30asteriskmonkeygoing to be using next gen d3 cards with g729 dsps
16:10.34asteriskmonkeyhehehe
16:10.41asteriskmonkeythe new ones comming like i said :D
16:10.46[TK]D-Fenderahhh.. the "unmentionable" ones :)
16:10.48dasuberdavidwho makes it?
16:10.57asteriskmonkeythe box?
16:11.01*** part/#asterisk redondos (n=redondos@190.48.41.62)
16:11.01dasuberdavidthe card
16:11.21asteriskmonkeycant tell you cause there in distro negotiations atm
16:11.30dasuberdavidah
16:11.49fugitivosangoma
16:11.54asteriskmonkeylol no
16:12.00fugitivomicrosoft
16:12.03salviadudis a red alarm on my zap channels a bad thing?
16:12.13fugitivosalviadud: no, green is a bad thing
16:12.34asteriskmonkeysalviadud: no it usually means something is misconfigured or not plugged in
16:12.50asteriskmonkeyit menas its going to blow up
16:12.57salviaduddamn
16:13.06asteriskmonkeyif it blinks slow at first then starts blinking faster..run
16:13.35salviadudwell, im using an old config
16:13.38salviadudfrom asterisk 1.06
16:13.45salviadudhas zaptel changed that much?
16:14.01mzowtf, still getting rejected from FWD?  FWD do es not love me =( *cry*
16:14.07fugitivoyou upgraded and now you get red light?
16:14.15salviadudyeah
16:14.20fugitivoweird
16:14.32fugitivowhat card?
16:14.39salviadudgeneric clone
16:14.42salviadudtp100
16:14.48fugitivois the module loaded?
16:14.53salviadudyeah
16:14.59salviadudzaptel, and wcfxo
16:15.00fugitivoztcfg
16:15.14fugitivoare the channels actually available?
16:15.21salviadudyeah, channel 01 and 01
16:15.23salviadudi mean
16:15.26salviadud01 and 02
16:15.33salviadudi got 2 of those cards
16:15.40salviadudfxs kewlstart
16:16.12mzohttp://pastebin.com/566905 ;p
16:16.31fugitivoare the cables ok? do you have dialtone?
16:16.51[TK]D-Fenderahhh.. the "unmentionable" ones :)
16:17.09salviadudno, the cables are not plugged
16:17.15fugitivowell
16:17.28fugitivoplug them
16:17.37salviaduderrrrrr
16:17.56salviadudcan't get a descent place to test this damn thing
16:17.58fugitivo[TK]D-Fender: the chinesse ones?
16:18.04zoneoutin 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??
16:18.22salviadudjust one more question
16:18.26salviadudin signalling
16:18.37salviadudif its fxs
16:18.42*** join/#asterisk sch19 (n=sch19@adsl-223-232-80.mia.bellsouth.net)
16:19.01salviadudshould it go signalling=fxsks or signalling=fxs_ks?
16:19.16fugitivofxs_ks
16:19.31fugitivofxsks=channel in zaptel.conf
16:19.34salviadudeven for zaptel.conf?
16:19.37fugitivosignalling=fxs_ks in zapata.conf
16:19.49fugitivo^^
16:19.50*** join/#asterisk feist (n=feist@nat-pool-msp.redhat.com)
16:20.00salviadudthanx man
16:20.42*** join/#asterisk Skwid_ (n=Skwid___@bas1-montreal42-1177928372.dsl.bell.ca)
16:22.28Kattyhi lads.
16:22.32*** join/#asterisk lobstu (n=mippy@dsl-146-240-254.telkomadsl.co.za)
16:22.37lobstujoin #rubyonrails
16:22.38lobstu:P
16:22.38*** part/#asterisk lobstu (n=mippy@dsl-146-240-254.telkomadsl.co.za)
16:22.38Kattyi've come to collect hugs.
16:22.45Kattyand mushy food.
16:22.52*** join/#asterisk saftsack (n=saftsack@p54A7DAAD.dip.t-dialin.net)
16:22.53ChrisUK>:P
16:22.53saftsackhi
16:23.06saftsackis the Elmeg IP290 a snom telephone?
16:23.10Kattyhi ChrisUK (=
16:23.12Kattymister fender!
16:23.28Kattyfugitivo: i'm /almost/ to a tolerable pain level without the pain killers :>
16:23.42Katty:<
16:23.54lunaphyte_mmm.  steak...
16:24.15[TK]D-FenderKatty: Can you send me some drugs by PayPal?
16:24.39Katty[TK]D-Fender: uhmm, no.
16:24.43lunaphyte_why does my disa dialtone go away after a second or so?
16:24.52Katty[TK]D-Fender: they told me to flush the narcotics i couldn't take.
16:25.02Katty13 pills of oxy-contin down the drain.
16:25.13*** part/#asterisk Nebukadneza (n=daddel9@i3ED6E1A6.versanet.de)
16:25.44Kattyfugitivo: odd you say that, because my surgeon told me to stop taking ibeuprofen and start taking extra strenght tylenol instead.
16:26.23mzoicky bad surgery :P
16:26.41fugitivoKatty: careful, don't drive when you take those pills
16:26.51Kattyfugitivo: oh?
16:26.59Kattyfugitivo: they don't make me sleepy
16:27.08fugitivoPM will make you sleep
16:27.13Kattyi'm not taking PM :)
16:27.18Kattyjust extra strength
16:27.18fugitivooh :)
16:27.32fugitivoi took PM once by mistake
16:27.33fugitivoat work
16:27.38FlyboySR22PM is gond during the PM, add a nice glass of wine and it really works !!
16:27.46KattyFlyboySR22: you're awful.
16:27.50FlyboySR22:-)
16:28.03Kattysomeday i'm going to drink wine.
16:28.06FlyboySR22We hope you feel better Katty
16:28.08FlyboySR22!!
16:28.12fugitivoKatty: you should
16:28.15fugitivowine is good
16:28.16Kattythanks, but i'm afraid it's going to take awhile.
16:28.22FlyboySR22:-(
16:28.24Kattyfugitivo: i'll get around to it eventually :)
16:28.27FlyboySR22That is no fun
16:28.31*** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com)
16:28.35Kattyhi justinu!
16:28.38fugitivotry argentinian red wine
16:28.49kpettitanyway to kill a call to a confrence room?
16:28.53Kattyi've had a few sips of wine...and the ones i like the most are super super sweet.
16:28.59Kattyalmost like nectar
16:29.06fugitivokpettit: unload app_meetme.so ?
16:29.08kpettitgetting weird artifacts in a meetme room, seems the only way to fix it is to reset asterisk
16:29.11Kattybicardi apple is pretty good
16:29.13justinuhey catty, how's the mouth
16:29.17justinukatty even
16:29.20Kattyjustinu: eh, it's healing.
16:29.20kpettitfugitivo, worth a try.
16:29.33justinubicardi makes wine?
16:29.38justinubacardi?
16:29.41Kattyjustinu: 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
16:29.42fugitivokpettit: tell me if that works :)
16:29.44Kattyjustinu: it's not a wine
16:29.49Kattyjustinu: more like a wine cooler maybe...
16:29.52justinuah, right
16:30.06justinuflavored beer ;)
16:30.12kpettitFeb 22 10:29:51 WARNING[7252]: loader.c:135 ast_unload_resource: Soft unload failed, 'app_meetme.so' has use count 3
16:30.17Kattyfugitivo: 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)
16:30.46kpettitfugitivo, know of anyway to force the unload?
16:30.52fugitivoKatty: well, some alcohol is evil, like some databases (mysql)
16:30.56FlyboySR22ah...my sister is JW...she drinks wine coolers :-)
16:31.06fugitivokpettit: hmm, no
16:31.21fugitivokpettit: if you enter the conf as admin and kick everybody?
16:31.22justinukatty: whats your association with witnesses?
16:31.23KattyFlyboySR22: yeah JWs say it's all good in moderation
16:31.28Kattyjustinu: i was raised as one
16:31.33Kattyjustinu: for..uhhm, around 18 years
16:31.34kpettithow can i kick everybody in the confrence
16:31.40*** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka)
16:31.40FlyboySR22Katty, which is smart by anoyone's standards :-)
16:31.46Kattyjustinu: i've spent my last two years away from home redoing my programming :)
16:31.47fugitivokpettit: no idea, i'd never did that
16:31.58fugitivobrb
16:31.58KattyFlyboySR22: indeed, but my mother is one of the extreme JWs.
16:32.08KattyFlyboySR22: and she leans towards all alchohol being bad.
16:32.09ManxPowergod hates me
16:32.15KattyFlyboySR22: so that's how i was raised.
16:32.35FlyboySR22Katty, Ah..understood..all of my brothers and sisters are JWs and some of them are ilke your mom
16:32.39justinukatty: interesting
16:32.51Kattyjustinu: yeah, it is pretty interesting.
16:32.57Kattyjustinu: the first couple months were major culture shock
16:33.12FlyboySR22Katty, 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!
16:33.16justinuthe witnesses often come to harass my hindu neighbors
16:33.51salviadudfugitivo
16:33.58salviadudi connected the line
16:34.03salviadudon alarms i get an OK
16:34.16salviadudi suppose it's cool
16:34.19salviadudis it?
16:34.42Hmmhesaysda da da dad da da daldsadadfasdfadsf;lksjfd;laskdjf
16:34.52justinui have a curious fascination with cults
16:34.55HmmhesaysI need a new domain name
16:35.03justinunot really sure if the witnesses are defined as a cult, but they share a lot of traits
16:35.09[TK]D-FenderI 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 :)
16:35.10mzoastersik is a cult
16:35.19KattyHmmhesays: you should come take care of me.
16:35.22Hmmhesaysvoipfister just isn't working out
16:35.36Hmmhesaysfeeling down?
16:35.40KattyHmmhesays: we can do theraputic shopping and such
16:35.42mzowhen JW's come to my door, i open the door wearing leather, and i ask them to join the orgy
16:35.45KattyHmmhesays: more like cut up
16:35.52KattyHmmhesays: and swollen....my face is still bruised.
16:35.54mzonot once do they ever stay, even when it's a cute girl =(
16:36.02kippiis there a user manual for the voicemail?
16:36.03Hmmhesaysawww that sucks
16:36.12FlyboySR22Katty, What happened..?
16:36.15KattyHmmhesays: i don't think 'sucks' starts to cover it :/
16:36.27KattyFlyboySR22: i had all four of my wisdom teeth cut out on friday.
16:36.35FlyboySR22Katty, OUCH !!!
16:36.42KattyFlyboySR22: yeah, it's defnately been ouch.
16:36.44*** part/#asterisk Skwid_ (n=Skwid___@bas1-montreal42-1177928372.dsl.bell.ca)
16:36.48FlyboySR22Katty, forget the wine, go right to burbon!!
16:36.52FlyboySR22:-)
16:37.00KattyFlyboySR22: the pain killer/narcotic they gave me for the pain....i was either allergic too, or it was too strong.
16:37.10KattyFlyboySR22: so..i couldn't take the pain killer.
16:37.11FlyboySR22Katty, so crap on top of crap
16:37.22KattyFlyboySR22: yeah, lots of crappy crap this weekend.
16:37.37FlyboySR22Katty, well hopefully you start to feel better very soon
16:38.04KattyFlyboySR22: i'm hoping so (=
16:38.18KattyFlyboySR22: currently at the stage where i've been on a liquid diet for almost a week, and my body is seriously complaining.
16:38.33KattyFlyboySR22: getting nauseas everytime i eat...really weak, etc.
16:38.36FlyboySR22Katty, wow - it is taking that long to heal...? Is that normal..?
16:38.43Hmmhesaysyes
16:38.46KattyFlyboySR22: well, it's healing......but...
16:38.50FlyboySR22Katty, its been so long since I had mine out I cannot remember
16:38.56KattyFlyboySR22: after that severe pain i had..i'm almost afraid to try to eat anything.
16:38.57Hmmhesaysit takes quite awhile
16:39.00KattyFlyboySR22: i don't want to be in pain.
16:39.06FlyboySR22Katty, I don't blame you
16:39.20justinui need help.... who has experience tuning EC on a single span T1 card (no DSP)
16:39.28justinui'm willing to pay if you're good
16:39.34*** join/#asterisk stse (n=stse@muedsl-82-207-237-090.citykom.de)
16:40.02Hmmhesayssounds kind of kinky
16:40.03Kattyjustinu: i'm experienced in hugging, does that count?
16:40.34justinuyes - but unfortunatly in this situation hugs aren't the suolution
16:40.36justinu:(
16:40.36Kattykthx.
16:40.37stseHi! 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)
16:40.43Kattyjustinu: :<
16:40.51Kattyjustinu: all you need is hugs, dobedobedo
16:40.57ManxPowerstse, how do you know they are being sent?
16:41.11zafhi, can anyone recommend a good softphone for Windows?
16:41.20ManxPowerAll softphones suck
16:41.21stseManxPower: they are not sent, I don't see them with ethereal.
16:41.25justinustse: make sure mailbox= is set in sip.conf
16:41.33ManxPowerstse, what is your mailbox= line in sip.conf?
16:41.39[TK]D-FenderI 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-Fenderdammit... stupid scroll-back errors..
16:41.50justinui just opened the door naked
16:41.52[TK]D-Fenderjsdhfhjlasjkdfhiu34y5ljk3hq45
16:42.02[TK]D-Fenderwhere's /clear when I need it...
16:42.42stseNo, I mean the notify messages to indicate that a watched channel is ringing/busy.
16:42.56justinuhints have to be set in the dialplan
16:42.57stsevoicemail is working.
16:43.19Katty[TK]D-Fender: or you could just /politely/ ask them to not return
16:43.35stsejustinu: there are hints, show hints and sip show subscriptions are showing the correct things (or so I think ;-).
16:43.40Katty[TK]D-Fender: never underestimate the power of being polite (=
16:43.44[TK]D-FenderKatty : That was a double paste, but it was COMEDY, not experience :)
16:43.44ManxPowerstse, Ah.  No idea.  That's a hint thing.
16:43.47lunaphyte_what is the difference between -v and -d ?
16:44.06[TK]D-FenderKatty : I am far too polite.... I've got to start cracking down on phone-spammers though...
16:44.26justinustse: i got that presence notification stuff working on my snom360
16:44.38idpromnutKatty: never underestimate the power of a 12gauge and a rocking chair on the porch :)
16:44.39justinujust so you know, it's possible
16:44.51stsejustinu: *sigh* others too, but I don't have success.
16:45.26justinu#if LINUX_VERSION_CODE >= KERNEL_VERSION(2,6,3)
16:45.27justinu#define HDLC_MAINTAINERS_ARE_MORE_STUPID_THAN_I_THOUGHT
16:45.27justinu#endif
16:45.29justinuheh
16:46.02Kattyidpromnut: i don't like guns.
16:46.06justinuECHO_CAN_KB1? MG2? MARK3?
16:46.12justinuwhat should I be using?
16:46.13*** join/#asterisk IOscanner (n=IOscanne@c-24-0-183-125.hsd1.tx.comcast.net)
16:46.14stsejustinu: 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. ;-(
16:47.08jbalcombanyone 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.
16:47.40justinujbalcomb: we know nothing about telrad, but a standard PRI config, with signalling set to pri_net should do it
16:47.47justinuand perhaps a T1 cross over cable
16:48.56saftsackhowto connect a pri card to the line? with a normal cat5 cable?
16:50.38saftsackis PRI supported by the zapteldrivers?
16:51.14Goralif 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)
16:51.24ManxPowersaftsack, no, it's supported by libpri and libpri uses zaptel
16:51.36ManxPowerGoral, um, there should be only 1
16:51.47saftsacksounds good :) does it run well?
16:51.51Bambri got following error
16:51.52BambrFeb 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
16:51.54saftsackor is it unstable like BRI?
16:51.59Bambrwhat's wrong? :)
16:52.03ManxPowerAny carrier would use PRI
16:52.04SkymarshalDoes a Global variable overrules a Channel variable or do they somehow coexist?
16:52.22ManxPowerBambr, you do not have the correct info in the [321] section of sip.conf
16:52.28Goralbambr i'm comming up with a few
16:52.46ManxPowerSkymarshal, the channel variable would override the global vatriable, I thingk
16:52.59Goralbut the first in line is Asterisk: The Future of Telephony is that it?
16:53.12SkymarshalManxPower: I think that too, but can I verify this somewhere?
16:53.13ManxPower~docs
16:53.14jbotit 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
16:53.24ManxPowerhttp://www.oreilly.com/catalog/asterisk
16:53.31BambrManxPower, actually i got like [term_default_321] section, and file name=321 there and also username=321
16:53.34ManxPowerSkymarshal, try it and see
16:53.37GoralTY ManxPower
16:53.41ManxPowerBambr, that's won't work
16:53.46ManxPoweryou need a [321]
16:53.50ManxPowername is not a valid option
16:54.11ManxPower[whater] is used for incoming auth, username=whatever is used for outgoing auth
16:56.02Bambrcan i add any variable to that section, just for my own use?
16:56.21Bambrlike myvar=myval ?
16:59.53*** join/#asterisk need_sccp_help (n=none@198.60.73.230)
17:00.01clyrradI would like to know that too, is it possible to make your own variables in that manner?
17:01.33clyrrador 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?
17:02.39xtrvdAll 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.
17:03.04xtrvdI'm not sure if that's too elementary of an answer, but I don't understand what else you are saying. Sorry.
17:03.37xtrvdoh, and those exist in the [globals] context.
17:04.27clyrradWhat 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
17:04.36clyrradso [sip_account]
17:04.41clyrradmyvar=foobar
17:04.55clyrradaccountcode=123
17:04.56clyrradetc...
17:05.04*** join/#asterisk trailhead26 (n=flombard@hou-nat129.novolink.net)
17:05.09xtrvdAhh, in the sip.conf... Wow, I wish I could tell you. I have no idea. =)
17:05.15xtrvdSorry.
17:05.18clyrradLOL
17:05.36clyrradI would love to know how to do that (if its even possible)
17:05.43HmmhesaysI so love it when gateways don't work right
17:06.34*** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca)
17:08.00stormfrhello, 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)
17:10.58djMaxis it possible to simultaneously ring two extensions but with different CID?
17:11.09FuriousGeorgeis "exterhost" only for sip?  the wiki page is down.  * caches my ip with my dynamiv *.dyndns.org addresses
17:11.40FuriousGeorgevery annoying, must be some way to change
17:12.08*** join/#asterisk Defraz (n=t0tal@tim.mychoice.cc)
17:12.12remissmoahahha.. 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)
17:12.38fu3hi
17:13.32fu3I 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.
17:13.43fu3I have loopstart signalling configured and still, no dice.
17:13.57fu3I understand loopstart sends a busy signal down the line to indicate a hangup, correct?
17:14.14fu3When a hangup occurs, there is no voltage drop, no polarity shift, no busy signal, etc.. just dialtone.
17:14.20justinufu3: try kewlstart
17:14.22fu3Any ideas?  Does that even make senese?
17:14.25fu3I've tried it
17:14.37Skid*whooooooosh*, damn did you see that? right over my head
17:14.38justinuit's uncommon
17:14.40Skid:-)
17:14.41*** join/#asterisk dca[laptop] (n=dca[lapt@sta-208-139-193-162.rockynet.com)
17:14.46fu3I thought ks was just a modified version of groundstart?
17:15.20brad_msswdca[laptop]: rockynet, eh ... you don't happen to be with teliax, do you?
17:15.49tronixfu3: ks = loopstart with far end remote disconnect supervision
17:15.51justinuks is loopstart with the battery drop detection, iirc
17:15.55*** join/#asterisk tholo (n=tholo@nat.sigmasoft.com)
17:16.06tholoWhat's the secret to make NMI problems disappear with TE410P...?
17:16.09fu3ahhh
17:16.32fu3so I should see a battery drop on my multimeter when a hangup occurs correct?
17:16.36tholoI have some new servers, and when I load the zaptel drivers I get that, then the kernel goes bye-bye...
17:16.57tholoRegardless of which slot I put the card in etc...
17:17.11fu3As 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.
17:17.42tronixtholo: good start would be to disable everything you can live without... usb ports, serial, irda, parallel port, etc. in BIOS
17:17.57fu3the asterisk box play's its introduction message right over the dialtone.
17:18.12justinufu3: that's not good
17:18.14tholoAlready done that.  Not that the IBM servers gives you much to change in the first place...
17:18.15fu3I know :)
17:18.22justinusounds like you're connected to a funky switch
17:18.27justinuthat doesn't do the battery drop
17:18.37fu3I've got the phone guys coming back out later.. so I hope to have something to ask/tell them.
17:18.54fu3justinu.. sounds like it to me too.  Like I said, I dont see any voltage fluxuation upon hangup.
17:19.12tronixfu3: 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
17:19.19fu3you got it!
17:19.21tronixsweet, thanks
17:19.41tronixtholo: 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
17:19.55jbalcombjustinu: ok, thanks. I guess I'll know tomorrow morning.
17:20.02justinuyou mentioned qwest...ask them if you're connected to a DMS100, or a 5ESS
17:20.12justinuif the answer is neither, i'm afraid you're on some vintage hardware ;)
17:20.18fu3I believe I am :)
17:20.19tronix1AESS? ;)
17:20.21fu3I will ask though.
17:20.22*** join/#asterisk skkip (n=Skipper@216.160.91.91)
17:20.23justinuheh
17:20.24tholoI 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...
17:20.30justinuNo 5. Crossbar ;)
17:20.32tronixhahaha
17:20.45justinubut I think the xbar switches will still do the loop current drop for disco sup
17:21.14fu3When I say to the telco guys about far end disco sup (I like that) they look at me funny..
17:21.23fu3then they say "yeah"
17:21.26fu3and then nothing. :|
17:21.34fu3I've got 350 lines!  you'd think they'd care ;)
17:22.05justinu350 analog lines?
17:22.15fu3yeah.. they come in over fiber
17:22.21fu3to 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)
17:22.28justinuwow... you should just ditch the analog and go w/ a DS3
17:22.36justinuPRIs over DS3
17:22.42fu3im planning on it, but i Have to make it work in a test environment first
17:22.50fu3and I need to sort out the other problems im having here in the mean time.
17:22.56Nuggetnah, just buy 350 clone X100Ps off ebay.
17:23.04fu3that seems unreasonable :)
17:23.23fu3I'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.
17:25.50salviadudhow about channelbanks?
17:25.51skkipanyone suggest a wireless sip phone on the cheap. I need at least 100 for trade show type events.
17:26.23kpettitskkip, good luck
17:26.49kpettithaven't heard of a wireless sip phone that works decent yet, let alone one that is cheap
17:26.55asteriskmonkeyi have :)
17:27.01kpettitdo tell
17:27.02asteriskmonkeyive got one that works great
17:27.07asteriskmonkeyits the wip5000 :)
17:27.16asteriskmonkeybut its not cheap :(
17:27.24skkipfrom where monkey?
17:27.36RoyKdoes it work if you sit > 5m from the access point?
17:27.39asteriskmonkeythe wililamsglobal place i work at
17:27.45asteriskmonkeythey have like 4 different ones
17:27.57asteriskmonkeythere is a quad band gsm with wifi sip that does handoff too :D
17:28.03asteriskmonkeybut that one is uber expensive :(
17:28.12skkipnice
17:28.20justinunokia is coming out with a quadband GSM/wifi voip phone
17:28.21asteriskmonkeyyou can put 2 sim cards in though :)
17:28.24skkipbut I just need them for local
17:28.24justinuE61 i think
17:28.29skkipcheck in check out deal
17:28.30justinutreo form factor
17:28.55asteriskmonkeycool, 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
17:29.00asteriskmonkeyi dont know how it does that though
17:29.03kpettithttp://www.voip-info.org/wiki/view/Hitachi
17:30.38kpettitasteriskmonkey, how many different wireless sip phones have you tried?  Do you like the wip5000 the best?
17:31.07asteriskmonkeyive tried 5 the wip is the best but i think the gsm/wifi ones comming in march will be superior
17:31.23kpettitgsm/wifi will be killer
17:31.39asteriskmonkeythey have it now
17:31.44twisted[asteria]no, murderers will be killer
17:31.46kpettitis 320 standard price for those?
17:31.47asteriskmonkeywe carry 2 phone that do that with passoff
17:31.53twisted[asteria]gsm/wifi will be needed :)
17:32.03jbalcombAnyone using the ARI web interface for voicemail and have a solution for the permissions of the VM files being root.root?
17:32.12asteriskmonkeythe new ones comming in march look like the razor but have full video suppport so streaming video conversations :D
17:32.33twisted[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)
17:33.16twisted[asteria]i say fax over single frame photos
17:33.21twisted[asteria]i'll make a flipbook and then you can call me
17:33.42asteriskmonkeylol
17:33.47kpettitwith 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)
17:35.49asteriskmonkeyjust reload
17:36.11kpettitsweet.  Trying to do voicemail with it now.
17:36.39*** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar)
17:37.38Hmmhesaysi need a free app for turning a divx video into dvd format
17:38.58*** join/#asterisk epablo (n=epablo@200.75.139.188)
17:39.09epabloHi people11
17:39.14fu3hi
17:39.40*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
17:40.03epabloAny idea on why I can be getting a "ZT_SPANCONFIG failed on span 1: Invalid argument (22)" when doing a ztcfg?
17:40.18fu3lets see the config file
17:40.23*** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de)
17:40.23shmaltzMy 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)
17:40.41kpettitHmmhesays, look at dvdrip
17:41.21Hmmhesaysdvdrip?
17:41.28epablofu3:   http://pastebin.ca/42791
17:41.48kpettitHmmhesays, google for it or look for it on freshmeat.net.
17:43.34epablofu3:  it's suposed to be a TE400P (tor2) card I bought on ebay
17:43.41fu3hmm
17:43.44*** join/#asterisk ful|work (n=fulgas@82.102.2.254)
17:44.02GerbilWrkhas anyone experienced clicking sounds during calls through Teliax?
17:44.50fu3that looks ok
17:44.52fu3strange
17:45.42jbalcombshmaltz i heard there is an issue with the firmware that causes something like that
17:45.49fu3well
17:45.55shmaltzjbalcomb, which one the latest?
17:46.06shmaltzjbalcomb, or is it with 7.1 as well?
17:46.50jbalcombshmaltz 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
17:47.35jbalcombs/hear/here
17:47.53shmaltzjbalcomb, he wanted what firmware?
17:48.04shmaltzI got 6.x, and the latest 7.x
17:49.07jbalcombshmaltz 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.
17:49.28shmaltzjbalcomb, does it happen to you?
17:49.47epabloshmaltz: i haven't used that provider.. but the 6.x firmware works better for me
17:49.50jbalcombshmaltz 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)
17:50.33shmaltzjbalcomb, how many simul phone calls you handle on those?
17:50.40shmaltzepablo, thanks
17:50.46*** join/#asterisk stoffell (n=stoffell@d5153FE41.access.telenet.be)
17:50.56jbalcombshmaltz we have the 7940s so 2 would be my guess
17:51.21shmaltzjbalcomb, b/c they telling me it happnes mostly when 3 or more calls
17:52.09jbalcombshmaltz 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
17:53.16jbalcombshmaltz 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)
17:54.50shmaltzjbalcomb, 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)
17:55.56jbalcombshmaltz 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?
17:57.16sevarddocelm0: you around?
17:57.19shmaltzjbalcomb, 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
17:57.42shmaltzjbalcomb, this script is somewhere in the user list archives from around a year ago
17:57.47jbalcombshmaltz ah, ok. im looking for it now.
17:58.29sevardDoes 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)
17:59.11shmaltzjbalcomb, http://lists.digium.com/pipermail/asterisk-users/2005-February/088614.html
18:01.44fu3ok
18:01.48*** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar)
18:01.50fu3I just got off the phone with qwest
18:01.57fu3they have no idea what "disconnect supervison" is
18:02.11fu3and they cant confirm my questions about what "is supposed to happen upon disconnect"
18:02.16ManxPowerfu3, It's called Far End Disconnect Supervision.  Read the tarrifs to get a USOC.
18:02.30fu3I said Far End Disconnect Supervision to them
18:02.35salviadudwow, those guys are amateurs
18:02.39fu3I dont know what the tariffs are, or a USOC.
18:02.41bweschkefu3: a.k.a. CPC (call progress control0 a.k.a. reversing line polarity momentarily at the end of the call
18:02.41ManxPowerhence my suggestion to read the tarrifs.
18:02.55fu3bweschke.. yeah when I said that, they couldnt confirm if it was supposed to happen or not.
18:02.59ManxPowerI've never heard of a real telco in the USA NOT providing FEDS
18:03.08fu3I didnt say CPC.. I should mention that.
18:03.15*** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar)
18:03.21bweschkefu3:  http://www.voip-info.org/wiki/index.php?page=Asterisk+Disconnect+Supervision
18:03.29fu3ive been over that a million times
18:03.32bweschkek
18:03.32fu3thanks though
18:03.35ManxPowerfu3, tarrifs == legal description of telco services as required by the regulators.  USOC is the Universal Service Ordering Code, it's the "part number"
18:03.40bweschkedid you try the lighted keypad thing?
18:03.42fu3ahhhh cool!
18:03.47*** part/#asterisk epablo (n=epablo@200.75.139.188)
18:03.49fu3no... I dont have one, and cannot find one.
18:03.54bweschkek
18:04.01fu3I had a qwest guy here TODAY
18:04.04bweschkeu have a dc voltmeter?
18:04.08ManxPowerfu3, you're trying to get me to do all the work, right?
18:04.10fu3and he promised me that we were using loopstart
18:04.14fu3yes, I have a multimeter.
18:04.20fu3ManxPower.. 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)
18:04.23fu3I want to learn!
18:04.27bweschkehook it up to the line - see if it dips momentarily when the call ends
18:04.31fu3it doesnt
18:04.32ManxPowerhttp://www.google.com/search?hl=en&q=site%3Awww.qwest.com+usoc&btnG=Google+Search
18:04.33fu3already verified that.
18:04.47bweschkethen u aren't getting it from them - but you knew that already
18:04.52fu3I do not get a battery drop, a polarity shift, OR a busy signal, or anything.
18:05.04fu3I dont know what you mean bweschke
18:05.09fu3i DO get my lines from qwest
18:05.17fu3ohh
18:05.18bweschkeyou aren't getting CPC signaling from them
18:05.19fu3duh :)
18:05.20fu3yeah.
18:05.23fu3thats right
18:05.36fu3but they swear their gear is working fine..
18:05.38fu3what do i tell them?
18:06.23bweschkeI'm sure it is - they're not required to provide it on standard analog service - unfortunately
18:06.23fu3this isnt
18:06.23fu3its a business trunk
18:06.23fu3350 lines
18:06.23fu3across a fiber, to a channel bank, to the demarc and then to the stations.
18:06.23bweschkestill.... depends what the prod definition of it is for them
18:06.32fu3I see..
18:06.35fu3garrh.
18:06.36bweschkeoh - well that's gotta be a config setting in the channel bank I'd think
18:06.44fu3The telco guy is coming back out here in a little while.
18:07.18fu3The guy also said that 40VDC is normal for our lines.
18:07.21bweschkecause 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
18:07.21fu3at least, he suspected.
18:07.27fu3my lines idle at 38-41 VDC
18:07.32fu3AT the DEMARC!
18:07.40*** join/#asterisk delmar (n=Delmar@203-114-178-231.inspire.net.nz)
18:08.01fu3bweschke.. good point.
18:08.16fu3all the batteries and everything are all right next to the CB.
18:08.40fu3Shouldnt my lines be running at around 48VDC though?
18:08.54fu3is a drop of 10 volts going to cause these kinds of problems?
18:08.59ManxPowerfu3, what state are you in?
18:09.04fu3Minnesota
18:09.10ManxPowerHEY! you have a channel bank?
18:09.11bweschkeno. not necessarily - I think there's a tolerance for analog line voltage
18:09.20fu3Not one I can use for testing
18:09.39fu3but yes, there are four CBs.
18:09.51ManxPowerA channel bank totally invalidates anything anyone has told you about CPC
18:10.02fu3great :)
18:10.19fu3I always liked square 1
18:10.20ManxPowerand invalidates most of the CPC info on the wiki
18:10.40ManxPowerfu3, you need to start by finding out if you have a CT1 (channelized T-1 voice) or a PRI
18:10.54fu3i'm almost certain (but not 100%) that it's a CT1.
18:11.07ManxPowerbecome %100
18:11.14justinumost CB's don't do PRI, no?
18:11.26bweschkejustinu: sure they do - the ones that do flex data/voice do
18:11.35justinuhmm, i've never had the pleasure of working with one
18:11.36fu3well.. the phone company told me it was a CT1 earlier, but I dont trust anything they say anymore.
18:12.01fu3brb
18:12.04*** join/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com)
18:12.40ManxPowerMost CT1s seem to provide CPC using CAS.
18:13.04ManxPowerfu3, So you have Telco->line->CB->????
18:13.07ManxPowerAnd what is the ????
18:13.37ManxPowerAdtran 850s can, 750s can't.
18:13.44ManxPowerfu3, what BRAND of channel bank do you have?
18:15.45fu3I can find out in a few minutes.
18:15.47clyrradhow 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"
18:15.58fu3I am telco->fiber->mux->CB->demarc->station
18:16.24bweschkefu3: DS3 mux?
18:16.46justinuprobably OC3 to DS3 to T1
18:16.54justinui have some facilities delivered like that
18:16.57ManxPowerclyrrad, I don't believe #include supports that
18:17.23clyrradManuxPower, will it work if I just include a variable?  instead of a path and variable?  Or what is the way around this?
18:17.26ManxPowerclyrrad, since #incude is read ONCE on load and never again
18:17.35bweschkeno. it doesn't. because it needs to process #includes before eval'ing the expressions for variables
18:17.41justinufu3: bottom line here is that the CB is before the demarc, so its your telco responsibility to provide that CPC loop current drop
18:18.01clyrradSo what is the solution if I need this type of functionality?
18:18.15clyrradIs there another function or app I can use to accomidate this?
18:18.17ManxPowerclyrrad, Goto(${ACCOUNTCODE},extension,1)
18:18.28clyrradbut its a File that i need included
18:18.36clyrradthe file has a bunch of dial plan stuff in it
18:18.53ManxPowerclyrrad, What are you trying to accomplish?
18:19.16clyrradI have a custom file, main menu.... for differnt DID's, each DID has its own main menu (ie seperate companies)
18:19.23ManxPowerclyrrad, #include works EXACTLY like a C #include
18:19.35clyrradwhen you dial in, based on the account code it should include the respecive main menu
18:19.41ManxPowerclyrrad, put each company in it's own context, then use Goto based on the DID
18:19.55clyrradI have each company in thier own context
18:20.11ManxPowerso what's the problem?
18:20.13clyrradI just have the functionaitly of the main menu in seperate files so as not to clutter up my extnesions.conf file
18:20.31ManxPowerclyrrad, Well you'll have to do an #include for each company manually in your dialplan
18:20.34clyrradand I need a way to get that file included with out having to hardcode the patch
18:20.44clyrradthats what i was getting at....
18:20.45clyrradhrm
18:20.45ManxPowerclyrrad, You can.t
18:20.46clyrradthat sucks
18:20.51ManxPowerclyrrad, #include works EXACTLY like a C #include
18:21.03djMaxso I upgraded * recently.  Now, sip show peers has unknown hosts for everything.  Did something change with the handling of sip peers?
18:21.11clyrradmake sense, thats why I was wondering if there was another way to do it with out hard coding
18:21.11need_sccp_helpAnyone familiar enough with sccp in asterisk that they could help me with the proper sccp.conf for cisco 12SP phones?
18:21.12ManxPower#include is processed before ANYTHING else.
18:21.22clyrradI see....
18:21.29ManxPowerclyrrad, hardcode it
18:21.38clyrradtoo bad there is not such a function, woudld come in handy
18:21.55clyrradthanks for clarification
18:22.54sevardDoes anyone have logs for this channel from now to about 3 weeks ago?
18:23.45iCEBrkrlol
18:23.52fu3justinu..  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?
18:23.56iCEBrkrsevard: Stop being a Nancy Boy.
18:24.03sevardiCEBrkr: sup bro
18:24.06iCEBrkr:D
18:24.15sevardI 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)
18:24.28iCEBrkrsevard: I don't think it's 100% ready..
18:24.37sevardI suggested that somebody should write a weather script that uses allison's voice in the wx/ directory and he went out and did it
18:24.41fu3Now, 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)
18:24.58iCEBrkrsevard: I understand that, I tested it a few times for him
18:25.06sevardiCEBrkr: I know it wasn't but I was under the assumption that he had a project page up
18:25.14iCEBrkrsevard: I don't think so.
18:25.18justinufu3: correct
18:25.23sevardiCEBrkr: how's she sound
18:25.38iCEBrkrsevard: Oh, I tested it before it had all the sound files worked out
18:25.47sevardneat
18:25.47djMaxcan anyone point me to how host=dynamic in SIP is "interpreted"?
18:25.48justinufu3: basically the channel bank should interrupt the loop current for approximately 500ms
18:25.56justinufu3: so it may be difficult to see that on your meter
18:26.06fu3ahhhh
18:26.11fu3yeah thats pretty short.
18:26.16justinuthe duration is configurable
18:26.21iCEBrkrsevard: 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
18:26.25justinuyou might try at home with a real POTS line and see what happens
18:26.44sevardiCEBrkr: cool. thanks for the update.
18:26.44fu3I will give it a whirl.
18:26.48iCEBrkrnp
18:26.55fu3I certainly appreciate all of everyone's advice.
18:26.59fu3thanks!
18:27.09need_sccp_helpHost=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)
18:27.24justinusorta
18:27.31justinuit actually means that you don't know the address of the client
18:27.48justinuand that it has to register before you can talk to him
18:28.13*** join/#asterisk perlmonky (n=perlmonk@pix.benchmark-systems.com)
18:28.17need_sccp_helpOK, that sounds good, more in depth than my answer
18:28.27djMaxso 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)
18:28.47djMaxbut I can make outbound calls just fine, which is strange.
18:28.55justinudjMax: sounds like a NAT binding issue
18:29.03justinudjMax: is qualify=yes set for the phones?
18:29.34djMaxin *?  no
18:29.35stsehi! Are here some experts concerning the hint extensions and the notify messages (terminated,busy,ringing)?
18:29.35sevardiCEBrkr: you a ladies man? :)
18:29.40*** join/#asterisk miller7 (n=none@gige-2.office-nl.irismedia.gr)
18:29.46RoyKhm
18:29.47justinudjMax: yeah, try setting that in sip.conf.
18:29.55RoyKsuddenly meetme stopped talking to me
18:29.56RoyK<PROTECTED>
18:30.03RoyK"that is not a valid conference number"
18:30.13RoyKbut
18:30.13RoyKconf => 10,10,10
18:30.18RoyKso it should all be fine
18:30.24RoyKdoesn't matter what i try
18:31.25*** join/#asterisk Defraz (n=t0tal@tim.mychoice.cc)
18:31.26bweschkestse: what is it u need to know?
18:31.57bweschkeRoyK: zaptel/ztdummy still loaded?
18:32.03djMaxso a phone doesn't have to register to make an outbound?
18:32.20need_sccp_helpNot necesarily
18:32.48RoyKbweschke: lol. doesn't matter...
18:33.02RoyKbweschke: the audio will suck if ztdummy isn't loaded, but meetme will work
18:33.13RoyKalso, ztdummy _is_ loaded
18:33.39justinudjMax: 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)
18:33.56djMaxso is there any way to force register, or just pull the plug?
18:34.43puzzledevening
18:35.01justinudjMax: depends on the phone
18:35.04stsebweschke: 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.
18:35.07djMaxk.  Plug it is.
18:36.22*** join/#asterisk WAudette70 (n=WAudette@c-67-170-156-3.hsd1.or.comcast.net)
18:37.38stsebweschke: 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)
18:38.15my007mssrvpleas some one help
18:38.58*** join/#asterisk kkun (n=none@198.60.73.230)
18:38.58[TK]D-Fenderstse : the first
18:38.58sevardHas anyone tried using Hamachi with SIP ?
18:38.59my007mssrvgenzaptelconf # channel 1, WCTDM, inactive.
18:39.31my007mssrvwhen i run genzaptelconf config file have all chanel like this one # channel 1, WCTDM, inactive.
18:39.58bweschkestse: 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)
18:40.21iCEBrkrsevard: am I a ladies man?? WTF?
18:40.28bweschkestate: the SIP/number SIP/number is the [number] in the sip.conf of the device that extensions 1000 belongs to in the dial plan
18:40.51sevardiCEBrkr: 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.
18:41.05iCEBrkrlol
18:41.08justinudamn
18:41.09*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
18:41.11my007mssrvhello all
18:41.14justinube careful...
18:41.16sevard:)
18:41.34stsebweschke: thanks, wo what does the example "exten => 200,hint,SIP/201&SIP/202&SIP/203"?
18:41.36justinusome women get all greedy when you do shit like that for them
18:42.03sevardI'm going to borrow an aircompresser(-or?) to blow up all the balloons
18:42.17justinuor a tank of helium
18:42.18[TK]D-Fenderstse : 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)
18:42.39sevardtanks of helium are more expensive but a better option, i'll look into it
18:42.41bweschkestse: 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
18:42.52bweschkestse: not advised for you to use it that way
18:42.58sevardjustinu: you have any more awesome ideas? :)
18:43.03justinuheh
18:43.19justinuthere should be a party supply store in your neighborhood
18:43.22justinuthey can arrange it all ;)
18:43.22my007mssrvpleas someone help me i have error msg say that Notice: Configuration file is /etc/zaptel.conf
18:43.22my007mssrvline 0: Unable to open master device '/dev/zap/ctl'
18:43.46sevardjustinu: there is but their services would empty my wallet and some
18:43.53justinuheh
18:43.57sevardthat goes against the greedy policy
18:44.04justinutry renting a tank from them
18:44.10stsebweschke: 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.
18:44.23bweschkeyep
18:44.26bweschkethat looks right
18:44.35sevardthat's what I had in mind.  I'd probably be ~30 bucks and would have a better effect than air.
18:44.51djMaxheh.  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)
18:44.54stsebweschke: but I still get the terminated notfiy the first time I start asterisk, after this I get nothing anymore.
18:44.58sevardI 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.
18:45.18iCEBrkr"Excuse me, I need enough balloons to fill a room that's 10x12, can you do that for me?
18:45.34justinuhow tall is the room? :P
18:45.39iCEBrkroops
18:45.42iCEBrkr10x10x12
18:45.44justinuheh
18:45.52[TK]D-FenderBalloon size desired?
18:45.56bweschkestse: 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
18:46.00justinuget the mylar ones
18:46.08iCEBrkrsevard: you're nutso
18:46.09justinuafter the birthday, you can release them and take down your local power grid
18:46.14sevardI think it's a 9 foot celing, probably 8x8, but with desks and crap it couldn't take more than 40-60 baloons
18:46.19sevardballoons
18:46.29sevardmylar ones! hahahahahaha
18:46.35iCEBrkrBZzzzzzzzzot
18:46.38stsebweschke: hm, after I restart asterisk, I always restart the phones to make sure they are correctly subscribed.
18:46.41[TK]D-Fenderbweschke : So Poly's going "sticky" is a bug on THEIR side? Hmmm...
18:46.55sevardThis reminds me of Danny Deckchair
18:46.57djMaxok, 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)
18:47.59sevardSo, room-full-of-balloons is still the best idea? :)
18:48.09bweschkeTKD-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
18:48.36stsebweschke: then I call the test number and get the terminated notify.
18:49.25bweschkestse: 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)
18:50.35justinudjMax: it should stay registered now that you've enabled the qualify (keep alive)
18:50.51justinudjMax: 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)
18:51.14djMaxyeah, looking a bit better.  "OK (83 ms)"
18:51.30justinuok, that's good... it means your phone is responding to the keepalives in 83ms
18:51.31djMaxbut strangely, no other phones have registered.
18:51.35sevardHas anyone used Hamachi (or any VPN) with Asterisk?
18:51.41stsebweschke: with pleasure, how should I get them to you?
18:51.45justinusevard: i'd by my woman something a bit more risque.... but that's just me ;)
18:51.55bweschkestse: bugs.digium.com
18:51.58justinus/by/buy/
18:52.00sevardjustinu: I was thinking about that
18:52.11justinui'm not sure how well you know this girl
18:52.21sevardwell enough :)
18:52.24bweschkesevard: yes - we send calls w/Asterisk through Openvpn tunnels. works ok for the most part
18:52.31justinumaybe a night at a nice hotel w/ a jacuzzi tub
18:52.38justinucover the bed in rose petals
18:52.42justinuchicks totally flip over thta
18:52.46sevardbweschke: I have Hamachi, a sort of VPN but I can't get my phone to register.
18:53.05sevardbweschke: I can do everything else over the VPN though, ssh, telnet, http, you name it.
18:53.20justinucan you ping the phone
18:53.45sevardjustinu: 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 :)
18:53.46bweschkesevard: 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?
18:53.55sevardjustinu: the asterisk box can ping the computer that the softphone is on
18:54.03justinuk
18:54.10stsebweschke: 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)
18:54.37sevardbweschke: 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
18:54.45bweschkestse: I'm a bug marshal - if you assign it to me and it's not - I promise I won't give you neg karma :)
18:54.50Kattyjustinu: or sneeze all over it.
18:55.06sevardbweschke: using Hamachi I can do everything I want minus Asterisk... but it technically should be working.
18:55.12justinukatty: alergic to roses too?
18:55.21Kattyjustinu: thankfully, no
18:55.44sevardKatty: female input go
18:55.52Kattysevard: huh?
18:56.00justinumy mom calls her cat katty
18:56.08justinua cat I gave her
18:56.17justinui dunno if it has a real name
18:56.20stsebweschke: I know you are. I read it. ;-) Do I need a acoount?
18:56.27sevardKatty: 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
18:56.27bweschkesevard: 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?
18:56.37djMaxok, 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"
18:56.45justinudjMax: NAT
18:56.54djMaxBut it's all on the same subnet
18:56.58justinuoh.
18:57.06sevardbweschke: Therein lies the problem.  I don't have much experience using UDP and wouldln't know how to test UDP connectivity.
18:57.21djMaxI mean, maybe it's something crazy dumb like I read seconds one place and msec another.
18:57.27justinutypically the problem you describe is caused by a NATing device closing it's binding after not seeing traffic
18:57.38Kattysevard: oh.
18:57.47justinuexpires are in seconds, iirc
18:57.50bweschkesevard: well - Asterisk doesn't know how to do any protocols outside of UDP (and it shouldn't because RTP over TCP would likely suck)...
18:57.52djMaxso normally that SIP session/registration is persistent?
18:58.12djMaxi.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
18:58.16justinudjMax: if your phone is on the same LAN as the asterisk machine, you shouldn't need qualify
18:58.34Kattysevard: anywhere near a beach?
18:58.37sevardbweschke: 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.
18:58.43justinudjMax: if you continue to have issues with registrations dropping off, try running a ping from the * server to the phone
18:58.56Kattysevard: actually, i'd seriously recommend a treasure hunt.
18:59.03djMaxif that were the problem, wouldn't * still show the IP in the peer list?
18:59.04justinulol, that's a great idea
18:59.07sevardKatty: Yes, Minnesota, land of 10,000 lakes.. except this time of year they have a good 2 foot layer of ice.
18:59.10Kattysevard: i know it sounds childish, but you could put a little twist on it.
18:59.24Kattysevard: 5 notes around the house, and then one that says to go to a resturant
18:59.33justinudjMax: if your phone doesn't re-register in time, asterisk will remove the IP from the peer list
18:59.41Kattysevard: and then secretly stash something in a purse or a pocket
18:59.47sevardKatty: 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
18:59.56bweschkesevard: 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
19:00.01djMaxyeah, that seems the best explanation, even though I set the timeout lower.
19:00.01justinudjMax: typically the only reason that might happen is NAT, or a network connectivity issue which stops the register messages from the phone getting to *
19:00.06Kattysevard: well the little notes could each be a little present.
19:00.16Kattysevard: note 1, look in bedroom.
19:00.21djMaxI guess now the question is whether qualify=yes is going to somehow make it go away
19:00.25Kattysevard: and the bed could be covered in flowers.
19:00.29Kattysevard: and another note on the bed.
19:00.35Kattysevard: and then note 2 could say look in bathroom
19:00.40Kattysevard: and you could have the whole spa thing going on
19:00.44sevardKatty: Heh.
19:00.45Kattysevard: but it'd be all for later
19:00.56Kattysevard: 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
19:01.11Katty_Paulo_: no it's not
19:01.15puzzleddiamonds
19:01.16salviadudpaulo, no way man
19:01.19salviadudchocolate
19:01.22sevardThat'd be nice... one twist, we're both college students and she's migating the cost of a dorm by staying with realitives...
19:01.23Katty_Paulo_: there are plenty of girls out there that don't want jewelry
19:01.25salviadudits the black gold
19:01.29justinugold is the shortest path to the heart of the wrong woman
19:01.46salviadudyeah, i agree, some women like nintendo
19:01.47Kattysevard: do you live with your parents?
19:01.49tronixsevard: one way to test udp: # nmap -P0 -sU -p <port> <host or IP>
19:01.51salviadudi love those women...
19:01.59justinui'd rather hang out with a woman who likes video games
19:02.03Kattysalviadud: glad to know i'm loved.
19:02.09sevardKatty: I live in ground level apartment with my room mate, but my room mate is disabled and never leaves the house.
19:02.22Kattysevard: you could include him!
19:02.23salviadudkatty, whats your fav nintendo game?
19:02.26Kattysevard: make him stash a note
19:02.29sevardKatty: wtf no 3way
19:02.34Kattysalviadud: hmm, though one.........zelda gold or rampage...
19:02.39Kattysalviadud: possibly contra
19:02.45salviadudcontra is badass
19:02.48justinuzelda: a link to the past
19:02.50Kattysevard: nonono
19:02.51justinubut that was SNES
19:02.54asteriskmonkeyrampage and super spike voly ball .. the orginal contra kicks ass :D
19:02.54mutanyone know anythign about tellab echo can shelvs?
19:02.58Kattysevard: one note could say See $person
19:03.02muthow large is a 255d shelf?
19:03.08Kattysevard: and $person could be secretly stashing a note
19:03.15Kattysevard: and that note could be the next clue
19:03.20justinumut: http://www.voip-info.org/wiki/view/Tellabs+Hardware+Echo+Cancellers
19:03.22EgonisI 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
19:03.25Kattysevard: it's a birthday party......treasure hunts don't have to be person
19:03.32sevardOH SNAP speak of the devil that was her on the phone just now, she's bringing me soup for lunch
19:03.36sevardshe's (##(&ing awesome
19:03.36Kattysevard: but i suppose you could always stash a note..uh..somewhere...person
19:03.38Kattya;
19:03.41Kattys/a;/al
19:03.48mutjustinu: um? that says nothin about em
19:03.52justinusevard: nothing like the love of a good woman
19:04.01sevardjustinu: this one's a keeper.
19:04.09justinumut: are you blind?
19:04.10Kattysevard: treat her well.
19:04.20justinumut: look under the heading "models of shelves"
19:04.20muti must be
19:04.21mutshow me
19:04.34sevardKatty: I've really not been, that's why I want to make this birthday special.
19:04.43muti don't see 255D info listed
19:04.46muti see 255A there
19:04.47WAudette70Katty:  Good ideas
19:04.52mutthats it
19:04.56*** join/#asterisk oogle (n=jart@justin.ctlinc.com)
19:05.02WAudette70sevard:  Good luck! Go all out on the keepers.
19:05.05sevardKatty: thanks for the input, very good ideas and I'm going to put them to use.
19:05.08oogleare 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)
19:05.51mutam i missing something justinu?
19:06.06sevardWAudette70: thanks bro:)
19:06.16Kattysevard: excellent (=
19:06.16justinui guess it doesn't have the details you're looking for
19:06.24Egonisexit
19:06.28mutyea.. like.. ANY
19:06.32justinuhowever, I believe the 255 is a 16 slot shelf
19:06.35sevardbweschke: you still here?
19:06.38mutk
19:06.42sevardI did 5060/udp open  unknown
19:06.43justinuw/ removeable alarm and access cards
19:06.45bweschkesevard: yes
19:06.45sevarderm
19:06.53ooglepastebin doesn't seem to like me
19:06.57sevardI did nmap -P0 -sU -p 5060 <vpn ip> and I got 5060/udp open  unknown
19:06.58mutpossible it's a 19" rack mountable shelf?
19:07.31justinutelco stuff is generally 21"
19:07.35justinuso I wouldn't bet on 19
19:07.39bweschkewhat about 5061 and 10000-20000 for the RTP or whatever port numbers the softphone wants to use?
19:07.46mutyea
19:08.15Kattymuttly.
19:08.16bweschkesevard: try a IAX based softphone
19:08.21WAudette70sevard:  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.
19:08.33Kattymuttly is a good name for a dog.
19:08.47justinumuttly is the name of a dog from a cartoon
19:08.52justinusmurfs?
19:08.52Kattyoh, is it?
19:08.57justinui think it was smurfs
19:09.05Kattywindshield wiper fluid = smurf juice.
19:09.07bweschkesevard: 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
19:09.26iDunnomuttly is from Dick Dastardly cartoons - such as Wacky Races and Catch The Pigeon
19:09.29tronixisn't iax 4569?
19:09.32sevard5061/udp closed unknown, isn't 5060 used for registration?
19:09.51tronixoh.. sevard's doing sip right now. gotcha
19:09.54iDunnothe dog in the smurfs was *not* called muttly.
19:09.57KattyiDunno: oh.
19:09.58bweschkesevard: yes - on the server... but this is a phone - not a server, right?
19:10.08KattyiDunno: was this an old thing?
19:10.24sevardbweschke: I'm nmaping the server
19:10.27iDunno(and it was actually muttely, but feh ;)
19:10.28bweschkesevard: some phones receive sip signalling for calls on 5061 - (eg the SPA3000)
19:10.30sevardbweschke: From the phone
19:10.35iDunnoKatty: well, reasonably, 80s IIRC
19:10.39iDunnohttp://www.hotink.com/wacky/dastrdly/
19:10.40sevardbweschke: the phone i'm using is an x-lite
19:10.42justinuthat's right.. dastardly  and his dog muttly
19:10.47Goraldo i actually put in a ip or leave this like this? bindaddr = 0.0.0.0
19:10.57Goralthe address i'm using is static
19:11.06salviadudleave it like that dude
19:11.11*** part/#asterisk WAudette70 (n=WAudette@c-67-170-156-3.hsd1.or.comcast.net)
19:11.19KattyiDunno: oh, that's not too old.
19:11.26iDunnorocked :)
19:11.37Goralso it will bind to any ip?
19:11.46salviadudexactly
19:11.46iDunnooh, erm, says 1969 ;)
19:11.53iDunnoso maybe I grew up to reruns of that :)
19:11.54Goralsalviadud ty
19:12.23*** join/#asterisk NotFreak (i=NotFreak@cp12193-e.tilbu1.nb.home.nl)
19:12.26bweschkesevard: well
19:12.33tronixI 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)
19:12.38justinuhttp://www.hotink.com/wacky/dastrdly/
19:12.43*** join/#asterisk NotFreak (i=NotFreak@cp12193-e.tilbu1.nb.home.nl)
19:12.46bweschkesevard: 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 -
19:13.11tronixBanana Splits was an interesting cartoon. one was a completely psychedelic episode... kids thought it was fun, adults knew the symbolism. ;)
19:13.30fu3with loopstart, that battery drop that occurs..  should it drop the line to 0VDC for those 500ms?
19:13.32sevardbweschke: alright, i'll do that, my gf is here, bbiab :)
19:13.59bweschkesevard: good call - * isn't your priority right now - or at least it shouldn't be. :)
19:14.32bweschkei've always got to remind myself of that when my wife and/or kids are looking for me.
19:14.40[av]baniyay new gxp firmware
19:14.41CMikeIf 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 ?
19:14.58CMike(ANI based)
19:15.03justinubani: any info?
19:15.13[av]bani?
19:15.27justinuon the new firmware
19:15.29[av]bani~gxp2000
19:15.29jbotrumour has it, gxp2000 is http://www.voip-info.org/wiki/view/GXP-2000
19:15.51justinucool
19:16.22justinuadded support for DHCP option 66!
19:16.24justinufinally
19:16.39[av]baniit defaults 0 though :(
19:16.45[av]banibut meh, better than nothing
19:17.14[av]baninot 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
19:23.11djMaxis there some way to tell it never to lose registrations, and if so is that safe?
19:23.38justinunot really
19:23.38GerbilWrkif you have an audio file playing, what do you put in to have it ignore key presses?
19:23.44[av]baniqualify=no
19:23.50justinudjMax: 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)
19:26.14DarthClueGerbilWrk, how are you playing the file?
19:26.15djMaxno hiccups
19:26.21ManxPowerAnyone 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)
19:26.32GerbilWrkexten => 1,1,Background(thevoice/mailing)
19:27.04djMaxno hiccups
19:27.07DarthClueuse Playback instead of Background
19:27.10ManxPowerdjMax, host=1.2.3.4 and don't have the phone register
19:27.12GerbilWrkok
19:27.22kkunAnybody familiar with sccp?
19:27.26djMaxsorry.  Wrong up arrow window.  So the problem seems to be some timing mismatch in re-registration.
19:27.42ManxPowerkkun, use SIP
19:27.47justinudjMax: are you sure?
19:27.48kkunCan't
19:27.50djMaxI set the expire time down to 100 seconds, now I'll see if the problem happens faster or what.
19:27.59kkunUsing 12sp phones, they don't support sip
19:28.19djMaxif this is really working on the polycom, I guess I should see reg messages every 100 seconds...
19:28.24ManxPowerah.  I'm sorry to hear that.  SCCP is just miserable
19:28.43*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
19:28.53justinudjMax: the registrar responding to the register messages can override the client preferred value
19:28.56kkunI know, any ideas / suggestions on how to setup sccp.conf?
19:29.10djMaxthat 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)
19:29.37justinui think there's a few other params... minexpiry also?
19:30.10djMaxyeah, min, default, max.
19:30.27kkunmanxpower,  the few 49xx phones I have are doing sip beautifully, I wish I could get sip firmware for the 12sp
19:30.39Qwell[]49xx?
19:30.42Qwell[]79xx?
19:30.49kkunRight, 79xx
19:30.56Qwell[]I just use sccp...it's pretty nice
19:30.58djMaxinteresting, no registrations from the phone.
19:31.01Qwell[]12sp+ is well supported I hear
19:31.02djMax(except for the first one)
19:31.20kkunqwell, can you help me with sccp.conf for a 12sp?
19:31.34Qwell[]kkun: should be pretty easy...
19:31.59kkunI can get the thing to register, and ring, but they won't dial
19:32.10Qwell[]kkun: need to put them into a context with a dialplan
19:32.41kkunThe context I specify in sccp.conf is the same default context as my sip phones, which work
19:32.52djMaxI guess stopping registration is the simplest fix.  Annoying though.  Especially because we never had this problem with * 1.0.
19:33.00Qwell[]kkun: in the lines section?
19:33.16kkun?
19:33.22kkunLine section?
19:33.29Qwell[]context is in the [lines] section of sccp.conf?
19:33.57kkunOk, I though it was in the general section
19:34.17Qwell[]that could work, but anything in [lines] overrides [general]
19:34.41GerbilWrkDarthClue, that worked, thanks
19:35.06*** join/#asterisk rtikk (n=rtikk@g220032.upc-g.chello.nl)
19:35.28kkunI have it in the lines  section as well, still no dice.
19:35.44Qwell[]and you restarted asterisk after making the change?
19:35.50justinudjMax: what kind of phones?
19:36.02kkunAs always
19:36.19kkunLet me show you my sccp.conf
19:36.25Qwell[]pastebin it
19:36.27Qwell[]~pb
19:36.28jbotpb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
19:36.47Qwell[]and show me the CLI output when you try to dial
19:38.23kkunhttp://pastebin.com/567253
19:39.11Qwell[]kkun: Which line/device is it?
19:39.17jbalcombkonyagayamada
19:39.22Qwell[]ahh, 130
19:39.33Qwell[]okay, near the bottom where you have "line => 130"
19:39.53Qwell[]either change that to "line => line130" or change the autologin= at the top to "130" instead of "line130"
19:40.00kkunCli shows that asterisk is receiving the numbers dialed, but not "dialing" it
19:40.33Qwell[]because there is no active line...
19:40.41Qwell[]at least...not a proper one
19:40.56Qwell[]the device shouldn't even be fully loading
19:41.15kkunOk, just a second
19:42.05*** join/#asterisk epablo (n=epablo@200.75.139.188)
19:42.19epablohi people.
19:43.25kkunqwell, rebooting
19:43.33Qwell[]rebooting?
19:43.36Qwell[]the phone, I hope
19:44.50epabloi'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)
19:45.18jbalcombanyone have an archive of the firmware releases for the GXP-2000?
19:46.43kkunqwell, thanks, that did it, I knew it had to be something stupid, 1 line of code.
19:47.10fu3can anyone tell me about the 500ms battery drop used with loopstart signalling?
19:47.18fu3to indicate a hangup.
19:47.50fu3more 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)
19:52.15sigmountehello ! what apps can i use under linux to record my own .gsm files ?
19:53.05hensema'evening
19:53.06Nivexsigmounte: 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.
19:53.26sigmountethanks you Nivex !
19:53.30fu3i'm surprised * hasnt included gsm recording.
19:53.55epabloYou can do it.. setting up an ivr
19:53.59fu3ahh
19:54.08hensemaI'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?
19:54.24sigmountei'm setting up an ivr by hand configuring my .conf , their is another way to do it ?
19:54.30hensemaso, in effect a reverse-CID: I want _our_ number to be displayed
19:54.56epablosigmounte: 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)
19:55.46JoanFrantziskohello everyone
19:55.50fu3HI!
19:55.58JoanFrantziskoi've a issue with Asterisk behind a firewall
19:56.04fu3you'll get over it
19:56.04fu3:)
19:56.07JoanFrantziskocan anybody help me?
19:56.25fu3i would but im just starting into asterisk myself.. sorry
19:56.34fu3there ARE people here :)
19:56.36ManxPowerJoanFrantzisko, the wiki is your friend
19:56.36Qwell[]JoanFrantzisko: ONly if yo ask a question
19:56.38ManxPower~docs
19:56.39jbotdocs 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
19:56.44Qwell[]I botched that one...
19:56.44fu3ask the question
19:57.04JoanFrantziskook
19:58.16W8TAHGood 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?
19:58.42epabloyes.. block it on the network
19:58.49JoanFrantziskowhen i call of the Internet, the receiver can hear me, but i can't hear the receiver
19:58.58NivexW8TAH de N8VNR: absolutely.  It doesn't care what IPs it's talking to, internal or external.
19:59.01ManxPowerJoanFrantzisko, that is correct and expected.
19:59.25W8TAHok -- cool -- what clients are recomeneded to connect to such a server?
19:59.27ManxPowerunless you do your basic homework about running Asterisk behind NAT, as is documented on the mailinglists, Wiki, various books and messages.
20:00.02epabloW8TAH:  you can start with a softphone.. xten-lite is good and free
20:00.17W8TAHepablo, excellent -- thank you
20:00.19ManxPowerhensema, SetCIDNum(${EXTEN})
20:00.34W8TAHNivex, Cool -- thanks
20:00.52*** join/#asterisk bkw_ (n=bkw_@m788636d0.tmodns.net)
20:00.57W8TAHepablo, do you have any experience setting it up on Gentoo? -- i know it is in portage
20:01.03ManxPowerASSUMING it's a PRI
20:01.04JoanFrantziskoManxPower: if we are use SIP Protocol, all is OK, but the problem is when we use H-323
20:01.22JoanFrantziskoManxPower: the Firewall is Astaro Security Gateway
20:01.24ManxPowerJoanFrantzisko, I don't believe that you can put Asterisk behind NAT if you are running H323
20:01.28epabloW8TAH: Nop.. i'm more of a RH guy,.,.but it should work fine
20:01.57W8TAHepablo, ok --
20:02.21djMaxjustinu: polycom IP500 phones
20:02.33W8TAHepablo, 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)
20:02.43djMaxheh, this is just plain weird.
20:03.07djMaxSo I see reg requests from the poly "frequently," but not frequently enough to avoid * kicking it out before it reregisters.
20:03.24djMaxit's almost like one of their clocks is "very" fast
20:03.25JoanFrantziskoManxPower: H323 can't run behind a NAT? because Astaro include open h323 kernel module
20:03.29ManxPowerdjMax, Dunno.  I have over 60 polycoms and have none of these problems.
20:03.30*** join/#asterisk exstatica (i=exstatic@aboutmylife.net)
20:03.39ManxPowerJoanFrantzisko, No, it's because of RTP
20:04.13ManxPowerH323/NAT puts the IP address/port information in the data portion of the voice packet and signalling packets, so normal NAT won't work
20:04.17epabloW8TAH:  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
20:04.27fu3you could do H323 with PAT.
20:04.30fu3I think :)
20:04.32ManxPoweralso usually the SIP or H323 proxies in firewalls are for CLIENTS behind NAT, not for SERVERS behind NAT.
20:04.48djMaxManxPower, I've never had these problems either.  Just after * 1.2
20:05.09W8TAHepablo, thanks a bunch -- looks like i got a gentoo box to build -- LOL
20:05.12ManxPowerdjMax, I've been running 1.2RC for a while and 1.2.4 for a week.
20:05.13JoanFrantziskoManxPower: what can i do?
20:05.16ManxPowerno known problems
20:05.20ManxPowerJoanFrantzisko, don't use H323
20:05.23epabloW8TAH:   Have fun!
20:05.25ManxPoweror use a different IP PBX
20:05.28W8TAHwill do
20:05.33ManxPoweror write support for it in asterisk.
20:06.23djMaxguess I could just update and such
20:06.28cpmGot one of them thar fancy x100p 'VoIP 1' fxs boxes in today.
20:06.48cpmI'm sorry to say, it beats the pants off of the IAXy.
20:06.52djMaxis registration (vs static IP) better/worse?
20:07.03ManxPowerdjMax, Neither.
20:07.05JoanFrantziskoManxPower: this issue with H323 behind a Firewall is caused for a bug in the linux kernel? or Asterisk not support the implementation?
20:07.11Mavvieis Corydon76 on this channel?
20:07.27*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
20:07.30KattyCorydon-w: Mavvie
20:07.36ManxPowerJoanFrantzisko, I'm sorry, I cannot help you further.
20:07.41ManxPowerBut Asterisk on a public IP
20:08.07Corydon-wWhat?
20:08.25KattyMavvie: Corydon-w
20:08.30fu3ManxPower.. you're the best!
20:08.30MavvieCorydon-w: can I discuss the closure of 6565 with you?
20:08.35ManxPowerfu3, why?
20:08.38fu3just because.
20:08.43fu3you actually try.
20:08.46fu3it's refreshing.
20:08.49Corydon-wMavvie: sure
20:09.08ManxPowercpm, it runs IAX2?
20:09.16cpmYup. That's all it runs in fact
20:09.21Corydon-wMavvie: try the #asterisk-bugs channel
20:09.21cpmno more SIP
20:09.34justinucan you put bootrom 3.1 on an IP 301?
20:09.46MstlyHrmlsyup
20:09.55justinufor some reason it's not picking it up
20:10.10justinuit grabbed the sip.ld and loaded that, but not bootrom.ld
20:10.18MstlyHrmlsjustinu: is it giving an error?
20:10.21justinunope
20:10.37MstlyHrmlsjustinu: does it say anything in the <mac>-boot.log?
20:11.01justinuah yes, it does
20:11.10cpmdon't care for the cli on this box, I mean not at all.
20:11.14jbalcombanyone have an archive of the firmware releases for the GXP-2000?
20:11.24*** join/#asterisk PakiPenguin (i=uppal@linuxpakistan/admin/pakipenguin)
20:11.27PakiPenguinhello everyone
20:11.35epablo<PROTECTED>
20:11.43PakiPenguinanyone here uses astguiclient?
20:11.52PakiPenguini need some quick help please
20:11.57justinuMstlyHrmls: http://pastebin.ca/42806
20:12.12MstlyHrmlsjustinu: looking
20:12.23justinuguess 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)
20:12.38justinui used that bootrom.ld file on my ip501/601s without trouble
20:12.50MstlyHrmlsjustinu: how big is your bootrom.ld file, and what version of 3.1 is it?
20:13.06saftsackhi
20:13.07justinusize is 422925
20:13.13sigmountedo the .gsm file have a special format to follow when creating some myself ?
20:13.46MstlyHrmlsjustinu: that seems small, let me have a look at mine
20:14.04justinunot sure of the exact version
20:14.16jbalcombepablo: is that 1.0.2.13 and 1.0.2.12?
20:14.27MstlyHrmlsjustinu: my copy of 3.1.2 is ~2.5 megs...
20:14.32justinuinteresting.
20:14.46MstlyHrmlsjustinu: it looks like you've got an incomplete version somehow
20:14.48justinumy version might only include the bootloader code for the 501/601 phones
20:15.00justinuinterestingly enough, my copy of rom 3.0.1 is 2.1 meg
20:15.01MstlyHrmlsjustinu: that's what I'm thinking
20:15.39saftsackis there a possibility to log and archive all outgoing faxes from my normal fax machine which is connected to an tdm fxs card?
20:16.01MstlyHrmlsjustinu: 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
20:16.08ManxPowersaftsack, no
20:16.22justinudo I have to ask my resller for the correct bootrom, or can I download it somewhere?
20:16.32ManxPowerunless you want asterisk to itercept the call, accept the fax (using rxfax), then retransmit the fax (using txfax)
20:16.33JoanFrantziskoManxPower: ok, i got it... i will try with H323 connected directly to the internet, and SIP behind the Firewall... thnxs :)
20:16.39ManxPowerand really, that's just asking for trouble
20:16.49bweschkejustinu: yes - any version of the X01 should have memory enough to support the later bootroms
20:17.19MstlyHrmlsjustinu: techncially you have to ask your reseller, but I'm sure if you asked around...
20:17.37justinuactually, hang on
20:17.39justinui might have this somewhere
20:17.49justinutoo many files, too many machines
20:18.02MstlyHrmlsbweschke: actually I've been able to load 3.x on all the Polycoms I have, including 300s and 500s
20:18.39bweschkeMstlyHrmls: 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
20:19.04MstlyHrmlsbweschke: the 300s and 500s won't do the HTTP, but they use the rest of 3.x just fine
20:19.15epablojbalcomb: i got GXP2000_Release_1.0.1.9.zip,  Release_1.0.2.16.zip
20:19.20bweschkecurrent bootrom is 3.1.3 - current fw is 1.6.5
20:19.23[av]banitis very silly polycom wont put http on the 501
20:19.38[TK]D-Fenderbweschke : 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...
20:19.55bweschkeTKD-Fender: yes that's true
20:20.07[av]baniepablo: 1.0.2._16_ ??
20:20.08MstlyHrmls[TK]D-Fender: doesn't matter what you load a 3.x on to, you can't downgrade to 2.x
20:20.14jbalcombepablo when did 1.0.2.16 come out? the wiki only has .13 Can you email them to me? PM for address
20:20.21justinuok, i have sip 1.6.3 (which is probably fine)
20:20.28justinubut no bootrom 3.1.3
20:20.37justinuanyone wanna help a brutha out?
20:20.38bweschkefrom 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.
20:20.42jbalcomb[av]bani how is you auto-provisioning project coming?
20:20.51[av]banijbalcomb: :D
20:20.58[av]banicoming along nicely, i got the grandstream stuff figured out
20:21.16justinubweschke: yeah, i'm aware of that... i like the 3.x roms because of http provisioning
20:21.25epablohttp://www.grandstream.com/BETATEST/
20:21.34jbalcomb[av]bani i have my personal phone being configured through TFTP and I think I'm digging it.
20:21.38epabloThere is were I got the new ones
20:21.47*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
20:21.53jbalcombepablo: they only have 1.0.2.3 there
20:22.07saftsackManxPower, 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]banijbalcomb: mine does via php, and you can do it totally out of the box, zero config.
20:22.30epablojbalcomb:   let me make sure that zip is for gxp
20:22.34jbalcomb[av]bani do you have anything you can share with me?
20:22.38[av]bani:D
20:22.49[av]banimaybe when i clean it up
20:23.08jbalcomb[av]bani I have 120 of those phones so I could certainly appreciate it and provide some additional feedback
20:23.16epablojbalcomb: You are right it's for other phones
20:23.26[av]banii thought you said you were fine with tftp and manual config, and pooh-poohed my script :)
20:23.30jbalcombepablo: ok, thats cool. good lookin' out though.
20:24.09jbalcomb[av]bani i said it wasn't a that big of a hassle given the notion of having to configure every phone
20:24.59jbalcomb[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
20:25.19jbalcomb[av]bani for each dept and only updating the firmwares a dept. at a time.
20:25.34MstlyHrmls[av]bani: actually I have a 501 here on my desk that's using HTTP
20:25.50ManxPowersaftsack, 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]banijbalcomb: 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]banijbalcomb: 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)
20:28.51jbalcomb[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)
20:33.06austinnichols101[av]bani: got an initial response from InterNap
20:33.15[av]baniaustinnichols101: and?
20:33.21[av]bani"we are looking into it"?
20:33.27austinnichols101[av]bani
20:33.35austinnichols101[av]bani: more or less.
20:33.35[av]baniaustinnichols101
20:33.42[av]baninice empty response
20:33.53[av]banithey should have been looking into it 2 years ago
20:34.12[av]baniwe'll see if it's all talk
20:34.25austinnichols101[av]bani: check separate window for a copy of the message
20:35.25austinnichols101[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.
20:35.43ManxPowerDoes anyone have a recommended vendor for RJ21X (Amphenol) cables?
20:35.54bweschkeManxpower: Graybar
20:35.56austinnichols101[av]bani: but you're right that the response is weak
20:36.22ManxPowerI was hoping to avoid them.  Their website ordering system is horrid
20:36.34ManxPoweractually their online catalob is horrible
20:37.01bweschkeManxpower: 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)
20:37.28ManxPowerbweschke, 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)
20:37.52bweschkewhat length? cat3 I assume?
20:38.09bweschkewhat connectors M/F ? M/M ? F/F ? any specific angles?
20:38.14ManxPowerReally, all I need is an Amphenol gender changer
20:38.41ManxPowerWe are replacing a TA850 with a TA750 and they have different genders
20:39.11AlexCTIHi Everyone, I need some help to interconnect 2 IAX2, some one can help me?
20:39.29ManxPowerthe 850 would have a male cable going into a female port, whereas the 750 needs a female cable going into a male port
20:40.00bweschkeso you need a F/F gender bender?
20:40.13ManxPowerI 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)
20:42.19hensemawhen I enter a digit, asterisk tries to connect me to the extension in the *calling* context, while Background() is run from a macro
20:42.30epabloFeb 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
20:42.37hensemawhat's up with that?
20:42.45jbalcombManxPower Graybar
20:42.45stoffellGXP-2000 owners already tried newest build?
20:42.45*** join/#asterisk afrosheen (n=test@txprotoa2.august.net)
20:42.58ManxPowerepablo, the jumpers on the card
20:43.00*** join/#asterisk gambolputty (n=root@64.74.225.135)
20:43.07jbalcombstoffell: 1.0.2.13? I've got it on 100+ GXP-2000s
20:43.21afrosheenManxPower: Thanks for your help with zap the other day, it's perfect but now I have echo on my sip trunk :p
20:43.33ManxPowerafrosheen, you can't have echo on SIP
20:43.34jbalcombepablo: sounds like you have it set up as an E1 rather than a T1
20:43.35stoffelljbalcomb, yes, it's released yesterday/today, better then before?
20:43.37epabloManxPower:  but ztcfg says its ok
20:43.41afrosheenwell, we do, and it's bad
20:43.43ManxPowerepablo, ignore tht
20:43.49*** join/#asterisk ptblank (n=MURDER1@68-169-161-61.lmdaca.adelphia.net)
20:43.55afrosheenI guess our provider has something messed up on their end
20:43.57epabloManxPower: ok
20:44.43bweschkeafrosheen: 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
20:46.23afrosheenis it possible my provider has a hybrid? the calls have to hit the pstn somewhere and it's not here
20:46.29jbalcombstoffell: 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.
20:46.46stoffell2 weeks ago?
20:47.06stoffellu, talking about 1.0.2.13? released 2/21/2006 ...
20:49.22clyrradI 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?
20:50.48djMaxok, now I know * is whacked out.
20:51.05djMaxI put static IPs in sip.conf, and it STILL has lost those IPs in sip show peers
20:51.54jbalcombstoffell: ah, sorry. I'm thinking of 1.0.2.8
20:52.10*** join/#asterisk ToTo (n=ToTo@host61-133.pool874.interbusiness.it)
20:52.26stoffelljbalcomb, 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
20:53.26justinuip301 all upgraded... thx everyone
20:53.52gammacoderstoffell: I'm running 1.0.2.13 on my gxp2000 - nothing bad to report yet
20:53.57jbalcombstoffell: ok, I've got it set up on my phone "Program-- 1.0.2.13    Bootloader-- 1.0.2.3"
20:54.27stoffellgammacoder, nice to see it fixes the serious issues
20:54.41jbalcombstoffell: I'll be releasing it to the IT Dept now. Assuming no additional problem I'll have all 100+ on it next week.
20:55.09stoffelljbalcomb, same here, but i will first test it for 24hrs on 2 phones at least :)
20:55.19epabloManxPower:  That was it.. thanks!
20:57.48jbalcombstoffell: 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)
20:58.26stoffelljbalcomb; heh, aaah, okay :-) so will check back here in 24 hrs ;)
20:58.50jbalcombstoffell: right on
20:59.00[av]banianyone confirm the display corruption is fixed with 1.0.2.13?
20:59.12jbalcomb[av]bani so i guess GS getting those TFTP settings in for DHCP really helps out your project eh?
21:00.32Dr-Linuxmy few clients complaint that when they talked, there voice is too much static. what could be the problem?
21:00.38[av]baninot really, i figured out workarounds
21:00.46epabloWell guys i'm leaving.. c'ya
21:00.49*** part/#asterisk epablo (n=epablo@200.75.139.188)
21:01.02afrosheenDr-Linux: that'll happen with echo cancellation on and too much tx/rx gain boosting
21:01.11jbalcombDr-Linux: what phone?
21:02.17Dr-Linuxafrosheen: but someone suggest me echo cancellation should be "yes"  ?
21:02.36Dr-Linuxjbalcomb: they are using X-Lite softphone
21:02.39afrosheenDr-Linux: it should be, correct, but you may have to tame the rx/tx levels if you're using a zaptel interface
21:02.55afrosheenor 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)
21:03.09jbalcombDr-Linux: ok, afrosheen is right on then
21:03.29Dr-Linuxafrosheen: my rx is rx=1.0 and tx is tx=0.0
21:03.43afrosheenhave your users turn the mic sensitivity down then
21:04.04afrosheenthey're probably your typical shouters that have no concept of mic sensitivity
21:04.19Dr-Linuxhhm.. may be thats mic problem
21:04.23afrosheenwe have a few of those here
21:04.36afrosheenwell it's a problem of a hot mic level plus the echo canceller
21:04.43Dr-Linuxafrosheen: but my hardphone/US users are fine :S
21:05.09afrosheenwelp, it's easy to see the problem now
21:05.09cpmthat's because softphones suck.
21:05.27jbalcombstoffell [av]bani is the firmware prefix/postfix part of the directory structure or just the filename?
21:05.52Dr-Linuxhhm...
21:06.20Dr-Linuxafrosheen: actually i can't change the setting bcoz US users are doing fine.
21:06.32Dr-Linuxafrosheen: so what you suggest me?
21:06.41afrosheenyeah, SO, change the microphone sensitivity on your softphone users
21:06.47[av]banijbalcomb: no idea, never tried it
21:07.21stoffelljbalcomb; prefix? files are called gxp2000.bin etc.
21:07.36Dr-Linuxafrosheen: it can be changed on the user's system? or in the xlite setting?
21:07.49afrosheendefinitely on their systems, maybe in xlite's settings
21:08.11afrosheenwindows lets you set mic levels in the audio control panel, plus you can enable/disable a 20db mic boost
21:08.47Dr-Linuxok nice
21:09.04afrosheenI'm glad you mentioned your hard phone users were ok, that really narrowed it down
21:09.05Dr-Linuxafrosheen: what codec should i use for xlite users?
21:09.16afrosheenbandwidth permitting, ulaw or alaw
21:09.36afrosheenit does well with sip and has high call quality, handles dtmf inband nicely
21:09.56Dr-Linuxafrosheen: our xlite(pakistan) users have no good bandwidth thts why i'm asking
21:11.03afrosheenwhat are they on, dialup?
21:12.01Dr-Linuxafrosheen: 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
21:12.09Dr-Linuxbut all US users are fine ..
21:12.09afrosheeng729 is also very nice, it costs a little extra but gives impressive call quality for the bandwidth it consumes, plus it trunks very well...
21:12.23Dr-Linuxsome pakistan users complaining every day
21:12.27brad_mssweh, if you have any amount of packet loss, g729 is aweful
21:12.40brad_msswulaw copes much better with packet loss
21:12.53afrosheenyou 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)
21:13.14Dr-Linuxg729 doesn't support xlite
21:13.22afrosheenthe pro version has support for it
21:13.42brad_msswafrosheen: 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]baniilbc 4-evar
21:13.55Dr-Linuxafrosheen: but pro is not free
21:14.06kuku5Can I have a cisco phone send messages to syslog?
21:14.17afrosheenwell, if it's an international business, I'm assuming you have a few bucks to spend on phones
21:14.18Qwell[]kuku5: What kind of messages?
21:14.23kuku5debug
21:14.34Qwell[]messages from the phone, or from *?
21:14.35Dr-Linuxooo what about iLBC, its my pirority codec?
21:14.51afrosheenilbc, 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)
21:14.58kuku5it restarts by itself, or hangs, I need to find out what. When it hangs I get messages like:
21:15.07kuku5"Busy here"
21:15.13ManxPowerAnyone 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
21:15.24kuku5Qwell: Want to get debug from the PHONE
21:15.31Qwell[]ManxPower: The Feature button + 60..
21:15.37Dr-Linuxafrosheen: can you tell again your last line in easy english ? :$
21:15.40ManxPowerQwell, analog phones don't have a feature button
21:15.42Qwell[]ManxPower: Let me know if you figure out how to simulate the feature button though :P
21:15.51Qwell[]I've actually been trying to find that out too
21:16.07ManxPowerQwell, I found some of the codes at http://www.tek-tips.com/faqs.cfm?fid=5351
21:16.20Qwell[]I found all of the codes...but nothing says how to use feature
21:16.30afrosheenDr-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
21:16.32Qwell[]same problem you're having, I assume?
21:16.48ManxPowerQwell, I have to do a FLASH before dialing the code
21:17.00Dr-Linuxok
21:17.01Qwell[]oh?
21:17.02ManxPowerMostly I'm looking for someone that has done this to confirm
21:17.09ManxPowerBTW, LINK=FLASH
21:17.12Qwell[]trying to do it through * or something?
21:17.14Dr-Linuxafrosheen: 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)
21:17.38afrosheendunno, I've never tried SJphone
21:17.40ManxPowerQwell, 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
21:17.42Qwell[]ManxPower: I'm just a user, so "LINK=FLASH" is foreign to me...
21:18.07afrosheenregardless, g729 costs a little extra due to licensing..but it has many benefits
21:18.34Qwell[]ManxPower: Are you saying that if you have a phone, you just do a flash, then dial the feature, or?
21:18.44Dr-Linuxafrosheen: i have g729 codec, but don't have softclient for it :P
21:18.49Qwell[]because if I could figure it out, it would save me a few headaches...
21:18.52ManxPowerfrom the URL I pasted: All access codes must be preceeded with a Flash Hook (Link)
21:18.56Qwell[]ahh
21:19.09Dr-Linuxafrosheen: my both of servers are running fine from last 3 months
21:19.25ManxPowerQwell, if you have an ATA2 or EATA (converts Norstar phone ports into analog ports) yes, you FLASH+code
21:19.37Qwell[]ManxPower: I've just got a nortel phone sitting on my desk
21:19.44Dr-Linuxtoday i got a complain, that one of my user mailbox is full ..
21:19.47Qwell[]not trying to use any thirdparty stuff
21:19.47ManxPowerQwell, that's prolly a nortel DIGITAL phone
21:19.54Qwell[]still no feature button on it ;/
21:20.09ManxPowerQwell, can you plug an analog phone into the line?
21:20.12Dr-Linuxso i was trying trying all day googled to findout how to increase the quota ..
21:20.23Qwell[]umm
21:20.31Qwell[]doubt it...haven't tried
21:20.43Dr-Linuxafrosheen: then finally i findout :)
21:20.44Qwell[]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)
21:21.16afrosheenincrease what quota?
21:21.37*** join/#asterisk Eitch (n=hugo@unaffiliated/eitch)
21:21.52Dr-Linuxafrosheen: user's voice mailbox
21:22.11Qwell[]flash isn't doing a whole lot, heh
21:22.28Qwell[]I freaking hate these phones.  No freaking mute button on them
21:22.32Dr-Linuxtoday i got to know, bydefault user can have 100 voicemails in his/her account
21:22.39asteriskmonkeythe citel gateways rule :D
21:22.47Qwell[]there is a "handsfree mute", but that puts you in speaker, and mutes that mic
21:22.51ManxPowerDr-Linux, actually 99
21:22.53asteriskmonkeyDr-Linux just change the limit to 1000 :D
21:23.08Dr-Linuxif it's increase caller will be listed, "sorry, the users can't accept more messages"
21:23.25ManxPowerAcutally since they start at 0, yes it is 99
21:23.31Dr-Linuxyep
21:23.36ManxPowerAnyone that has 99 messages in their mailbox is a moron and should be shot.
21:23.39afrosheenyeah. 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
21:23.42Dr-Linuxbut actually i didn't know how to change it :)
21:24.05afrosheenManxPower: you should bring a shotgun over here
21:24.05ManxPowerDr-Linux, voicemail.conf.sample is your FRIEND.
21:24.08asteriskmonkeyi would go that far some people like savign them for legal reasons
21:24.12Qwell[]ManxPower: Got anything else in that bag of tricks of yours, for me?
21:24.24ManxPowerQwell, get rid of the piece of crap phone
21:24.37ManxPowerYou obviouly have a digital set and I can't help with that
21:24.40Qwell[]ManxPower: indeed
21:24.49Qwell[]we're gonna try to get a PRI off the AT&T in the other building
21:24.53afrosheenasteriskmonkey: that's why our server emails the vm to each user..they can save them locally rather than on the server
21:25.17Qwell[]there are only two depts still on this nortel.  Mine, and securitybuilding management
21:25.30Dr-Linuxyep, i saw a help on WIKI and i added an option in voicemail.conf maxmsg=500  :)
21:25.35Qwell[]s/yb/y\/b/
21:25.39Qwell[]silly boy
21:25.40Qwell[]bot
21:26.03afrosheens /silly boy/silly bot
21:26.10afrosheenhe's not paying attention today is he
21:26.21Qwell[]he is...if it's "jbot valid" regex
21:26.42*** join/#asterisk Dandan (i=dandan@ellie.pacanka.com)
21:27.47Dr-Linuxasterisk doesn't have voice recognition app yet?
21:28.02ManxPowerDr-Linux, if you read the .sample files you'll fine all sorts of cool options
21:28.11Qwell[]Dr-Linux: voice recognition isn't something somebody "just writes"
21:28.34afrosheenit'll cost an arm and a leg when it's available, companies pay huge money for that feature
21:29.38Dr-LinuxManxPower: is there some new voicemail option after 1.2.1 version?
21:29.56asteriskmonkeyyou can always try sphynx for vr
21:30.08*** join/#asterisk redondos (n=redondos@190.48.41.94)
21:30.31Dr-Linuxasteriskmonkey: yes i installed that, but that doesn't work for me
21:30.43asteriskmonkeyit required making of custom dictionaries
21:30.49asteriskmonkeyits uver time consuming setting it up
21:31.40Dr-Linuxactually i don't understand what custom is in *
21:32.20Dr-Linuxi saw many sample extensions.conf files, there is wide of use this work "custom" but i don't understand what's this
21:33.04asteriskmonkeyin linuix or unix type at the cli "dict custom"
21:34.13*** join/#asterisk gammacoder (n=chatzill@64-132-192-33.gen.twtelecom.net)
21:34.26Dr-Linuxi'll in office
21:34.42jbalcombI *heart* Grandstream. "Allow DHCP Option 66 to override server" is perhaps an unfriendly description of the option?
21:35.00afrosheencustom means 'different than what you get to begin with', so if you buy a car and change the wheels, bam, you have 'custom wheels'
21:35.00Qwell[]jbalcomb: option 66 is tftp I believe
21:35.30Qwell[]which is very, very useful on cisco phones
21:35.37ManxPowerDr-Linux, no, MAJOR releases
21:35.41Qwell[]plug a brand new phone in...bam, it works
21:35.58jbalcombQwell[] 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"
21:36.16Dr-Linuxafrosheen: yes that i know, but how it works in *
21:36.20Qwell[]jbalcomb: any admin should know what 66 is :)
21:36.24jbalcombQwell[] Yeah, seems like between GS and [av]bani the GXP-2000 will be there soon
21:36.26gammacoderGrandstream's DHCP Option 66 isn't enabled by default - you have to enable it for each phone - kinda defeats the purpose of auto-provisioning
21:36.49ManxPowergammacoder, Much like the VLAN setting for polycom phones.
21:36.53jbalcombQwell[]: haha.. agreed. ;) too bad everyone isn't an admin.
21:36.53afrosheenDr-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
21:36.56Dr-Linuxanybody is using SPA-2100 ?
21:37.03ManxPowerDr-Linux, they are great
21:37.21Dr-Linuxafrosheen: great
21:37.41*** join/#asterisk jpablo (n=jpablo@200.94.130.194)
21:37.46afrosheenso for example, Iloveshrimp.conf will have a line like #include extrashrimp_custom.conf
21:38.05Dr-LinuxManxPower: i have a problem with spa 2100, i forgot web interface admin password.
21:38.08afrosheenWhen Asterisk reads iloveshrimp, it sees it needs to read in the custom file as well
21:38.14ManxPowerDr-Linux, They make a good bookend
21:38.28ManxPoweryou tried the factory reset via the analog port?
21:38.40Dr-Linuxi need to do some modification from web, but i forgot the password
21:39.12Dr-LinuxManxPower: yes that i know, but i don't wanna reset the fectory
21:39.17Dr-Linuxlet me explain my problem
21:39.36jpablohey 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 ?
21:40.03Dr-LinuxManxPower: i got this spa 2100 from a VOIP provider, so it has 2 lines provisioned from them for always ...
21:41.06Dr-Linuxso whenever i need to configure it with my asterisk extension, i simpley turn of provisioned option, and add my asterisk extensions, then it work
21:41.13Dr-Linuxbut i forgot the password
21:41.47Dr-LinuxManxPower: so if reset the fectory i will lost the voip provider things too
21:41.50Dr-Linuxwhat you say? :S
21:42.07afrosheenyeah reset it then call your ISP saying lightning struck a tree in your yard and reset your phone
21:42.29afrosheens /ISP/Voip provider
21:43.10Dr-Linuxafrosheen: 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)
21:45.00ManxPowerDr-Linux, What I say is that I've never lost the password to the SPA-2100
21:45.04Dr-Linuxso all i need is to login to web interface and disable there provisioned option temprary :)
21:45.09afrosheenjeez you gotta be the cheapest guy I ever met
21:45.14afrosheen;)
21:45.30ManxPowerDr-Linux, Did the box come preconfigured from the provider?  If so, they can lock you out.
21:46.17Dr-LinuxManxPower: yeah it is, but they didn't lock it out since 3 months
21:46.37Qwell[]doesn't mean they didn't lock you out
21:47.18[TK]D-FenderDr-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)
21:48.00EgonisDumb... 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
21:48.15Qwell[]Egonis: idiots.  no, you can't
21:48.25Dr-Linux[TK]D-Fender: i agree sir
21:48.35Qwell[]It uses SIP, but it's against their TOS to use anything but what they give you
21:48.45afrosheenvonage != *
21:48.52Dr-Linux[TK]D-Fender: i'm the one who ever set the admin password :)
21:49.18afrosheenwait, how do you express not compatible with... !+= ?
21:49.27Dr-Linuxactually we bought this device, if it works with data application. but i doesn't work
21:49.31Qwell[]Egonis: Get another provider..
21:49.46ManxPowerEgonis, if you have a Vonage SOFTPHONE account you can.  Vonage uses SIP.
21:50.12Qwell[]it's still against their TOS though, isn't it?
21:50.17ManxPowerNotFreak, there is no unlimited calling on the softphone account.
21:50.27*** join/#asterisk nurfe (n=rgff@h24-207-10-162.dlt.dccnet.com)
21:50.35EgonisManxPower: ooh, handy! ty
21:50.41ManxPowerQwell, maybe, but since the account is not unlimited......
21:50.54ManxPoweroh, and you have to have a normal account in order to get the softphone account
21:51.13ManxPowerCan you guess where I learned this?
21:51.17ManxPowerFROM THE MAILING LISTS!
21:51.25ManxPowerThe ASTERISK mailing lists.
21:51.36*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)
21:51.41Egonisexit
21:51.58afrosheenlooks like Egonis spends a little time with a shell open
21:52.31iqHi, I've a TE110P. Do I need rob-bit T1 or ISDN T1? I will prefer rob-bit T1.
21:52.48fu3ManxPower.. another fine example of why you're the best.
21:53.15ManxPowerfu3, I'm available for consulting if you throw reasonably large piles of cash at me.
21:53.47ManxPoweriq, You need whatever your line needs.
21:53.59ManxPowerBut ALWAYS ISDN PRI is best
21:54.07ManxPoweraka EuroISDN or ISDN110
21:54.18fu3I dont know of many problems that cannot be solved with large amounts of cash.
21:54.25ManxPowerfu3, exactly.
21:54.26fu3but thanks :)
21:54.39[TK]D-Fenderfu3 : Or really REALLY, REALLY ... big lasers :D
21:54.44fu3hahahaha
21:54.59iqManxPower: so the board will auto detect the signaling?
21:55.16ManxPoweriq, no.  you have to tell the board what signaling your line requires.
21:55.22ManxPowerBut if you can, ALWAYS order PRI
21:55.26*** join/#asterisk cogineo (n=soner@81.215.105.114)
21:55.49ManxPowerIf 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)
21:58.37iqManxPower: Do I define it under /etc/zaptel.conf or /etc/asterisk/zapdata.conf ?
22:00.26stoffellnow, i know a bit about Mark Spencer creating *, but anyone knows his age more or less? :)
22:01.12Qwell[]stoffell: Ask him
22:01.39stoffellok 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)
22:01.52Qwell[]stoffell: Ask him here
22:02.30Qwell[]stoffell: really though, I'm pretty sure he's got a bio or two online :)
22:02.46stoffellhm, ok, will google for 'm Qwell ;)
22:03.03Qwell[]*cough*stalker*cough*
22:03.05Qwell[]:P
22:03.28stoffelllol, you wouldn't believe why i'm asking this..
22:03.37Qwell[]probably not
22:03.40Qwell[]so...do tell
22:03.51*** join/#asterisk jyukes_ (n=jameshot@c-69-248-195-94.hsd1.nj.comcast.net)
22:03.55stoffellmy gf is asking, because i'm dragging her to his speech on fosdem ;)
22:04.05fu3hahaha
22:04.09fu3what a stalker ;)
22:04.15fu3thats what the last six stalkers said
22:04.28stoffelloh boy, gotta change my nick now.. :p
22:04.31fu3haha
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)
22:05.13fu3ok.. even using an oscilloscope, I cannot detect a 500ms battery drop upon hangup.
22:05.19stoffelloh boy, apr 1977.. tnx Qwell for the hint ;)
22:05.19fu3so I doubt thats happening
22:06.55paulhuynhhello
22:06.57paulhuynheveryone
22:08.01*** join/#asterisk Morex (n=blah@host86-140-246-153.range86-140.btcentralplus.com)
22:08.11MorexHello all
22:08.31MorexAnyone have any thoughts on how an agent might indicate to the system that he/she is wrapping up a call?
22:08.40Qwell[]Morex: by hanging up
22:08.52MorexWon't they get the next call straight away?
22:08.59stoffellQwell, i was thinking the same but didn't dare to type it .. :p
22:09.16Qwell[]Morex: yes.  there is a wrapuptime option though for that
22:09.26MorexWhat if not every call requires a wrap up?
22:09.55Qwell[]have a pause button on your phone?
22:10.03MorexNope
22:10.06paulhuynhi'm using asterisk@home and i did an upgrade
22:10.23Qwell[]Morex: I mean...add one
22:10.27paulhuynhhow can i merge the old cdr from the old server to the new old
22:10.29MorexLOL
22:10.33Qwell[]have something that calls PauseQueueMember
22:10.42MorexFrom inside AgentLogin?
22:10.50Qwell[]dunno, I don't use agents
22:11.08MorexAh OK
22:12.17MorexSo no obvious solution then...
22:12.39[TK]D-Fenderpaulhuynh : 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-FenderPauseQueueMember works on staticly defined agents....
22:13.36Qwell[]well then
22:13.44MorexYou can get it to work with dynamic agents too
22:13.54MorexBut how do I activate it from inside a call?
22:14.15[TK]D-FenderMorex : You could do a featuremap to it with features.conf I suppose
22:14.37[TK]D-Fenderor 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>
22:15.13Qwell[]or set DND
22:16.14[TK]D-FenderDND 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-FenderAt the minimum it will waste an agent-cycle on you.
22:16.32*** join/#asterisk gomez_ (n=gomez@se400.pppoe03-571.bih.net.ba)
22:16.42gomez_hello
22:16.51MorexI was thinking of putting every call into a long wrapup, then have them exit from it some how
22:16.55gomez_is there anyone with gpx-2000?
22:17.03stoffellgomez_, what you want to know?
22:17.13gomez_well, i have problem
22:17.18gomez_with my phone
22:17.21stoffellshoot
22:17.33gomez_i have registered extension
22:17.39gomez_everything
22:17.47*** join/#asterisk DeadZen (n=DeadZen@adsl-2-119-219.mia.bellsouth.net)
22:17.50[TK]D-FenderMorex : give them a big enough wrapup to handle "average" and let them make the choice to go on "pause"
22:17.52gomez_is set up
22:18.00gomez_but phone is still
22:18.15gomez_i meen user on line 1 is still Not registered
22:18.36DeadZenis there anyway to make asterisk call you? with like just a audio message
22:18.46stoffellgomez_, try asterisk -r -> sip show peers
22:19.41TamariskAre 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.
22:20.12DeadZenreally neat
22:20.37mikefooTamarisk: training book is available via the www
22:20.43Dr-Linuxanybody is using Wakeup call?
22:20.44mikefoo'world' wide web  :)
22:20.58*** join/#asterisk zeedo (n=zeedo@80-192-53-14.stb.ubr04.glen.blueyonder.co.uk)
22:21.00TamariskIs that the Oreilly book on line?
22:21.19Dr-Linux~thebook
22:21.21jbotmethinks thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online!
22:21.21mikefoothats the recommended one
22:21.46DeadZen_0_0_: is there docs on .call files?
22:22.12Qwell[]DeadZen: Yes
22:22.18DeadZenasteriskdocs.org ?
22:22.23TamariskI have downloaded that one, unfortunatly still find it hard going trying to work through it
22:23.07paulhuynhwhat is the channel for amp
22:23.18TamariskI 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
22:23.25Tamariskmessage
22:23.32DeadZenQwell: where?
22:23.33zeedoTamarisk: yeh it highlights on your username
22:23.38Qwell[]~docs
22:23.39jbothmm... 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
22:23.47Qwell[]DeadZen: Pick one...
22:24.01DeadZenhopefully its in the pdf
22:24.10Tamariskzeedo, so this should be highlighted to you, yes?
22:24.16zeedoTamarisk: yep
22:24.23TamariskI can learn
22:24.28zeedoTamarisk: it's configurable in your client though, so you can switch it off
22:24.37paulhuynhanyone here use sixtel or cvc?
22:24.41_0_0_try here: www.voip-info.org/wiki-Asterisk+auto-dial+out
22:24.47mikefoopauldy: I was looking at sixtel
22:24.58mikefoogoing to signup this weekend probably.
22:25.24TamariskTo be honest I hate IRC, but it does span continents with great speed, but sometimes my typing abilities and manors can and do affend
22:25.36Tamariskoffend!
22:25.53DeadZenmanners!
22:26.25mikefooI need a realiable inbound voip provider, in the US, anyone have a recommendation?
22:26.30mikefoowas looking at asterlink
22:26.35Tamariskthere you go English was never a good subject for me, I need to enable spellcheck
22:26.37Qwell[]mikefoo: asterlink is good
22:26.39paulhuynhsixtel is good service
22:26.46Qwell[]right file?
22:26.48paulhuynhbut not very good when it come to support
22:26.48bweschkemikefoo: 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)
22:28.19mikefoobweschke: any idea if I can have simultaneous inbound calls?
22:28.31mikefooI would need unlimited to some extent.
22:29.14mikefoopaulhuynh: 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)
22:29.15Dr-Linuxafter 35 days i did "reload" today
22:30.04Dr-Linuxi afraid to do "reload" if may be it nver come back!
22:30.17stoffellDr-Linux , hehe :)
22:30.24mikefoolol
22:30.36Dr-Linuxlol
22:30.46Dr-Linuxreload is not neccesary, right?
22:31.00stoffellDr-Linux; only if you change something
22:31.19mikefooreload just re-reads config files
22:31.24Dr-Linuxstoffell: well, if i change then also i do not reload :P
22:31.48stoffellDr-Linux; well, that's a good way to insure stability, ..... if it never goes down.. :)
22:31.54Dr-Linuxi only reload specific file
22:32.34stoffellDr-Linux you must have a solid dialpattern if you didn't reload for 35 days :)
22:32.41MceHi, 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?
22:32.57Qwell[]wardialer app?
22:33.03Mceiwar specifically
22:33.11mog_workyeah you can use asterisk to bulk call people
22:33.12Qwell[]for telemarketting purposes?
22:33.13Dr-Linuxstoffell: why? what do you mean ?
22:33.14paulhuynhMCE
22:33.17Qwell[]if so, get out
22:33.20Mceno
22:33.20paulhuynhPM me we will talk
22:33.33Mcenot for that
22:33.36stoffellDr-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)
22:34.11Dr-Linuxstoffell: well, i reload but i do not use "reload" command
22:34.46Dr-Linuxstoffell: if just use  "extension reload"  or "sip reload"  anyone if needed
22:34.55Dr-Linuxi do not bother entire reload
22:35.18stoffellDr-Linux; okay, indeed a good practice, .. entire reload is useless in that case
22:35.21Dr-Linuxbcoz when reloading...... during that time .. my heart also reloads :P
22:35.39stoffell:)
22:35.43MorexSolution to queue issue turns out to be set ackcall=always
22:35.57MorexThis allows agents to press # before accepting calls even in AgentLogin
22:36.06MorexIt's in the source code but undocumented elsewhere.
22:36.13Dr-Linuxstoffell: somethime i think, i should remove all other extra config files :S
22:36.31Qwell[]Dr-Linux: Just noload the other modules
22:37.22Dr-LinuxQwell[]: great
22:38.04[TK]D-Fenderstoffell : No, its a question of not asking for more than you've got :)
22:38.06Dr-Linuxin which case asterisk process gets auto killed, and it needs start again? :S
22:40.45*** join/#asterisk LeonardoCabelo (n=leonardo@unaffiliated/leonardocabelo)
22:40.53DeadZenasterisk is the coolest thing since sliced babies
22:41.12FuriousGeorgesure, if your into that sort of thing
22:41.18DeadZenlol im j/k
22:41.21*** join/#asterisk bjohnson (n=bjohnson@i216-58-92-216.cybersurf.com)
22:41.23DeadZeni think i got it to out dial me
22:41.26DeadZenim trying to leave a msg now
22:42.02FuriousGeorgemake sure you remember to leave yourself your number
22:42.05DeadZenit says my Status is OK (127 ms)
22:42.15DeadZenbut i can't make my sjphone ring yet
22:42.15DeadZenheh
22:42.22Dr-Linuxi wish to have bandwidth over here .. aww
22:42.37LeonardoCabeloSomebody know how to configure the G668 Soyo HardPhone in the asterisk
22:42.48DeadZenhehe
22:42.50DeadZenhard phone
22:42.55Dr-LinuxDeadZen: you are using SJphone?
22:42.59DeadZenyes
22:43.11DeadZentried using a sip component for delphi to make my own
22:43.11FuriousGeorgeanyone know why it is that when there is an unavailable CID from my pots provider asterisk assigns "Asterisk" to the CID?
22:43.18FuriousGeorgeanyway to replace that with an unknown
22:43.18DeadZenbut it says my status is unavailable so i can only dial out
22:43.20DeadZenboo
22:43.26Dr-LinuxDeadZen: what codecs SJphone supports ?
22:43.36FuriousGeorgeanother 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)
22:43.51DeadZenwell gsm ulaw for sure
22:44.01DeadZenresti dunno.. i suppose alaw
22:44.05ManxPowerI always force the CLID in sip.conf.  Never trust the phone
22:44.19ManxPoweror the user for that matter
22:44.20afrosheenDr-Linux: yeah now that I think about it, you should use gsm
22:44.22Dr-LinuxDeadZen: does it support g729 ?
22:44.32DeadZeni dunno
22:44.34FuriousGeorgeManxPower: CLID?  did you mean CID or were you not talking to me?
22:44.35ManxPowerfree softphones do not support G729
22:44.40DeadZeni think they mostly do 726
22:44.47ManxPowerFuriousGeorge, Calling Line ID aka Caller ID
22:44.55afrosheenthe quality kinda stinks but it's free, and anyone used to talking on a cellphone won't notice the quality
22:44.55DeadZenwhy do you need 729
22:45.07DeadZenquality superior?
22:45.12FuriousGeorgeManxPower: i didnt know you could set that in sip.conf, i guess i stick it in the general section?
22:45.26Dr-Linuxafrosheen: yep gsm eats 13 kb, but with network header etc i reachs to 30 kb
22:45.36FuriousGeorgeanyone ever hear of externhost not to be confused with sternip?
22:45.37afrosheenI advised him earlier to try it, it does well with limited bandwidth, trunks well, etc.
22:45.42Dr-LinuxDeadZen: we don't have fuckin bandwidth here
22:45.44ManxPowercallerid=Murphy, Tonda <9857683227>
22:45.47afrosheenlol
22:45.55FuriousGeorgeManxPower: im talking about for incoming
22:45.55ManxPowerin each sip [whateverpeerfrienduser] section
22:45.55DeadZeni dunno sjphone has shitty quality
22:45.56afrosheenfunny when foreigners curse :)
22:46.05TamariskAn 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?
22:46.06DeadZenalmost unusable
22:46.09ManxPowerFuriousGeorge, incoming from an ITSP or something, not incoming from a phone?
22:46.09[TK]D-FenderFuriousGeorge : Yes, I know people who use EXTERNHOST, and others that use EXTERNIP.  What of it?
22:46.14FuriousGeorgewhenever its unknown i get Asterisk@10.0.0.10
22:46.26Dr-Linuxafrosheen:: but spa hardphone doesn't work with GSM
22:46.37FuriousGeorge[TK]D-Fender: wondering what the difference is.  im trying to get * not to cache my *.dynu.com dynamic ip'ed boxen
22:46.37mikefooAnyone care to buy a 7960?
22:46.37afrosheenoh man that blows
22:46.38DeadZenand thats the day Curious George became
22:46.40DeadZenFurious George!
22:46.54ManxPowerThe only low bandwidth codec most phones support is G.729
22:46.58trixterwhat that the day when curiours george was no longer curious?
22:46.59FuriousGeorgeManxPower: yeah, unknown cid from PSTN come in as Asterisk@10.0.0.10
22:47.01FuriousGeorgeor something
22:47.12ManxPowerFuriousGeorge, Ah.  No idea.  I don't use SIP for PSTN access.
22:47.13[TK]D-FenderFuriousGeorge : Use higher rate of EXTERNREFRESH to keep it udated
22:47.16DeadZenoh thats dick cheney
22:47.18afrosheenFuriousGeorge: you can set a system default CID can't you
22:47.22FuriousGeorgeone rig even just randonly displays one of my zap CIDs on unknwn
22:47.22Qwell[]mikefoo: I'll give you like $100
22:47.33Qwell[]assuming it works
22:47.48ManxPowerit would be pretty trivial to do a match on that callerid and reset it to something you want.
22:47.54FuriousGeorgeafrosheen: i would think so, i know i can put it in the dialplan logic, but i'd think there was an option
22:48.00Dr-Linuxbut iLBC CPU hungry .. but i have a strong server with daul processor
22:48.15ManxPowerFuriousGeorge, I'll bet you could change it in chan_sip.c
22:48.20FuriousGeorge[TK]D-Fender: externrefresh where?  in sip.conf?  cuz my problem is with my iax register=> to my dynu.com boxes
22:48.25Qwell[]guess not
22:48.36DeadZenanyone ever heard of astatech ?
22:48.41[TK]D-FenderFuriousGeorge : Not sure about IAX.CONF.
22:48.49DeadZenthey make a .net and delphi sip component
22:48.59Qwell[]delphi?  wtf?
22:49.20DeadZenyah delphi.. you know that thing before .net
22:49.21DeadZenheh
22:49.24FuriousGeorge[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
22:49.28Qwell[]that thing before like...1990?
22:49.39mikefooQwell: ofcourse it works, its sccp, as of now, are you able to flash it to anyother protocol if needed?
22:49.39DeadZennah they kept it up till bout 2005
22:49.49DeadZenthey're selling it though
22:49.50Qwell[]mikefoo: of course.  I'd use sccp though
22:49.56DeadZenand borlands doing life cycle management
22:50.03[TK]D-FenderFuriousGeorge : Maybe if you set it up in parallel in SIP.CONF it might take it...
22:50.40mikefooQwell: ok I will get back to you in a day or so.
22:50.43rayvdI'm a sturgeon.
22:50.54Qwell[]mikefoo: alright, I'll be around
22:50.55FuriousGeorge[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
22:50.56FuriousGeorge?
22:50.58DeadZenhahah
22:51.02DeadZenthat was soo cool!
22:51.08DeadZenI just called myself and said You sound cute!
22:51.18DeadZenhaha i just made my day
22:51.34DeadZennow to make a goofy dial plan and play tricks on my buddies
22:51.38[TK]D-FenderFuriousGeorge : No, just try setting the externhost & externrefres in SIP.CONF and that may keep it from caching in iax.conf.
22:51.59FuriousGeorge[TK]D-Fender: ill let you know if that worked next time my ip's change
22:52.10[TK]D-FenderDeadZen : I havea special exten that changes my outbound callerid to 867-5309 :)
22:52.12ManxPowerDeadZen, must be a telecom newbie
22:52.18FuriousGeorgeis there any way to not allow my sip clients to display the domain on the CID?
22:52.35DeadZenthats cool fender
22:52.35ManxPowerFuriousGeorge, preprocess the calls
22:52.41DeadZenyah I'm new to telecom
22:52.49DeadZenbut I'm a good app programmer
22:52.51ManxPowerI've found that you really have to preprocess calls
22:52.52DeadZengonna try to mix em
22:53.03FuriousGeorgeManxPower: you mean answer() and check the cid by hand, as it were?
22:53.09ManxPowerwhy answer?
22:53.42ManxPowerFuriousGeorge, 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.
22:53.44FuriousGeorgewell, not necessarily requiring an answer, but basically set the cid myself
22:54.02FuriousGeorgenp
22:54.05FuriousGeorgeill work on that too
22:54.47ManxPowerI 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)
22:55.01FuriousGeorgeDeadZen: set your cid to all your friends mom's / girlfriends and call them all day till they complain to your provider
22:55.16ManxPowerFuriousGeorge, for example in the USA all callerid should be 10 digits long.  If it's not then fix it up
22:55.16FuriousGeorgeManxPower: makes sense
22:55.24[TK]D-FenderManxPower : 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)
22:56.10Qwell[]just 9+ them
22:56.26Qwell[]9+7, 9+11
22:56.31ManxPower[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
22:56.54ManxPowerwith a few exceptions like 911 should not REQUIRE 9-911
22:57.11Qwell[]BUT...9-911 SHOULD be valid
22:57.13Qwell[]as should 19911
22:57.17ManxPowerQwell, of course
22:57.18Qwell[]erm, 91911
22:57.24[TK]D-FenderQwell : 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.
22:57.25Qwell[]or something
22:57.33ManxPowerwell 91911 should not be valid
22:57.36ManxPower911 and 9911
22:57.44Qwell[]91911 makes sense though
22:57.52ManxPower[TK]D-Fender, users hate to wait.
22:57.58Qwell[]it won't be a valid exten anyhow, so...
22:58.01ManxPowerMy users complain ALL THE TIME about that.
22:58.22[TK]D-FenderManxPower : true... but they'd balance that against the ability to sue the missed calls list to INSTANTLY dial DL #'s and save a LOT.
22:58.25FuriousGeorgewhy use 9 at all anymore?
22:58.30ManxPoweryes, but 9 + 1 + 916 + number could be misdialed as 91911
22:58.31[TK]D-FenderManxPower : Its a trade-off for sure...
22:58.47Qwell[]ManxPower: yes, true, but as could 916-xxxx
22:58.49ManxPowerFuriousGeorge, use 9 so users don't have to wait for Digittimeout before calls to extensions are processed
22:59.03ManxPowerAnd I have to have a long digittimeout or users will yell.
22:59.06Qwell[]assuming 916 is a valid prefix
22:59.09[TK]D-FenderFuriousGeorge : 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.
22:59.19Qwell[]or, no
22:59.25Qwell[]any areacode that starts with 6
22:59.31Qwell[]so, 9-1-626-xxx-xxxx
22:59.32DeadZencan e911 trace the call
22:59.41Qwell[]ManxPower The above could just as easily be misdialed
22:59.48FuriousGeorgethat makes sense.  i didnt think about it b/c we are moving to sipphones
22:59.54FuriousGeorgewhich you gotta send anyway
23:00.06*** join/#asterisk Kizmet (n=Kizmet@freematrix/sponsor/kizmet)
23:00.38ManxPowerMy 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
23:00.39Qwell[]91911 kinda makes sense to me.  Users know that in order to dial long distance, they have to do 91-
23:00.44FuriousGeorgewoot, forensic files is on
23:00.52ManxPoweryeah, but 911 is not a toll call
23:00.55Qwell[]so, in a panic, I could realistically see 91911 be dialed
23:01.10DeadZenwhats the number for 911?
23:01.16Qwell[]or just 1911
23:01.20trixterthe real number is 912
23:01.32ManxPowerReally, I should add a Wait(1) for all 911 calls, so users can panic and hangup when they realize what they did.
23:01.34Qwell[]ManxPower: I'm just more paranoid I guess. :)
23:01.35DeadZenwasnt that off the simpsons
23:01.36ManxPowerI think 916 is Toronto
23:01.42DeadZenwhen he became a mason
23:01.46FuriousGeorgecant you see yourself somehow beeing sued too b/c someone didnt know to dial 91 before the 911?
23:01.50trixterno a stonecutter
23:01.52trixterthey are different
23:01.56DeadZenahh
23:01.59DeadZensup trixter
23:02.00Qwell[]FuriousGeorge: No, because 911 and 9911 would also work
23:02.05trixternot much
23:02.06FuriousGeorgeah
23:02.17DeadZenim trying to put sip into my app
23:02.18DeadZenhehe
23:02.18KizmetI have a 'You are now being connected to emergentcy services.'
23:02.18rayvd916 area code?
23:02.20Kizmet:)
23:02.27*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
23:03.01FuriousGeorgeManxPower: 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?
23:04.57FuriousGeorgei just dont see why showing the domain is the deault behavior.  i know where i'm at no one cares
23:05.16FuriousGeorge* wasnt even dopmain aware till like 6 months ago, right?
23:06.10*** join/#asterisk Nukemizer (n=Nuke@67.137.28.165)
23:06.59Kizmethow would i do a dial plan match for 911 and 000 in the same line ?
23:07.07Kizmeti was thinking like [000|911]
23:07.09Kizmetor somthing
23:07.18Qwell[]Kizmet: You wouldn't
23:07.21[TK]D-FenderKizmet : Just easier to make one Goto the other.
23:07.39Kizmet:(
23:07.46KizmetG.H.E.Y!
23:07.50Dr-Linuxwhat's 911 in asterisk? do i need to define this extension manually or its some default option?
23:08.08FuriousGeorgedefault :)
23:08.21*** join/#asterisk brockj49464 (n=brockj49@63.87.56.235)
23:08.56Dr-LinuxFuriousGeorge: how come defualt? :S
23:09.17[TK]D-FenderDr-Linux : Ummm, what kind of question is that?!  You decide EVERYTHING yourself.  You should kno better...
23:09.23FuriousGeorgeand set your zipcode in asterisk.conf and the number for the local pizza place is stored in ${TAKEOUT}
23:09.34Kizmet<PROTECTED>
23:10.00[TK]D-Fender.. and did you know the word "gullable" isn't in the dictionary? ;)
23:10.02FuriousGeorgei the extension 000 or s?
23:10.08FuriousGeorgelemme go look
23:10.10[TK]D-FenderKizmet : thats because you did it BACKWARDS.
23:10.16Kizmeti did :o
23:10.17FuriousGeorgeon dictionary.com
23:10.26[TK]D-Fender"s" is not a priority.....
23:10.45[TK]D-FenderFuriousGeorge : Guess you already fell for it... CHUMP :)
23:10.51FuriousGeorgeLIAR
23:10.58FuriousGeorge(gets me every time)
23:11.01Dr-Linuxgullable? :S
23:11.53Kizmet<PROTECTED>
23:12.08Dr-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-FenderKizmet : Show us the WHOLE dialplan not just 1 stupid line from it....
23:12.28NivexI really ought to put a 911 trap in my dialplan that does Playback(no-911-1)
23:12.38Dr-Linuxactually this 911 is new here in pak
23:12.44*** join/#asterisk Seldon1975 (n=someone@toronto-HSE-ppp4239697.sympatico.ca)
23:12.56Kizmet[TK]D-Fender, If i was to do that it would span about 48 pages on pastebin -_-
23:13.14[TK]D-FenderKizmet : How about eveything in that CONTEXT then?
23:13.16Seldon1975sorry 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?
23:13.32NivexSeldon1975: show version
23:13.33Kizmetheh that is everything in that context
23:13.33[TK]D-FenderSeldon1975 : "show version"
23:13.34Kizmetheh
23:13.39Seldon1975thanks
23:13.41[TK]D-FenderKizmet : just do it.
23:14.14Kizmet[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 ;)
23:14.27Seldon1975can it tell you the zaptel version?
23:15.08Seldon1975Kizmet, D-Fender; can the * console tell you the zaptel version?
23:15.09[TK]D-FenderKizmet : 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-FenderSeldon1975 : no clue.
23:15.22Seldon1975i mean Nivex :{
23:15.28[TK]D-FenderKizmet : Fine start with those 2.
23:15.36Seldon1975D-Fender is there ay way you know of to check the Zaptel version
23:15.45ManxPowerKizmet, what verison of Asterisk?
23:15.49Kizmet[TK]D-Fender, lol the other pages are simply mappings to different states in america, australia, uk, etc etc.
23:15.51Kizmet1.2.4
23:16.01[TK]D-FenderKizmet : 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)
23:16.07ManxPowerKizmet, and that's in extensions.ael?
23:16.09[TK]D-FenderKizmet : So pastebin away...
23:16.14KizmetManxPower, yes.
23:16.22DeadZenjesus
23:16.32DeadZenvaxvoip charges 5500 for their sip ocx
23:16.44DeadZen20 simultaneious calls 1500.. . 100.... is 5500
23:16.47ManxPowerVAXen were always expensive
23:16.48*** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar)
23:16.56DeadZenwhat the hells wrong with them
23:17.00Kizmethttp://pastebin.com/567632
23:17.22ManxPowerDeadZen, thats pretty standard pricing for telecom
23:17.34DeadZenseriously?
23:17.37ManxPowerNortel wants like $8,000 for a PRI card and the software for their stuff
23:17.44ManxPowerDeadZen, Yeah.
23:17.52DeadZendamn
23:17.56DeadZeni picked a good business to get into thenm
23:17.57[TK]D-FenderKizmet : Your goto is no good.
23:18.11Kizmet[TK]D-Fender, post a revision maybe :)
23:18.21ManxPowerDeadZen, that's for a ONE PORT PRI card
23:18.30[TK]D-FenderKizmet : You are trying to jump to a CONTEXT named "000" and there is no "s" as an exten in this case.
23:18.43Kizmetwhoops :S
23:18.50ManxPower[TK]D-Fender, is the context required for that?
23:19.11ManxPowerMust be an AEL thing, since you only need a context in a Goto if it's jumping out of the current context.
23:19.14[TK]D-FenderManxPower : Certainly not.  the target is WITHIN the context.
23:19.40ManxPower<PROTECTED>
23:20.01[TK]D-FenderYup... kinda blatent isn't it :)
23:20.07ManxPower1 arg = priority, 2 args = exten and priority, 3 args = context, exten, and priority
23:20.17[TK]D-FenderThey keep doing that.... hiding it in the BIG print :)
23:20.40ManxPowerso his goto is going to extension 000, priority 1 in the current context
23:21.25[TK]D-FenderManxPower : actually in AEL does priority even exist?  If only by a "label" if present?
23:21.43ManxPower[TK]D-Fender, I have no idea.  AEL is still too new for me to trust it enough to use it.
23:22.23ManxPowerbut I do admit that I totally lust for AEL
23:23.21[TK]D-FenderManxPower : And even if it WERE stable... who cares??  it doesn't offer anything NEW, it just shortens up the code a little and introduces all new ways to make debugging a ^&%#$ PITA :/
23:23.21ManxPowerI suppose I should write my dialplan in res_perl, if it's been ported to 1.2x
23:24.25[TK]D-FenderManxPower : 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-FenderAnd be treated like "light"-AGI?
23:25.44ManxPower[TK]D-Fender, THIS is why: http://pastebin.ca/42838
23:26.18ManxPower[TK]D-Fender, think of res_perl as AEL, but as Perl instead of some custom bastard of a pseudo language
23:26.37ManxPoweri.e. the perl is resident and not forked each time
23:27.29*** join/#asterisk mattems (n=pronmatt@cust2229.vic01.dataco.com.au)
23:27.37mattemshey 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-FenderManxPower : But thats just 1 uber macro :)  big deal... mine was almost that big, but because of 1.2 far cleaner looking ;)
23:28.32ManxPower[TK]D-Fender, that's why I want some kind of language.
23:28.47ManxPowerThere'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)
23:29.02mattemswe are setting up an asterisk system and just having some issues with selecting a card
23:29.28ManxPowermattems, What specific issues?
23:29.31[TK]D-Fendermattems : Halmark has never done me wrong....
23:30.06ManxPower[TK]D-Fender, also I still have many 1.0.x servers
23:30.27[TK]D-FenderManxPower : I just upgraded mine... you can too!  We can make it like an intervention!
23:30.40ManxPowerI've already upgraded 4 of my servers
23:30.47ManxPowerI still have at least 4 more to do
23:30.53mattemswe want to have two lines in and 2 lines out and one of the lines will be a duet line
23:31.00mattemscan asterisk pick up the duet?
23:31.04[TK]D-Fenderlast Friday I upgraded my A104d firmware, Wanpipe, and full * from 1.0.9.
23:31.11ManxPowermattems, no
23:31.23ManxPowerbut if you define "duet" we might have suggestions for a work around.
23:31.30mattemsok
23:31.33ManxPoweralso what kind of lines?  lines to analog phones or lines to the telco?
23:32.09mattemsduet: two numbers running on the same line, if people call from one number it will ring differently
23:32.40ManxPowermattems, You might be able to do it with Asterisk.  In the USA we call it "RingMaster" or "Distinctive Ring"
23:33.04ManxPowerThere'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)
23:33.14mattemsand the lines we have are the ones from the telco
23:33.27mattems2 from the telco and 1 voip number
23:33.35mattemscan we hook em all up
23:33.38mattems??
23:33.43mattemsand with what card
23:33.44mattems?
23:33.54ManxPowerthe TDM400P card will provide up to 4 ports
23:34.06mattemsill have a look now
23:34.09ManxPowerbut you will not be happy with plugging your VoIP adapter into an Asterisk card
23:35.05mattemspretty much we just want to be able to dial out on the voip line to save costs
23:35.13mattemswe are a small org.
23:35.38DeadZenthere's this one sip component i found $299
23:35.48DeadZenbut the sites down and the emails down and the demo works great..
23:35.53mattemslol
23:35.54ManxPowerDeadZen, what are you specifically looking for?
23:35.56[TK]D-Fendermattems : You wouldn't even NEED *.  An SPA-3000 could do that for you.
23:35.58DeadZengot i hate murphy and her stupid laws
23:36.32*** join/#asterisk Soul (n=Soul@87-196-41-22.net.novis.pt)
23:36.49DeadZenManxPower: 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)
23:37.21ManxPowerWhy not just use EAGI in Asterisk?
23:37.30DeadZenim not familiar with it
23:37.42*** join/#asterisk WAudette (n=WAudette@c-67-170-156-3.hsd1.or.comcast.net)
23:37.56mattems[TK]D-Fender: i dont think the SPA-3000 will do the trick
23:37.58ManxPowerIt's like regular AGI, but better!  (better = you have access to incomning audio and can generate outgoing audio
23:37.59DeadZenso 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
23:38.29DeadZenyah this one i could get up and running right now though
23:38.33*** join/#asterisk DirtyD (n=Miranda@ool-44c24fb7.dyn.optonline.net)
23:38.55DeadZeni even hexed the demo so it uses my servers addres by default hehe
23:38.58DeadZenso i know it'll work fine
23:39.05DeadZenaudio quality is even better then sjphone
23:39.05WAudettemattems:  Have you seen the new SPA-9000?
23:39.06DirtyDHow can I connect Asterisk to a Avaya Definity extension?
23:39.16[TK]D-Fendermattems : 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-Fendermattems : I've used all kinds fo SPA models for even direct PBX bridging...
23:39.45DirtyDIs there any type of converter that I can use to convert a digital Definity jack to something Asterisk can work with?
23:40.13WAudetteDirtyD:  That's a good question.  Been wondering the same thing myself.
23:40.28[TK]D-FenderDirtyD : Intel makes a low density one, but Citel's are a great deal.
23:40.34DeadZenhttp://cti-research.com/files/siptest.rar  that's what i want
23:40.49DeadZenbut not the demo i want to buy the product but the companies gone haha
23:41.08ManxPowerDirtyD, even if there was, those sorts of things never support CPC and that's bad,
23:41.16ManxPowerimagine 24 hour long voicemails
23:41.29mattems[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
23:41.30ManxPowerall 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-Fendermattems : So you are looking at * as being your full PBX then?  Not just a front end to 'extra' features?
23:42.54mattemsyes
23:43.05mattems:)
23:43.14DeadZenare dual phones like
23:43.20DeadZenbefore their time and crappy?
23:43.44[TK]D-Fendermattems : Oh.. in that case yeah.. you want * :)
23:43.58*** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at)
23:44.07Dr-LinuxManxPower: you just said something about AGI and EAGI, can you tell me what can i use communicate with external API , like socket based
23:44.09Dr-Linux?
23:44.26ManxPowerDr-Linux, I don't understand
23:45.17DirtyDD-Femder: I only need one to handle 1 extension.
23:45.46DeadZendirtyd... he was talking to me about something else
23:46.07Dr-LinuxManxPower: i mean there is an API script at other non-asterisk server. i wanna call it from dialplan on socket based communication ...
23:46.10Qwell[]ouch...Sun is about to get a reaming from somebody at my work
23:46.13DeadZenbrb
23:46.15Qwell[]poor Sun tech
23:46.21*** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net)
23:46.25DeadZenSun sux
23:46.43DeadZenI know I used to work on their 32 processor systems
23:46.43Qwell[]a RAID array died, which holds a ton of stuff that we use to connect to outside vendors...
23:46.44DeadZenhad two of em
23:46.50DeadZenslow as crap.. everything craps out on it
23:46.54Qwell[]which means people aren't getting paid from crdeit card services...
23:47.02DeadZennice
23:47.15Qwell[]millions of transactions per day
23:47.18ManxPowerDr-Linux, AGI is like CGI for Asterisk
23:47.21Qwell[]that poor tech :(
23:47.27mattems[TK]D-Fender: so i cant plug my voip line into asterisk?
23:47.30DeadZencgi for asterisk?
23:47.37DeadZenyou can pass audio through cgi?
23:47.37ManxPowerIs RAID REALLY more reliable than non-RAID?
23:47.44[av]banihttp://www.jalopnik.com/cars/news/more-on-the-enzo-incident-on-pch-with-photos-156121.php
23:47.45Qwell[]ManxPower: not really
23:47.47ManxPowerDeadZen, EAGI supports passing audio
23:47.52DeadZenyes it is...
23:47.53Qwell[]but, a few drives died at once
23:48.03DeadZenI got two raid 5 sets
23:48.08DeadZenworks great...
23:48.11Qwell[]"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]banii use raid1...
23:48.19DeadZenraid 5 is real raid
23:48.26DeadZenthe others are cheap labor intensive options
23:48.32Dr-LinuxManxPower: 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?
23:48.34CunningPikeGreets - is there a way to read SIP responses like 'SIP/2.0 486 Busy Here' in the dialplan?
23:48.40DeadZeni bust a drive i put a new one in and the others rebuild it
23:48.46DeadZenbut it all comes down to your raid controller
23:48.48DeadZensoftware raid is a joke
23:48.59DeadZenand you need a reputable provider.. I like adaptec.. they make good shit
23:49.04CunningPikeI'm trying to differentiate between DND, phone not registered, call rejected etc for voicemail purposes
23:49.28DeadZenmy 2400a's still chug... good throughput... i feel safe putting 350 gigs of stuff i cant lose on it
23:49.38MooingLemurCunningPike: I think some of that information is in ${HANGUPCAUSE}
23:49.57CunningPikeHmm - doesn't that only cover ZAP?
23:49.57MooingLemurthat's for Zap channels I think though
23:50.21CunningPikeI've tried ${DIALSTATUS}, but it returns 'BUSY' for everything
23:50.41CunningPike${CAUSECODE} is blank :(
23:50.43Qwell[]"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).
23:51.03DeadZeni dont like sun
23:51.12DeadZeni like freebsd, gentoo and redhat fc4
23:51.15DeadZenin that order
23:51.22Qwell[]and no ACH...man...
23:51.28DeadZenouch
23:51.29DeadZenbrb
23:51.31Qwell[]this is going to seriously affect business
23:51.38CunningPikeAnd, I don't think sipGetHeader will work, because it's not a header
23:51.50DeadZenyou guys need to look into failovers
23:51.51Qwell[]So, if you guys don't get your checks direct deposited on time...now you know why :P
23:52.00Qwell[]DeadZen: it has failover
23:52.03Dr-LinuxManxPower: the question i asked is not possible in to asterisk?
23:52.23DeadZenand theres still millions of transactions lost?
23:52.25MooingLemurCunningPike: 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). "
23:52.34Qwell[]DeadZen: Not lost...just not able to be gotten right now
23:52.37websaeQwell: what do you do?
23:52.38CunningPikeYup - done that :D
23:52.41MooingLemurheh
23:52.48Qwell[]websae: programmer at a bank
23:52.48MooingLemurI'm out of ideas :)
23:52.54CunningPikeMe too :)
23:52.54DeadZenim sure you got a handle of it
23:53.01websaewhich bank eeeks
23:53.06Qwell[]websae: Can't say :)
23:53.14DeadZenI wrote an online bank before
23:53.16websaeProprietary information, understandable
23:53.17Qwell[]I'd get pretty reamed myself, right now...heh
23:53.34websaewould I know the bank here in Milwaukee, WI?
23:53.41websaeby the way anyone here from Wisconsin?
23:53.42Qwell[]websae: We're a major US bank
23:53.47Qwell[]so...I'd hope so
23:54.08websaeqwell: fantastic
23:54.29websaeqwell: good thing you mostly likely have redundancy on a "Major US Bank" network ;)
23:54.37Qwell[]oh, tons
23:54.54DeadZendoesn't sound like it helped much ;-)
23:54.56Qwell[]not sure why it isn't working...it should have moved to a different server
23:55.00Qwell[]DeadZen yeah...no clue why
23:55.11DeadZenmaybe it was an inside job
23:55.11DeadZenheh
23:55.17Qwell[]nah
23:55.17DeadZenlike from office space
23:55.26websaei have yet to meet an asterisk user from "wisconsin"
23:55.27websaehahaha
23:55.37Qwell[]websae: Kris Kielhofner
23:55.43Qwell[]of astlinux fame
23:55.52websaeohh yes
23:55.56websaeLake Geneva
23:56.00Qwell[]yeah
23:56.00MorexGot another queues question
23:56.02websaeis he ever on here?
23:56.05MorexHow would you put a caller on hold
23:56.11MorexThen get him/her back again?
23:56.13Qwell[]websae: I'm told he doesn't IRC much, if at all
23:56.25Qwell[]Morex: That's usually a function of the phone itself
23:56.28websaeahh---do you run astlinux?
23:56.37Qwell[]websae: Nope...I never have, actually
23:56.42MorexQwell: Not in this case
23:56.53*** join/#asterisk Eitch (n=hugo@unaffiliated/eitch)
23:57.49websaewww.kriscompanies.com
23:57.52websaei wish i had his job :)
23:58.44Qwell[]Morex: What type of phone?
23:59.06MorexJust a regular one
23:59.08MorexNot using SIP
23:59.14MorexOr VOIP for that matter
23:59.17Qwell[]how is it connecting to *?
23:59.17MorexLegacy PSTN stuff
23:59.23DeadZendamnit im bummed
23:59.30MorexThrough an Avaya Index and an E1 PRI
23:59.32DeadZenmy software almost had full voip
23:59.45DeadZenand i was gonna code the web app to auto configure the client
23:59.47DeadZenboo fucking hoo
23:59.49MorexSo DTMF only

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