| 00:00.09 | seanbright | l2cache: yeah... no friggin clue. can you pastebin all of the output from 'make config'? |
| 00:00.24 | seanbright | and hurry up because i am outside and it's getting cold |
| 00:00.28 | rednode | any contractors for asterisk around who are willing to do some work remotley or write up a basic step by step installation and configuration guide? |
| 00:00.41 | seanbright | rednode: voip-info.org & ~thebook |
| 00:00.45 | seanbright | save yourself some money |
| 00:00.47 | seanbright | ~thebook |
| 00:00.48 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 00:01.00 | rednode | companys money :P dont have time to read 650 pages by the weekend :S |
| 00:01.02 | seanbright | and have some pride, god damnit. |
| 00:01.04 | seanbright | heh |
| 00:01.04 | l2cache | seanbright: http://pastebin.com/d67ff7176 |
| 00:01.26 | *** join/#asterisk coppice (n=chatzill@177.162.17.210.dyn.pacific.net.hk) |
| 00:02.49 | seanbright | l2cache: ahhhh |
| 00:02.52 | seanbright | lame, but... |
| 00:03.12 | seanbright | find this line |
| 00:03.14 | seanbright | $(INSTALL) -D -m 644 zaptel.sysconfig $(DESTDIR)$(RCCONF_DIR)/zaptel |
| 00:03.18 | seanbright | and change to: |
| 00:03.25 | seanbright | install -D -m 644 zaptel.sysconfig $(DESTDIR)$(RCCONF_DIR)/zaptel |
| 00:03.29 | seanbright | that will be $125 |
| 00:03.57 | l2cache | sweet |
| 00:04.01 | l2cache | change that in what file? |
| 00:04.04 | l2cache | makefile |
| 00:04.05 | l2cache | lol |
| 00:04.06 | seanbright | yes |
| 00:04.10 | seanbright | the one you pasted |
| 00:04.12 | seanbright | i accept paypay |
| 00:04.15 | seanbright | sean.bright@gmail.com |
| 00:04.19 | seanbright | kthxbye |
| 00:04.20 | seanbright | heh |
| 00:04.28 | *** join/#asterisk sacitec (n=tobi@201.144.211.82) |
| 00:04.43 | seanbright | or you could just add: |
| 00:04.46 | seanbright | INSTALL=install |
| 00:04.52 | seanbright | somewhere at the top of the Makefile |
| 00:05.00 | seanbright | if you wanna be wacky about the whole thing |
| 00:05.51 | seanbright | rednode: if you need to install that quickly, you might try freepbx or trixbox |
| 00:06.02 | seanbright | rednode: but we don't "support" those here |
| 00:07.39 | seanbright | l2cache: make sure to thank me when that fixes your problems. it would be rude not to. |
| 00:07.49 | rednode | seanbright unfortunatley it needs to be Asterisk |
| 00:07.51 | seanbright | l2cache: and i need almost constant validation. |
| 00:08.00 | seanbright | rednode: freepbx and trixbox are asterisk |
| 00:08.06 | rednode | huh? |
| 00:08.26 | StephenF | shh thats a secret |
| 00:08.27 | seanbright | rednode: they install asterisk, some boilerplate diaplan stuff, and wrap it all with a GUI |
| 00:08.30 | [TK]D-Fender | seanHorribly inaccurate |
| 00:08.37 | seanbright | [TK]D-Fender: shut up |
| 00:08.37 | rednode | ahh ok |
| 00:08.41 | seanbright | :) |
| 00:08.44 | rednode | thanks |
| 00:08.47 | SpeedDragon | [TK]D-Fender i finaly make it work |
| 00:08.53 | SpeedDragon | now i can call outside |
| 00:08.55 | [TK]D-Fender | seanbright: I could just shut you up instead ;) |
| 00:09.07 | SpeedDragon | only thing left to do is receive calls from outside |
| 00:09.14 | seanbright | [TK]D-Fender: ok, say that freepbx and trixbox are not just asterisk installs with a GUI slapped on top |
| 00:09.28 | seanbright | [TK]D-Fender: to make my statement "horribly inaccurate" neither of them can use asterisk |
| 00:09.31 | seanbright | annnnnnnnnnnd go |
| 00:09.43 | seanbright | no? no? |
| 00:09.48 | seanbright | sweet. i love winning. |
| 00:09.57 | *** join/#asterisk envisean (n=envisean@166.129.94.21) |
| 00:10.02 | [TK]D-Fender | seanbright: FreePBX is a set of GUI scripts for configuring *. It has no distro around it ro anything at all. it is independently useless. |
| 00:10.29 | seanbright | right. still waiting for the horribly inaccurate part. |
| 00:10.40 | [TK]D-Fender | seanbright: trixbox is a complete distro that bundles a pile of stuff including * & FreePBX and gives GUI interfaces to configure everything based on its toaster design |
| 00:10.53 | drmessano | Inaccurate |
| 00:11.00 | drmessano | Trixbox does NOT include FreePBX |
| 00:11.10 | *** join/#asterisk infinity1 (i=brendon@saleen.netcal.com) |
| 00:11.21 | [TK]D-Fender | drmessano: I suppose. |
| 00:11.22 | sacitec | mmm, i think it does |
| 00:11.36 | drmessano | It does NOT |
| 00:11.40 | [TK]D-Fender | drmessano: On the premise they have now FORKED it I take you as implying? |
| 00:12.00 | seanbright | regardless. someone asking (all due respect to rednode) *basic* questions about asterisk doesn't need to understand the difference. |
| 00:12.03 | drmessano | What is included in trixbox is no longer FreePBX.. it is already forked and modified |
| 00:12.04 | seanbright | you have to know your audience. |
| 00:12.20 | [TK]D-Fender | seanbright: Misguiding someone who understands little screws them up MUCH worse. |
| 00:12.24 | sacitec | well, based on freepbx |
| 00:12.28 | seanbright | [TK]D-Fender: i disagree. |
| 00:12.41 | rednode | :S i think ill stick to Asterisk :P |
| 00:12.42 | drmessano | Considering the direction both projects have gone, it is a far cry from FreePBX now |
| 00:12.54 | [TK]D-Fender | seanbright: So you believe him completely lost so whats one more misconception? |
| 00:12.57 | drmessano | Duh, its all fucking asterisk |
| 00:13.07 | [TK]D-Fender | drmessano: Without the lube! |
| 00:13.13 | seanbright | [TK]D-Fender: no, i think he wants to get an asterisk install up and running quickly |
| 00:13.14 | [TK]D-Fender | uNF! |
| 00:13.22 | drmessano | Thats my #1 peeve.. Bitch about dialplans created by apps if you want, but in the end, THEY MAKE DIALPLANS that ASTERISK RUNS |
| 00:13.29 | [TK]D-Fender | seanbright: As he said, he's getting a contractor. |
| 00:13.30 | drmessano | Asterisk IS the running APP |
| 00:13.49 | seanbright | [TK]D-Fender: while he may be getting a contractor, he never explicitly said he was getting on. |
| 00:13.56 | seanbright | s/on./one./ |
| 00:14.05 | drmessano | That's like saying "I don't use Notepad, I use text files" |
| 00:14.09 | drmessano | Duh |
| 00:14.40 | seanbright | he said "most likeley [sic]" |
| 00:15.03 | [TK]D-Fender | seanbright: rednode>im not qualified at all for this project but unfortunatley im the most qualified person in the company, so most likeley ill hire a contractor |
| 00:15.13 | seanbright | he said "most likeley [sic]" |
| 00:15.23 | seanbright | i rest my case |
| 00:15.26 | seanbright | beers on [TK]D-Fender! |
| 00:15.57 | [TK]D-Fender | seanbright: thanks for letting me finish :) |
| 00:16.14 | StephenF | rednode: dude... look what you did |
| 00:16.16 | seanbright | well we were both scrolling up |
| 00:16.17 | drmessano | Yay for "I don't use a GUI, I use Asterisk" for the gold medal at the special olympics |
| 00:16.19 | seanbright | i just am faster |
| 00:16.20 | seanbright | heh |
| 00:16.28 | [TK]D-Fender | [sic] <- ultimate BS editing term of our lifetimes |
| 00:16.43 | l2cache | seanbright: When I do a 'service zaptel start' no stdout |
| 00:16.50 | [TK]D-Fender | seanbright: Yes, a headless chicken you are. Run along now! |
| 00:17.12 | seanbright | [TK]D-Fender: you first! |
| 00:17.23 | seanbright | l2cache: no stdout? what do you mean? |
| 00:17.30 | seanbright | (and yes, i know what stdout is) |
| 00:17.36 | seanbright | (just not how it applies) |
| 00:17.49 | l2cache | no output |
| 00:18.01 | seanbright | ok |
| 00:18.07 | seanbright | sorry to hear that |
| 00:19.21 | seanbright | [TK]D-Fender: you in AZ? |
| 00:19.47 | rednode | lol |
| 00:20.11 | seanbright | ... @ astricon? |
| 00:20.32 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 00:21.29 | seanbright | l2cache: /etc/rc.d/init.d/zaptel start |
| 00:21.35 | [TK]D-Fender | seanbright: No. |
| 00:23.15 | l2cache | still no output from that command |
| 00:23.34 | seanbright | l2cache: interesting. |
| 00:23.58 | seanbright | l2cache: less /etc/rc.d/init.d/zaptel |
| 00:24.25 | seanbright | [TK]D-Fender: not interested or not able? |
| 00:25.25 | [TK]D-Fender | seanbright: Not spending my money on the trip |
| 00:25.39 | seanbright | ohhh |
| 00:25.45 | seanbright | thinly veiled... but i getcha |
| 00:25.48 | seanbright | :) |
| 00:26.11 | [TK]D-Fender | seanbright: Wafer-thin indeed |
| 00:26.13 | seanbright | my sister had to go and get married this weekend |
| 00:26.24 | seanbright | so i can't go to astridevcon |
| 00:26.29 | seanbright | which upsets me so |
| 00:27.06 | Qwell | I still say you should have her reschedule it next time :p |
| 00:27.34 | seanbright | this better be the only time she gets married |
| 00:27.46 | l2cache | seanbright: you want a full output of zaptel? |
| 00:27.54 | seanbright | l2cache: no, but there is something in it? |
| 00:28.03 | l2cache | yes |
| 00:28.13 | seanbright | l2cache: just run 'ztcfg -vvvvvvv' |
| 00:28.35 | l2cache | no such command |
| 00:28.39 | seanbright | yikes |
| 00:28.45 | seanbright | um |
| 00:28.54 | seanbright | your install is jacked up |
| 00:29.01 | l2cache | yep |
| 00:29.14 | l2cache | new ver of zaptel maybe? |
| 00:29.24 | seanbright | l2cache: where did you get zaptel again? |
| 00:29.29 | seanbright | l2cache: gnudialer? |
| 00:29.29 | l2cache | gnudialer.org |
| 00:29.32 | seanbright | yeah... |
| 00:29.38 | seanbright | is there anyone in #gnudialer/ |
| 00:29.39 | seanbright | ? |
| 00:29.46 | l2cache | everyone is afk |
| 00:29.49 | l2cache | forever |
| 00:29.58 | seanbright | ah |
| 00:30.03 | seanbright | well i'm out of ideas |
| 00:30.13 | seanbright | and i'm freezing my ass off |
| 00:30.16 | seanbright | must go home |
| 00:30.18 | seanbright | back later. |
| 00:40.41 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-2ec2ded7a726527b) |
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| 01:20.57 | infinity1 | anyone have a link for the latest polycom firmware? |
| 01:21.32 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.67) |
| 01:23.04 | adr3nalin3 | anyone have a problem where asterisk just won't start? |
| 01:23.13 | infinity1 | adr3nalin3: yea. check the logs |
| 01:24.00 | adr3nalin3 | infinity1: thanks zaptel problem |
| 01:24.31 | StephenF | whats an easy way to check if a channel variable is blank, or undefined? |
| 01:24.45 | StephenF | I want to use a gotoif a certain variable is blank |
| 01:24.55 | [TK]D-Fender | StephenF: Thats exactly where |
| 01:25.07 | StephenF | ok, does this look right then: GotoIf($[${OUTGOING_CIDNUM}xxx = xxx]?5:2) |
| 01:25.32 | [TK]D-Fender | StephenF: supposed that'd do |
| 01:25.37 | StephenF | I saw someone do something like that, but i dont really understand what is happening |
| 01:25.40 | StephenF | lol, ok |
| 01:25.54 | StephenF | I want it to check if OUTGOING_CIDNUM is blank or not |
| 01:26.05 | StephenF | is there a better way? |
| 01:26.07 | _ShrikE | StephenF: look at the isnull function |
| 01:26.12 | StephenF | ahh ok |
| 01:26.29 | [TK]D-Fender | no need |
| 01:26.42 | [TK]D-Fender | GotoIf($["${OUTGOING_CIDNUM}"=xxx = xxx]?5:2) |
| 01:26.50 | [TK]D-Fender | GotoIf($["${OUTGOING_CIDNUM}"=""]?5:2) |
| 01:27.20 | StephenF | oh ok, so if it matches "blank" |
| 01:28.03 | StephenF | i will try that then, thanks |
| 01:35.32 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 01:37.36 | adr3nalin3 | Hey guys I am having trouble with a TE122P, [TK]D-Fender you helped me with a similar problem earlier on a TDM400P analog. I am getting the same kind of issue where the telco doesn't seem to recognize the numbers dialed. Any ideas? The old phone system says that the switch type was a AT&T 5ESS. I have tried appending a pause and I do get different results but the call does not go through. |
| 01:38.41 | [TK]D-Fender | adr3nalin3: No such thing as a "pause" with PRI |
| 01:38.54 | [TK]D-Fender | adr3nalin3: Look at real debug. |
| 01:39.22 | adr3nalin3 | adr3nalin3: wasn't sure if there was. Will do. |
| 01:41.05 | seanbright | adr3nalin3: do you have nsf defined in your zapata.conf for your at&t span? |
| 01:41.52 | seanbright | we have to use 'nsf = sdn' for our at&t PRI |
| 01:45.31 | *** join/#asterisk vipcarrier (n=vipcarri@ool-44c65232.dyn.optonline.net) |
| 01:46.32 | vipcarrier | hello |
| 01:46.47 | vipcarrier | I'm trying to configure Asterisk with remote mysql server and got an error |
| 01:46.48 | vipcarrier | WARNING[28318]: config.c:1331 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available |
| 01:46.57 | adr3nalin3 | anybody know what this is about: Loading zaptel framework: WARNING: /etc/modprobe.conf line 1: ignoring bad line starting with 'options' |
| 01:48.00 | adr3nalin3 | heh, centos does automatically, just commented out |
| 01:51.47 | vipcarrier | any one can give me a tip where to dig a problem? |
| 01:53.44 | adr3nalin3 | [TK]D-Fender: I am getting --> app_dial.c:1183 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) ........from debug. |
| 01:54.34 | vipcarrier | any one can help me with asterisk real time? |
| 01:57.42 | [TK]D-Fender | adr3nalin3: that can be the remote side saying that the # is busy, or that there is no channel to make an attempt on. |
| 01:58.05 | [TK]D-Fender | sdrAnd you have clearly not learned your lesson which is to PASTEBIN EVERYTHING. |
| 01:58.35 | StephenF | what is the default VM menu greeting saying? Something like "comedian mail"? |
| 01:58.55 | Qwell | StephenF: yes |
| 01:59.03 | [TK]D-Fender | StephenF: Voicemailmain, yes. |
| 01:59.09 | [TK]D-Fender | StephenF: voicemail no. |
| 01:59.09 | StephenF | umm why does it say comedian mail? |
| 01:59.14 | adr3nalin3 | [TK]D-Fender: pastebin: http://pastebin.com/m1dea094d |
| 01:59.15 | Qwell | why not? |
| 01:59.18 | StephenF | yeah voicemailmain |
| 01:59.19 | StephenF | lol, ok |
| 01:59.28 | [TK]D-Fender | StephenF: Its a joke take on the old "Meridian Mail" system |
| 01:59.43 | StephenF | ohhk, thats what I thought it was saying at first. |
| 01:59.53 | StephenF | Im thinking why is it saying Meridian Mail... |
| 01:59.57 | adr3nalin3 | me too^^ |
| 02:00.12 | StephenF | those crazy asterisk guys |
| 02:00.24 | [TK]D-Fender | adr3nalin3: I might wonder if 7 digit numbers are legal where you are calling out... |
| 02:00.49 | [TK]D-Fender | adr3nalin3: Next suspect is your pridialplan & prilocaldialplan (both are often advised to be set to "no" in zapata.conf |
| 02:01.10 | [TK]D-Fender | "unknown" rather..... |
| 02:02.35 | adr3nalin3 | good points especially the first one. I shall check these. |
| 02:10.13 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-209-36.phlapa.east.verizon.net) |
| 02:10.59 | adr3nalin3 | [TK]D-Fender: as always thanks for your help. I haven't got it yet but I need to start re-connecting the 3COM system. /puke |
| 02:14.49 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 02:14.49 | *** mode/#asterisk [+o denon] by ChanServ |
| 02:20.35 | jeev | is it possible to show timestamp on console screen ? |
| 02:27.00 | *** join/#asterisk newmember (n=chatzill@S010600036d1139bb.cg.shawcable.net) |
| 02:28.27 | *** join/#asterisk moy (n=moy@63-255-103-7.ip.mcleodusa.net) |
| 02:31.11 | StephenF | can I connect a remote phone to my * box over SIP without a VPN? |
| 02:31.21 | StephenF | yes right? |
| 02:31.29 | StephenF | And is that secure? |
| 02:31.52 | vipcarrier | <PROTECTED> |
| 02:32.00 | vipcarrier | any one can help with this? |
| 02:32.08 | vipcarrier | I'm trying to run mysql on remote host |
| 02:33.42 | *** join/#asterisk logicwrath (n=no@c-68-42-253-39.hsd1.mi.comcast.net) |
| 02:34.59 | [TK]D-Fender | vipcarrier: And the reason you aren't showing us the complete CLI output of the error, your configs, verification that th remote MySQL instance is up and contactable is...? |
| 02:35.22 | [TK]D-Fender | StephenF: Yes, and "not so much" respectively. |
| 02:35.37 | StephenF | so is it common practice to use a VPN? |
| 02:35.49 | [TK]D-Fender | StephenF: No, most simply don't care. |
| 02:36.10 | [TK]D-Fender | StephenF: Between branch offices sure, but individual remote phone, no |
| 02:36.32 | StephenF | oh ok, so basically without VPN my Voice traffic could be sniffed. And if i dont care that my calls could be listened then that doesnt matter much |
| 02:37.15 | StephenF | is the registration information sent over clear text? So if sniffed a malicious user could spoof the remote phone and make calls out through the PBX? |
| 02:37.58 | [TK]D-Fender | StephenF: Yes, that is possible. Easier to just listen in |
| 02:38.09 | StephenF | ok |
| 02:44.04 | *** join/#asterisk Sinist3r (n=IamLegio@209.160.40.98) |
| 02:44.44 | Sinist3r | Is there a step by step procedure to configuring and testing asterisk? |
| 02:44.54 | Sinist3r | I've already compiled and installed it. |
| 02:45.01 | Sinist3r | Just need help with configuring |
| 02:45.28 | jaytee | ~book |
| 02:45.29 | jbot | book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 02:46.31 | Sinist3r | thanks |
| 02:48.39 | *** join/#asterisk rcy (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 02:52.23 | *** join/#asterisk jameswf-home (n=james@ip68-2-99-240.ph.ph.cox.net) |
| 02:55.07 | *** join/#asterisk growltiger (n=growltig@ip70-179-54-235.sd.sd.cox.net) |
| 02:57.12 | [TK]D-Fender | Sinist3r: And no there is no real way to test * without simply using it. |
| 02:58.29 | jameswf-home | Your license has been formally accepted by our legal department << sounds so peppy like i won cash |
| 03:00.26 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 03:18.20 | logicwrath | Are these ring group contexts wrong? http://pastebin.com/d60d1f36c when it bridges the call to the outside line i get no audio. Is that an RTP problem? |
| 03:19.26 | jameswf-home | look okay to me |
| 03:19.49 | jameswf-home | how do you call em |
| 03:20.16 | logicwrath | one cell going into * and then going out to diff cell using those contexts |
| 03:20.31 | jeev | hi jameswf-home |
| 03:21.58 | logicwrath | it seems like a bridging problem, as soon as the call gets transferred to the outside line i lose the ringing on the first cell |
| 03:22.48 | logicwrath | its quite possible my firewall is not routing the rtp ports properly as im not a cisco expert using range access-lists |
| 03:23.13 | logicwrath | could this be cause by rtp ports? |
| 03:29.24 | [TK]D-Fender | ~sipnat |
| 03:29.25 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 03:29.26 | [TK]D-Fender | ^^^^^^^^^ |
| 03:29.49 | logicwrath | k, i will review thanks |
| 03:34.30 | logicwrath | is NAT a concern when a call initiates from outside cell and ends/fails to outside cell? I would think the NAT issues would arise from user extentions |
| 03:35.07 | logicwrath | i am still adjusting some things per those docs, i am just curious |
| 03:36.32 | logicwrath | can i specify multiple localnet= lines if I have multiple internal subnets with VPNs? |
| 03:37.16 | [TK]D-Fender | logicwrath: Any NAT involvement is a concern |
| 03:37.34 | [TK]D-Fender | logicwrath: and yes you can specify multiple local subnets 1 per line |
| 03:37.45 | logicwrath | ty |
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| 04:08.12 | *** join/#asterisk Juxt (n=denis@c-76-110-186-156.hsd1.fl.comcast.net) |
| 04:08.36 | Juxt | i'm trying to compile asterisk-addons and getting a ton of errors on chan_ooh323 |
| 04:08.42 | Juxt | i dont even want ooh_323 |
| 04:08.48 | Juxt | clear |
| 04:08.51 | Juxt | ls |
| 04:10.02 | [TK]D-Fender | Juxt: feel free to NOT choose them in menuselect |
| 04:10.28 | Juxt | ha! didnt know there was menuselect. thanks! |
| 04:13.42 | Juxt | should asterisk-addons be placed inside of asterisk directory or something? make cant seem to find asterisk.h, etc. |
| 04:18.02 | tzafrir_laptop | Juxt, what errors? |
| 04:18.13 | tzafrir_laptop | this is a bug and shouldn't happen |
| 04:18.45 | *** join/#asterisk jplank (n=GBove@96.sub-75-208-192.myvzw.com) |
| 04:18.47 | Juxt | app_addon_sql_mysql.c:19:22: error: asterisk.h: No such file or directory |
| 04:19.00 | Juxt | i feel like something isnt in the path, etc. |
| 04:19.32 | Juxt | my asterisk source is in /usr/src/asterisk-1.4.21.2 and asterisk-addons are in /usr/src/asterisk-addons-1.4.7 |
| 04:21.39 | tzafrir_laptop | Juxt, you need asterisk installed |
| 04:21.46 | tzafrir_laptop | or e.g. asterisk-dev installed |
| 04:21.47 | Juxt | i have it installed |
| 04:21.55 | Juxt | just in a custom path /opt/asterisk |
| 04:22.17 | tzafrir_laptop | ./configure --with-asterisk=/opt-asterisk |
| 04:22.30 | tzafrir_laptop | hmm.... bug of the configure script |
| 04:23.44 | Juxt | that worked |
| 04:28.46 | boolean12 | Has anyone gotten festival to work in 1.6? |
| 04:37.00 | logicwrath | still unable to bridge incoming cell phone call to other outbound cell phone: http://pastebin.com/d60d1f36c when the call bridges, i get no audio on either end however, the line is active on both ends. still need some advice troubleshooting |
| 04:40.16 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 04:41.13 | [TK]D-Fender | logicwrath: And you are failing to show the call with SIP debug. All the dialplan in the world won't save you from a broken netwoking situation. |
| 04:42.05 | logicwrath | why does it have anything to do with networking when the call originates and is destined from cell phones |
| 04:42.16 | jeev | http://www.the-asterisk-book.com/unstable/applikationen-setcallerpres.html what exactly is 'screen' ? |
| 04:43.46 | [TK]D-Fender | logicwrath: Because you've already mentions NAT involvement, and you are using SIP prooviders |
| 04:44.13 | [TK]D-Fender | logicwrath: And "cell phone" says absolutely nothing. its how you get to the PSTN and your other endpoints that counts |
| 04:47.05 | logicwrath | is this enough debug, i may have to adjust the putty console logging: http://pastebin.com/d2cbedaf3 |
| 04:47.33 | *** join/#asterisk SanityIO (n=SanityIO@77.242.106.48) |
| 04:49.31 | [TK]D-Fender | logicwrath: pastebin your sip.conf masking only passwords |
| 04:51.33 | logicwrath | http://pastebin.com/d6d45ef35 |
| 04:51.54 | *** join/#asterisk SanityIO_ (n=SanityIO@77.242.106.48) |
| 04:53.20 | *** join/#asterisk moy (n=moy@63-255-103-7.ip.mcleodusa.net) |
| 04:53.53 | [TK]D-Fender | logicwrath: describe your server's path to the internet |
| 04:54.02 | logicwrath | it is behind a cisco pix 501 |
| 04:54.05 | *** join/#asterisk terracon (n=greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 04:54.15 | logicwrath | i confirmed it was not rtp by setting up rtp.conf for 10 ports |
| 04:54.20 | logicwrath | and statically mapping them |
| 04:54.24 | logicwrath | direct to * |
| 04:54.45 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 04:54.45 | logicwrath | 5060 is also statically mapped in PIX |
| 04:55.13 | logicwrath | i have tried with and without fixup protocol sip and fixup protocol sip udp |
| 04:55.43 | [TK]D-Fender | EW... PIX is NASTY |
| 04:55.59 | [TK]D-Fender | this series is jsut about the WORST thing you could be behind |
| 04:56.38 | logicwrath | i get that impression from some of the posts ive read, ive always liked PIX's otherwise |
| 04:56.53 | logicwrath | i own about 6 of them |
| 04:56.53 | [TK]D-Fender | logicwrath: Setup more ports, put them in the 10000+rang starting from 10000. All ports should be UDP. Also you should have "nat=yes" under [general], and "nat=no" for your itsp peers |
| 04:57.13 | [TK]D-Fender | logicwrath: Yes, the "otherwise my work fine, but its hell for * |
| 04:57.22 | [TK]D-Fender | may* |
| 04:57.44 | logicwrath | ive done 10000-10010 statically mapped already |
| 04:57.51 | logicwrath | i will try again if you want to see debug |
| 04:58.07 | [TK]D-Fender | logicwrath: Make sure the PIX is doing NO SIP transform |
| 04:58.23 | logicwrath | ill disable fixup again as well |
| 04:59.49 | jameswf-home | [TK]D-Fender: be a pal ship one from canada |
| 05:00.42 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 05:02.48 | jameswf-home | umm hi |
| 05:04.07 | [TK]D-Fender | jameswf-home: C'mon you know John McCain invented them ;) |
| 05:04.36 | jameswf-home | [TK]D-Fender: but he invented em in canada to avoid taxes |
| 05:05.07 | jameswf-home | I am just happy al gore kept the internet in the US |
| 05:05.48 | logicwrath | ok, i specified rtp.conf for 10000 - 10010, reloaded asterisk, and hard mapped in pix, disabled fixup on sip and sip udp and saved the sip debug here http://pastebin.com/m66824373 |
| 05:09.30 | jplank | john mc cain didn't invent the blackberry? |
| 05:10.35 | jplank | did anyone catch the AGI porn conference today? or the ask the guru's? |
| 05:10.41 | logicwrath | i also added nat=no to the trunks |
| 05:12.36 | jameswf-home | AGI porn?? like a videophone? |
| 05:12.45 | jplank | no |
| 05:13.06 | jameswf-home | dial a fax porn |
| 05:13.16 | jplank | I found out what one dev did for fun |
| 05:13.23 | jplank | pbx_lua |
| 05:15.24 | jeev | heh |
| 05:15.36 | jplank | I was sooo hyped up for pbx_lua |
| 05:15.41 | jplank | I don't know what I was thinking |
| 05:16.04 | jplank | I was thinking all these cool things |
| 05:16.09 | jplank | kind of like AGI |
| 05:16.31 | jplank | turned out to be a replacement for extensions.conf and the like |
| 05:17.14 | *** join/#asterisk erogevets (n=chatzill@noc-gw.maxnet.net.nz) |
| 05:20.36 | jplank | I was really hoping to catch the AGI porn thing |
| 05:20.42 | jplank | I wanted to see some cool AGI scripts |
| 05:21.03 | jameswf-home | jplank: you can always write your own |
| 05:21.31 | jplank | yea, I'm not that creative to thing of cool things to do with it |
| 05:21.50 | jplank | I just use it to complete obstacles |
| 05:21.59 | jplank | think of cool things* |
| 05:22.22 | jameswf-home | I built a "Dial-A-Distro" box once... |
| 05:22.44 | jameswf-home | put in a cd and tyoe a code for the distro you want |
| 05:22.52 | jameswf-home | *type |
| 05:23.21 | jplank | hmmm |
| 05:23.28 | jplank | most be a big install CD |
| 05:23.38 | jeev | lol |
| 05:23.50 | jplank | I heard the funniest thing yesterday at astricon durning asterisk 123 |
| 05:24.01 | jplank | guy was talking about ubuntu |
| 05:24.20 | jplank | and he said its an african word that means "isn't able to compile debian" |
| 05:25.06 | [TK]D-Fender | ok, checkout time. Later all |
| 05:25.59 | jameswf-home | I have found pure debian has a tendency to leave one in dependency hell |
| 05:26.16 | jplank | I think that was his point |
| 05:26.51 | jameswf-home | people slam ubuntu but IMHO it just works so... |
| 05:27.18 | jplank | I use it at work all the time |
| 05:27.31 | jplank | makes a quick server |
| 05:28.16 | jplank | esp with the LAMP install option (LAMP, XAMPP?) |
| 05:29.09 | *** join/#asterisk DaveCanoe (n=Dave@strike.dclg.ca) |
| 05:30.38 | jblack | Yeah. I don't see much of the dependancy hell that I remember from debian. |
| 05:30.52 | jplank | for a novice it is |
| 05:31.15 | jameswf-home | I could use gentoo but why |
| 05:31.21 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 05:31.33 | jblack | Dependancy hell is a punishment, even for 'vets'. |
| 05:31.41 | jameswf-home | the answer to everything is NOT rebuild the kernel |
| 05:31.55 | jeev | sup jack black |
| 05:32.02 | fiddur | Debian dependencies is not a hell, it's just a nice country road with lot's of turns :) |
| 05:32.18 | jblack | jeev: Your /whois kung-fu is weak. |
| 05:32.44 | jplank | fiddur wins |
| 05:37.05 | jeev | ;_ |
| 05:39.34 | drmessano | ..for everything else, there's Mastercard Black Platinum Silver Gold Uranium |
| 05:44.20 | jplank | I dont get people who hang themselves from fish hooks, I get some people are into weird things, but seriously? |
| 05:44.47 | jeev | lol |
| 05:55.41 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-207-199.phlapa.east.verizon.net) |
| 05:58.17 | jblack | Perversion-ess probably follows f(y)=1/x formula. |
| 05:59.40 | jblack | Somewhere out there, someone exists that would just _love_ the idea of being dropped into a vat of dog poo. Hopefully, just one. |
| 06:00.46 | *** join/#asterisk _gm (n=gmustafa@202.133.78.60) |
| 06:02.32 | jameswf-home | obama smot poker http://cdn.liveleak.com/17/media17/2008/Sep/10/LiveLeak-dot-com-224531-obama2.jpg |
| 06:02.58 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:06.04 | C4away | photoshop |
| 06:06.10 | C4away | the reflections are all wrong |
| 06:07.33 | jplank | look at his knee |
| 06:07.52 | jplank | forget photoshop, that was done in mspaint |
| 06:08.22 | C4away | photoshop the noun form of the verb photoshopped, not the application by Adobe |
| 06:09.24 | jplank | sorry, forgot the sarcasm tags |
| 06:09.51 | C4away | anyway, on a serious asterisk-related note ... |
| 06:10.16 | C4away | is there any way to add the ability to escape from voicemail without pressing * ? |
| 06:10.18 | C4away | or 0 |
| 06:10.34 | C4away | for example "to reach me on my cell phone press 1 now, or leave a message after the tone" |
| 06:11.15 | C4away | I could put the user in their own context and set operator=no for their options on their voicemail user |
| 06:11.22 | C4away | then create an 'a' extension in their context |
| 06:11.35 | jplank | make a IVR that times out to a application that records a message? |
| 06:11.53 | C4away | well |
| 06:12.12 | C4away | how I have it now is I have an application that plays a message and then sends them to the users voicemail with just the 's' option |
| 06:12.19 | C4away | so it skips all messages and just beeps |
| 06:12.38 | C4away | but I have to go through and customize all of the options that end up at that user's voice mail |
| 06:12.47 | C4away | anyway, just wondering if there was an easy way to do that |
| 06:12.58 | C4away | or if I could pursue my IVR hack |
| 06:13.32 | jplank | I think the IVR way would really be the only way |
| 06:13.39 | jplank | I could be wrong though |
| 06:14.02 | C4away | that's what my research lead be to beleive as well |
| 06:14.05 | jameswf-home | found palin mccain porn too but well yeah |
| 06:14.12 | C4away | hmm |
| 06:14.17 | jplank | link? |
| 06:14.20 | C4away | not sure I want to see mccain naked |
| 06:14.21 | jplank | ....i think... |
| 06:14.59 | jplank | I met allison today at astricon....it was very weird |
| 06:15.04 | C4away | heh |
| 06:15.07 | C4away | weird how? |
| 06:15.18 | C4away | personality? or the fact that every time she talked you wanted to press the # key? |
| 06:15.20 | jplank | some guy I was talking to introduced me to her |
| 06:15.43 | jplank | she turned to me and was like "congratulations......" |
| 06:15.48 | C4away | what? |
| 06:16.00 | jplank | you know the recording when you first install * |
| 06:16.04 | C4away | yea |
| 06:16.10 | jplank | freaked me out |
| 06:16.13 | C4away | haha |
| 06:16.16 | jameswf-home | jplank: http://brentroad.com/message_topic.aspx?topic=539420 |
| 06:16.19 | vipcarrier | hello |
| 06:16.20 | *** join/#asterisk mchou (n=mchou@c-76-103-44-118.hsd1.ca.comcast.net) |
| 06:16.28 | vipcarrier | I'm trying to run a remote mysql server for asterisk |
| 06:16.33 | jplank | like, her normal voice is the voice she uses for asterisk |
| 06:16.38 | vipcarrier | and I'm getting the following error [Sep 25 02:17:53] WARNING[29862]: config.c:1331 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available |
| 06:16.54 | jblack | sounds like you didn't enable the mysql module |
| 06:17.02 | vipcarrier | yes I did |
| 06:17.04 | *** join/#asterisk [netman] (n=netman@108.Red-83-32-25.dynamicIP.rima-tde.net) |
| 06:17.24 | jameswf-home | yes hun nuh uh |
| 06:17.39 | jplank | omg james, thats awesome |
| 06:17.39 | vipcarrier | sipusers => mysql,asterisk,sipusers |
| 06:17.48 | jplank | the one with mccain giving the thumbs up |
| 06:17.59 | vipcarrier | sippeers => mysql,asterisk,sipusers |
| 06:18.07 | vipcarrier | voicemail => mysql,asterisk,vmusers |
| 06:18.35 | vipcarrier | and I did it in res_mysql.conf |
| 06:18.54 | vipcarrier | [general] |
| 06:18.54 | vipcarrier | dbhost = 192.168.40.2 |
| 06:19.01 | vipcarrier | dbname = asterisk |
| 06:19.01 | vipcarrier | dbuser = astrealtime |
| 06:19.10 | vipcarrier | dbpass = XXXXX |
| 06:19.16 | *** join/#asterisk Chris-NB (n=chris@nfw.ecos.at) |
| 06:19.16 | vipcarrier | dbport = 3306 |
| 06:19.22 | vipcarrier | what did I do wrong? |
| 06:19.46 | C4away | first five X's is not a very strong password |
| 06:20.00 | C4away | I use five *'s instead, much more secure |
| 06:20.01 | vipcarrier | don't worry about my password ;-) |
| 06:20.20 | jplank | what sad is I was thinking the same thing as C4away |
| 06:20.31 | jplank | I held back on saying it though |
| 06:20.36 | vipcarrier | but still why I'm getting that message |
| 06:20.41 | C4away | I never hold back on saying stupid things |
| 06:20.44 | C4away | it's part of my charm |
| 06:20.57 | jplank | vipcarrier: what do you think this is a asterisk help channel or something? |
| 06:20.59 | vipcarrier | on my mysql server's i have enabled user astrealtime@192.168.% |
| 06:21.19 | vipcarrier | jplank I think some one can give me a tip |
| 06:21.21 | C4away | I've never got realtime to work |
| 06:21.24 | C4away | mostly for lack of trying |
| 06:21.39 | jameswf-home | my password is youllneverbielievemypasswordissimplypassword |
| 06:21.45 | jplank | lol |
| 06:22.02 | jplank | vipcarrier: I've given up trying to get help in this channel a long time ago |
| 06:22.02 | C4away | actually it is amazing how secure a blank password is on windows |
| 06:22.09 | jplank | someone usually yells at me |
| 06:22.14 | vipcarrier | okey so how to do u connect few asterisk and few opensers's with few mysql's ??? |
| 06:22.24 | jplank | usually either fender or drmessano |
| 06:22.30 | C4away | I have beat on a customer's computer for over an hour trying to guess their password only to call them the next day and ask, they say "oh it's just blank" |
| 06:22.35 | jameswf-home | a few days if practice |
| 06:22.38 | C4away | learned that one the hard way |
| 06:23.24 | jplank | password usually works for 80% of the users at my office |
| 06:23.44 | *** join/#asterisk sircco (n=sircco@dh207-69-105.xnet.hr) |
| 06:23.51 | jplank | sad thing is our IT admins password is just as easy to guess |
| 06:24.00 | C4away | I prefer using a very strong password policy with a 7 day expiration and no reuse of passwords |
| 06:24.03 | sircco | what is best way to get query into variable in asterisk dialplan |
| 06:24.13 | C4away | it garuntees that they will write it within easy reach of the keyboard |
| 06:24.19 | jplank | lol |
| 06:24.32 | jplank | or calling you once a week for a password reset |
| 06:25.15 | jplank | we used to do 40 day expirations |
| 06:25.20 | jameswf-home | sircco: please be less vague |
| 06:25.29 | jplank | we found users were just doing password, password1, password2 ect |
| 06:25.34 | jplank | kind of defeated the purpose |
| 06:25.59 | jplank | and they still kept forgetting the password |
| 06:26.52 | jplank | then again, after 100 years, they'd have a pretty strong password |
| 06:27.02 | sircco | jameswf-home: ok i have extension that calls channel, i want to get some other data from mysql into extensions.conf. I saw i can do that with app_dbquery. Maybe there is some other way to do this? |
| 06:27.03 | jameswf-home | I worked for AT&T rhey had rules 8+ charicters letters numbers symbols must differ atleast 60% from previous 10 |
| 06:27.49 | jameswf-home | sircco: AGI |
| 06:28.12 | sircco | jameswf-home: thanks! |
| 06:33.41 | *** join/#asterisk h-idrisi (n=h-idrisi@212.103.170.132) |
| 06:34.32 | jplank | night all, hoping to catch the 9am keynote tomorrow |
| 06:34.53 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 06:45.45 | *** join/#asterisk rvhi (n=chatzill@udp255518uds.hawaiiantel.net) |
| 06:46.39 | rvhi | how do i find out what's the local calling area for a npa nxx? |
| 06:46.46 | rvhi | e.g. 209-825 |
| 06:47.03 | rvhi | what are other 209-xxx are local calling area? |
| 06:51.14 | jameswf-home | ping hi365 /msg |
| 06:51.23 | fiddur | Hmm, I just tried conf2ael in 1.6.0-rc6... It converted all ',' to '|'... I thought you weren't supposed to use '|' in 1.6.0, or is it the opposite in ael-files? |
| 06:53.50 | *** join/#asterisk matsk (n=Mats@90.235.26.58) |
| 07:02.31 | tzafrir_laptop | fiddur, a leftover from 1.4? |
| 07:03.43 | fiddur | tzafrir_laptop: seems likely... just a bit confusing |
| 07:03.54 | *** join/#asterisk barakuda (n=chatzill@aliens4.betex.ru) |
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| 07:20.48 | *** part/#asterisk zydoon (n=zydoon@213.150.170.26) |
| 07:22.53 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 07:22.53 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 07:44.39 | *** join/#asterisk whymarkwh (n=dsfsdfsd@196.211.34.2) |
| 07:44.49 | whymarkwh | hi anyone active? |
| 07:46.32 | kaldemar | just ask. |
| 07:49.49 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 07:50.44 | whymarkwh | linking asterisk system to panasonic i answer the call then noop the extension to see what digits are goming down(getting dtmf digits from the panasonic) the problem i haveis, i do _X. but the panasonic sends a # first then my it says it an invalid extension '#' in context incoming how do i overcome this? |
| 07:51.59 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-33-82-252-12-6.adsl.proxad.net) |
| 07:52.40 | gr0mit | how are you linking the two, whymarkwh ? |
| 07:53.19 | whymarkwh | divert from panasonic to fxo on asterisk system |
| 07:53.36 | gr0mit | so where is the # coming from? |
| 07:54.22 | gr0mit | what type of panasonic? |
| 07:54.47 | kaldemar | whymarkwh: _#X., X only matches to 0-9. then remove the # in your dialplan. or just remove the # in the panasonic end. |
| 07:55.11 | whymarkwh | thx let me try that |
| 08:01.47 | whymarkwh | it answers the call but can noop it |
| 08:03.11 | whymarkwh | the # is coming from the panasonic |
| 08:03.19 | kaldemar | "but can noop it"? |
| 08:03.42 | gr0mit | whymarkwh, what model of Panasonic? |
| 08:05.20 | *** join/#asterisk easycrypt (n=savek@ip-186.emscb.ruhr-uni-bochum.de) |
| 08:07.17 | whymarkwh | dont know, waiting for the guy from pana to get back to me |
| 08:07.47 | whymarkwh | can't noop it kaldemar |
| 08:07.52 | whymarkwh | sorry |
| 08:08.04 | kaldemar | how are you trying to noop it? show the dialplan. |
| 08:08.30 | whymarkwh | exten => #_X.,1,NoOp(****${EXTEN}****) |
| 08:08.52 | whymarkwh | also tried exten => _#X.,1,NoOp(****${EXTEN}****) |
| 08:10.47 | kaldemar | #_X. is plain wrong. patterns start with _. |
| 08:11.41 | whymarkwh | i tried it the other way now i get "Invalid extension '#6', but no rule 'i' in context 'incoming'" |
| 08:12.20 | *** join/#asterisk maxhbp2005 (n=maxhbp20@123.237.12.194) |
| 08:12.39 | maxhbp2005 | hi all |
| 08:12.44 | whymarkwh | hi |
| 08:13.02 | maxhbp2005 | i need to know that how can we take file name which is recorded by one touch recording feature |
| 08:13.56 | maxhbp2005 | it is giving in asterisk cli after stoped of monitro |
| 08:14.10 | maxhbp2005 | but i want that filename in any variable |
| 08:14.14 | maxhbp2005 | is it possible? |
| 08:14.17 | maxhbp2005 | any ideas? |
| 08:16.02 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:17.06 | gr0mit | whymarkwh, please can you tell us the big picture. What pbx you have, how you are connecting it to asterisk , and what you are trying to achieve. |
| 08:22.49 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 08:23.08 | *** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it) |
| 08:23.48 | Mimmus | good morning, is really possible that a single VoIP trunk on a LAN breaks all faxes? |
| 08:24.04 | Mimmus | using aLAW |
| 08:25.12 | *** part/#asterisk maxhbp2005 (n=maxhbp20@123.237.12.194) |
| 08:28.41 | MikeJ | Mimmus: what do you mean? |
| 08:30.29 | Mimmus | I have a PRI/SIP gateway in front of Asterisk |
| 08:31.02 | Mimmus | than a channel-bank directly connected to a PRI board on the Asterisk box |
| 08:31.11 | Mimmus | fax machines are connected to CB |
| 08:31.28 | Mimmus | faxes entering from PRI are almost all broken |
| 08:32.02 | MikeJ | is the pri/sip gateway connected directly to the asterisk box? |
| 08:32.18 | MikeJ | or do you have it on a switch or other gear? |
| 08:32.27 | Mimmus | switch |
| 08:32.34 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 08:32.47 | MikeJ | I would skip the switch.. and just cross connect.. but I doubt that is the issue |
| 08:33.07 | MikeJ | you could have a timing issue on the channel back .. I would guess that is more likely |
| 08:33.11 | Mimmus | I will try... |
| 08:33.46 | *** join/#asterisk rcy (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 08:33.49 | Mimmus | but it is a good quality switch |
| 08:33.56 | Mimmus | HP Procurve |
| 08:34.24 | MikeJ | generally you can;'t count on voip for faxing.. but I know plenty of people who have setups like that that work fine |
| 08:34.33 | MikeJ | but look at timing issues |
| 08:34.51 | Mimmus | I know this but situation is now unmanageable |
| 08:34.52 | MikeJ | and try to test when NOTHING else is happening on asterisk |
| 08:35.25 | Mimmus | I ordered a FXS-SIP gateway with T38 support but it has very looooong times for shipping |
| 08:36.02 | MikeJ | I didn't think asterisk really supported t38 |
| 08:36.40 | Mimmus | No, I will bypass Asterisk, faxes will go straight from PRI gateway to FAX gateway by T38 |
| 08:36.53 | MikeJ | makes sense |
| 08:36.54 | Mimmus | it is the only really supported config |
| 08:37.14 | Mimmus | but in the meanwhile I'd like to alleviate the situqation |
| 08:37.25 | *** join/#asterisk jarod14 (n=jarod14@LMontsouris-152-63-1-19.w80-12.abo.wanadoo.fr) |
| 08:37.43 | MikeJ | I would look for timing issues on the channel bank ast connection or other things on that line.. irq issues and such |
| 08:38.53 | Mimmus | a difficult field... any reference? |
| 08:39.02 | whymarkwh | gromit: i have panasonic TDA 200 where i they have programmed divert from incoming did to go to asteriks on fxo port connected to the panasonic. the panasonic sends down #66550 i need the 6550 to do the routing in asterisk. Now asterisk tels me # is an invalid extension, therefore i can not route the call in asterisk as soon as asterisk sees the # it considers it to be an invalid extension. |
| 08:39.16 | whymarkwh | hope that make sence. |
| 08:39.58 | *** join/#asterisk blogbasti (n=blogbast@calypso.planet-ic.de) |
| 08:51.32 | gr0mit | whymarkwh, so you should be looking to have a line along exten => _#6XXXX,1,(do-wotever) |
| 08:52.33 | gr0mit | however, if you are using an ATA of some sort, be aware that #nn is often used to initiate supplemnetary services |
| 08:52.35 | MikeJ | whymarkwh: make the # part of your extension |
| 08:52.49 | MikeJ | Mimmus: there should be stuff on google "asterisk irq" |
| 08:52.54 | gr0mit | so you might see strange things |
| 08:53.02 | Mimmus | MikeJ: I will try |
| 08:59.59 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 09:00.03 | gr0mit | Mimmus, make sure you have echocancelwhenbridged=no |
| 09:00.10 | ghenry | Can you place calls via web service requests to * ? |
| 09:00.17 | whymarkwh | gr0mit: you right i see strange thing one time i get 625 then just a 0 the a 3 then a 66255 without the last digit. i am baffled |
| 09:00.18 | gr0mit | asterisk echo canceller really messes with modems |
| 09:00.41 | whymarkwh | i am using a diguim 4port fxo card |
| 09:01.05 | gr0mit | aah ok |
| 09:01.08 | yang | ~seen SteveTotaro |
| 09:01.09 | jbot | stevetotaro <n=Administ@pool-70-17-230-174.balt.east.verizon.net> was last seen on IRC in channel #asterisk, 13d 12h 41m 7s ago, saying: 'i can do a dial from the h exten using a local chan'. |
| 09:01.20 | gr0mit | do you not have any ISDN ports on the box? |
| 09:01.45 | MikeJ | gr0mit: good point.. Mimmus tone detection can cuase problems for faxes too |
| 09:03.11 | gr0mit | whymarkwh, in my experience, avoid any analogue interconnects like the plague. they are nothing but Big Trouble. |
| 09:28.24 | *** join/#asterisk magenbrot (n=magenbro@ov.odn.de) |
| 09:30.24 | *** join/#asterisk magenbrot (n=magenbro@ov.odn.de) |
| 09:32.47 | *** part/#asterisk sircco (n=sircco@dh207-69-105.xnet.hr) |
| 09:42.25 | *** join/#asterisk magenbrot (n=magenbro@ov.odn.de) |
| 09:43.21 | *** join/#asterisk magenbrot (n=magenbro@ov.odn.de) |
| 09:44.46 | *** join/#asterisk magenbrot (n=magenbro@ov.odn.de) |
| 09:47.08 | *** join/#asterisk magenbrot (n=magenbro@ov.odn.de) |
| 09:47.09 | Mimmus | sorry for absence |
| 09:48.50 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 09:48.53 | implicit | in #asterisk-bugs |
| 09:49.00 | implicit | is anyoen there |
| 09:51.17 | Mimmus | gr0mit: I had echocancelwhenbridged=yes, good point |
| 09:51.38 | gr0mit | ok, so change that to no |
| 09:52.02 | Mimmus | done, where is "tone detection" ? |
| 09:56.15 | gr0mit | dont worry bout that. |
| 09:56.24 | gr0mit | you will need to restart asterisk |
| 09:56.35 | gr0mit | then faxes should be fine...(i hope!) |
| 09:58.06 | Mimmus | gr0mit: reload chan_zap.so is not enough? |
| 10:01.41 | *** join/#asterisk MaliutaLap (n=nikolai@kiev.lusan.id.au) |
| 10:02.36 | tzafrir_laptop | module unload chan_zap.so |
| 10:02.42 | tzafrir_laptop | module load chan_zap.so |
| 10:02.43 | tzafrir_laptop | ? |
| 10:03.22 | tzafrir_laptop | err.... sorry, wasn't reading.. |
| 10:04.15 | gr0mit | well try it |
| 10:08.12 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 10:10.38 | *** join/#asterisk ZefK (n=ZefK@wsc-fo.b.astral.ro) |
| 10:19.36 | *** join/#asterisk MaliutaLap (n=nikolai@kiev.lusan.id.au) |
| 10:24.55 | *** join/#asterisk FabiOne (n=FabiOne@151.13.190.20) |
| 10:24.58 | FabiOne | hi all |
| 10:29.50 | *** join/#asterisk FciSoft (n=FabiOne@151.13.190.20) |
| 10:30.13 | FciSoft | i've a big problem with qualify=yes in my sip.conf |
| 10:30.43 | FciSoft | the *'s cli will be flooded with |
| 10:30.44 | FciSoft | [Sep 25 12:29:19] NOTICE[27163]: chan_sip.c:12780 handle_response_peerpoke: Peer '0941630003' is now Reachable. (97ms / 2000ms) |
| 10:31.18 | FciSoft | even the ssh connection go down |
| 10:31.51 | *** join/#asterisk kotique (n=picachu@host-static-89-41-72-115.moldtelecom.md) |
| 10:32.04 | kotique | IF(${REGEX("^(170[123456]|11800)$" "${EXTTOCALL}"})?${SetVar(_SPYGROUP=spyit)}) |
| 10:32.10 | kotique | how do I write this correctly ? |
| 10:49.55 | *** join/#asterisk henk (n=hank@netwichtig.de) |
| 10:50.00 | henk | hi |
| 10:51.51 | *** join/#asterisk simNIX (n=simNIX@82-204-21-111.dsl.bbeyond.nl) |
| 10:56.26 | kotique | is it possible to execute application inside application in dialplan ? |
| 10:56.46 | *** join/#asterisk whymarkwhy (n=dsfsdfsd@196.211.34.2) |
| 11:02.51 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 11:03.16 | *** part/#asterisk a-s (n=user@89.38.174.194) |
| 11:05.31 | Mimmus | gr0mit: nope, faxes fails with typical voip problems |
| 11:06.56 | *** join/#asterisk future (n=future@balancer.phuture.sk) |
| 11:08.50 | *** join/#asterisk SpeedDragon (n=SpeedDra@sm4-84-90-136-254.netvisao.pt) |
| 11:09.39 | kotique | <PROTECTED> |
| 11:09.42 | kotique | wtf |
| 11:10.50 | tzafrir_laptop | kotique, could you paste the full line? |
| 11:11.04 | tzafrir_laptop | (from extensions.conf) |
| 11:11.06 | kotique | exten => s,n,ExecIf(${REGEX("^(170[123456]|11801)$" ${EXTTOCALL})},Set(_SPYGROUP=spy)) |
| 11:12.18 | kotique | Executing [s@macro-record-enable123:3] ExecIf("SIP/11801-b580aef0", "1|Set(_SPYGROUP=spy)|") in new stack |
| 11:13.19 | kotique | is there any way to do it otherwise ? |
| 11:13.20 | tzafrir_laptop | henk, hi |
| 11:13.26 | kotique | like 1 ? shit : shit2 |
| 11:13.45 | henk | tzafrir_laptop: hey :) hows it going? |
| 11:14.53 | *** join/#asterisk salzh (n=root@116.232.40.78) |
| 11:20.24 | kotique | execif 1,2,3 |
| 11:23.51 | *** join/#asterisk kalel008 (n=tamato@209.203.41.26) |
| 11:23.56 | kalel008 | hi |
| 11:24.05 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177581966.dsl.bell.ca) |
| 11:24.12 | kalel008 | i have setup sendmailand tested it |
| 11:24.35 | kalel008 | how can i manually send the vmessage |
| 11:25.18 | kalel008 | to test if asterisk can send email |
| 11:28.04 | tzafrir_laptop | hmm... to answer kotique: Set,_SPYGROUP=spy |
| 11:28.52 | tzafrir_laptop | can you send a message with mail / mailx / mutt ? |
| 11:28.53 | kalel008 | anyone know? |
| 11:29.03 | kalel008 | oh letme try |
| 11:29.40 | tzafrir_laptop | echo test body | mail -s "test subject" yourname@example.com |
| 11:29.49 | kalel008 | yeh that works |
| 11:30.01 | kalel008 | using sendmail |
| 11:30.43 | kalel008 | but asteriskdoesnt send anything |
| 11:32.04 | tzafrir_laptop | do you see anything in the logs of the mail server? Which is it, BTW? |
| 11:32.11 | tzafrir_laptop | sendmail? postfix? exim? |
| 11:32.18 | kalel008 | sendmail |
| 11:33.06 | tzafrir_laptop | do you see anything in the logs? |
| 11:33.11 | tzafrir_laptop | in mailq ? |
| 11:33.20 | kalel008 | let me check |
| 11:34.27 | kalel008 | only shows the tests i send |
| 11:34.33 | kalel008 | nothing from asterisk |
| 11:35.55 | kalel008 | its like asterisk doesnt do anything |
| 11:36.11 | kalel008 | and i have aded the email in the voicemail.conf |
| 11:37.39 | kalel008 | so i dont know |
| 11:39.09 | kalel008 | were is the local mailboxes saved for sendmail |
| 11:39.14 | *** join/#asterisk af_ (n=getsmart@88-149-241-240.dynamic.ngi.it) |
| 11:40.36 | kalel008 | because it could be sending to the root alias |
| 11:40.38 | tzafrir_laptop | kaldemar, can you pastebin your voicemail.conf? Possibly with masked-out emails and passwords? |
| 11:44.56 | kalel008 | http://pastebin.com/m5998cf87 |
| 11:47.36 | *** join/#asterisk mateo_au (n=chatzill@12.144.159.231) |
| 11:58.12 | *** join/#asterisk coppice (n=chatzill@177.162.17.210.dyn.pacific.net.hk) |
| 11:59.38 | kalel008 | lol sorry wasnt me right |
| 12:02.15 | kalel008 | anyone know how to use sendmail or should i use something else |
| 12:04.10 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:09.44 | *** join/#asterisk sabo-subotica (n=asd@79.101.28.227) |
| 12:10.21 | sabo-subotica | Hi! i have a problem with asterisk may i ask for help? |
| 12:11.04 | kalel008 | maybe just start by asking :) |
| 12:11.21 | sabo-subotica | ok:) |
| 12:11.43 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:11.58 | sabo-subotica | im having problems with an analogue telephone line attached to an FXO card, im getting some random inbound calls like every 5-10 minutes |
| 12:12.47 | sabo-subotica | when i attack an analogue telephone on that line i get no inbound calls, so i suppose there is nothing wrong with the line itself but the configuration of asterisk(or the fxo card itself) |
| 12:14.40 | DarKnesS_WolF | how can i save sip debug into a log file ? |
| 12:15.07 | tzafrir_laptop | The problem is that after you attacked the analogue phone it broke down :-( |
| 12:15.25 | kalel008 | lol |
| 12:16.00 | sabo-subotica | i did not attack anything |
| 12:16.44 | *** join/#asterisk mchou (n=mchou@c-76-103-44-118.hsd1.ca.comcast.net) |
| 12:17.18 | kalel008 | i would suggest to check your context in the cards config |
| 12:17.28 | kalel008 | i think i could be wrong tho |
| 12:17.45 | tzafrir_laptop | sabo-subotica, where are you from? |
| 12:17.50 | tzafrir_laptop | What country? |
| 12:18.01 | sabo-subotica | Serbia |
| 12:18.10 | tzafrir_laptop | Can you please pastebin your zapata.conf? |
| 12:18.41 | sabo-subotica | [channels] |
| 12:18.41 | sabo-subotica | usecallerid=no |
| 12:18.41 | sabo-subotica | echocancel=yes |
| 12:18.41 | sabo-subotica | echocancelwhenbridged=no |
| 12:18.41 | sabo-subotica | #echotraining=800 |
| 12:18.42 | sabo-subotica | rxgain=0.0 |
| 12:18.44 | sabo-subotica | txgain=0.0 |
| 12:18.46 | sabo-subotica | signalling=fxs_ls |
| 12:18.46 | tzafrir_laptop | ~pb |
| 12:18.47 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 12:18.48 | sabo-subotica | ;group=0 |
| 12:18.50 | sabo-subotica | context=analog |
| 12:18.52 | sabo-subotica | channel=1 |
| 12:18.54 | sabo-subotica | signalling=fxo_ls |
| 12:18.56 | sabo-subotica | ;group=1 |
| 12:18.58 | sabo-subotica | context=internal |
| 12:19.00 | sabo-subotica | channel=2 |
| 12:19.07 | tzafrir_laptop | sabo-subotica, this is called flooding |
| 12:19.15 | *** join/#asterisk mtx2 (n=mtx@66.226.228.204.cpe.speedyquick.net) |
| 12:19.24 | sabo-subotica | sorry |
| 12:19.26 | tzafrir_laptop | considered annoying by many people... |
| 12:19.26 | kaldemar | tzafrir_laptop: i refuse to paste my voicemail.conf ;) |
| 12:19.26 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:21.23 | mtx2 | i have a config problem with getting a second pri working on my tormenta quad pri card |
| 12:22.33 | mtx2 | could use a bit of config help? any one have experience configuring a second pri? |
| 12:23.05 | viraptor | is there any nice way to change asterisk database externally? (I'm worried about concurrent access) without doing `asterisk -rx 'database put ...'`? |
| 12:29.10 | [TK]D-Fender | mtx2: pastebin your zaptel.conf & zapata.conf |
| 12:29.11 | [TK]D-Fender | ~pb |
| 12:29.12 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 12:30.08 | [TK]D-Fender | viraptor: I'f you're worried about concurrency then thats the only way I can see. Otherwise its just a BDB file |
| 12:31.45 | tzafrir_laptop | sabo-subotica, I can't think of any reason. Strange |
| 12:32.11 | tzafrir_laptop | Can you connect a phone alongside the FXO and see if it rings when you get the random rings? |
| 12:33.25 | mtx2 | D-Fender: http://pastebin.com/m10c989ac |
| 12:35.52 | [TK]D-Fender | mtx2: should be span=1,1,0 then span 2,2,0 |
| 12:36.02 | mtx2 | everything looks right, but i get the message "==Primary D-Channel on span 2" up 4 time and theen pri_find_dchan: No D-chaannels available. |
| 12:36.08 | *** join/#asterisk andrewyager (n=andrewya@30-53-145-203.whitelabelinternet.com.au) |
| 12:36.36 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 12:36.59 | kalel008 | were can i check what is wrong sendmail does work but asterisk not sending anything |
| 12:38.56 | lyroy | voicemail.conf.... attach=yes |
| 12:39.07 | mtx2 | D-Fender: it makes no difference which span does the timing, I still get the same error |
| 12:39.31 | [TK]D-Fender | mtx2: PB - cat /proc/interrupts |
| 12:39.53 | sabo-subotica | tzafrir_laptop i triend to attach an analogue phone on the phone line and i didnt get any inbound calls |
| 12:40.14 | sabo-subotica | tho i didnt connect the analogue phone and the asterisk card at the same time, perhaps i should check it |
| 12:40.45 | kalel008 | yeh it is there attach = yes under servermail = |
| 12:42.09 | mtx2 | D-Fender: the t1 is actually connected and "greened" up |
| 12:42.14 | mtx2 | D-Fender: http://pastebin.com/d77cf85d3 |
| 12:46.37 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 12:46.38 | Mimmus | hi, I'm looking for best zapata.conf settings to use with a PRI card with a channel-bank attached |
| 12:46.48 | Mimmus | I have only fax machines on CB |
| 12:46.58 | Mimmus | I just disabled echo cancelling |
| 12:47.35 | [TK]D-Fender | mtx2: a TOR2 in a dual quad-core CPU system? |
| 12:49.19 | riddlebox | Mimmus, whats the problem? |
| 12:50.01 | mtx2 | D-Fender: actually, it has 4 1.9 Ghz Xeon Processors in it, hyper threaded. |
| 12:50.08 | Mimmus | riddlebox: sorry, I already spoke about this a couple of hours ago |
| 12:50.30 | Mimmus | I have a PRI/SIP gateway in fron of Asterisk |
| 12:50.49 | Mimmus | and a channel-bank connected to an internal PRI board |
| 12:51.08 | Mimmus | faxes entering from PRI and directed to fax machines are almos always BROKEN |
| 12:51.20 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:51.34 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 12:51.36 | riddlebox | Mimmus, what do you mean by broken? |
| 12:51.49 | Mimmus | I know that with VoIP it is not a good setup but I have only a LAN trunk with aLaw |
| 12:51.51 | [TK]D-Fender | mtx2: Try other portson it just for fun. |
| 12:51.56 | [TK]D-Fender | (the card) |
| 12:52.08 | Mimmus | riddlebox: incomlete, errors and MANY complaints from users |
| 12:52.56 | riddlebox | Mimmus, so what kind of line are you using a pots line from the PSTN, or VoIP from a provider? |
| 12:53.25 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:53.26 | Mimmus | riddlebox: PRI line from telco |
| 12:53.49 | riddlebox | Mimmus, thats not voip then |
| 12:54.24 | Mimmus | riddlebox: yes, SIP from gateway to Asterisk |
| 12:54.31 | riddlebox | ohhh |
| 12:54.58 | Mimmus | but they are on the same switch and I use G711 aLaw |
| 12:55.06 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 12:56.24 | riddlebox | Mimmus, forgive me, I am sick right now and a little slow, isnt there something like T.39 or something for faxing over voip? |
| 12:56.34 | Mimmus | I know that it is not a reliable setup but I'm not able to explain this high rat eof failures |
| 12:56.38 | mtx2 | D-Fender:moving to span3 ... |
| 12:58.09 | riddlebox | Mimmus, and you can put a butt set on the ports to the fax machines and make and recieve calls ok? |
| 12:59.18 | Mimmus | riddlebox: yes, voice call are OK, fax are also OK if I connect PRI line directly to Asterisk PRI board |
| 12:59.30 | *** join/#asterisk BiG_NoBoDy (n=ruslanas@77.221.67.60) |
| 12:59.46 | BiG_NoBoDy | hello |
| 12:59.51 | BiG_NoBoDy | is any one alive? |
| 13:00.01 | andrewyager | [TK]D-Fend: we were chatting the other day about my issue with queues not recording the call length; upgrade - still no joy. |
| 13:00.12 | riddlebox | Mimmus, why not just put the pri directly to asterisk then? |
| 13:01.21 | Mimmus | why board is a first generation Sangoma card and it supports only E1 (like the channelbank) or T1 (like PRI line), not mixed config on its ports |
| 13:01.32 | Mimmus | and then I prefer external gateway |
| 13:01.53 | BiG_NoBoDy | may be someone knows how to make asterisk send fax to an email when it is retrieved |
| 13:02.00 | riddlebox | hrmm |
| 13:02.23 | [TK]D-Fender | BiG_NoBoDy: "core show applicaion system" |
| 13:02.27 | [TK]D-Fender | BiG_NoBoDy: "core show application system" |
| 13:02.43 | Mimmus | riddlebox: I know it is a working setup because I'm using it in another site with another board (Digium :-)) |
| 13:02.44 | riddlebox | BiG_NoBoDy, I think a quick google search on that one will give you lots of results |
| 13:02.45 | mtx2 | D-Fender: updated to span3 - anything look different to you ? http://pastebin.com/d32c058c2 |
| 13:02.52 | mtx2 | D-Fender: updated to span3 - anything look different to you ? http://pastebin.com/d32c058c2 |
| 13:03.04 | mtx2 | D-Fender: updated to span3 - anything look different to you ? |
| 13:03.14 | mtx2 | D-Fender: http://pastebin.com/d32c058c2 |
| 13:03.30 | mtx2 | d |
| 13:03.36 | BiG_NoBoDy | riddlebox > it shows but it does not work ... |
| 13:03.42 | BiG_NoBoDy | i might be a looser :) |
| 13:03.53 | mtx2 | d-fender: http://pastebin.com/d32c058c2 |
| 13:04.01 | riddlebox | BiG_NoBoDy, can you pastebin your context dealing with it |
| 13:04.01 | *** part/#asterisk mtx2 (n=mtx@66.226.228.204.cpe.speedyquick.net) |
| 13:04.49 | riddlebox | Mimmus, try slowing down the baud rate on the ax machines |
| 13:04.59 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 13:05.00 | riddlebox | s/ax/fax |
| 13:05.06 | BiG_NoBoDy | i came to work and here is working debian with postfix, asterisk 1.6, hylafax, and avantfax |
| 13:05.48 | Mimmus | riddlebox: not so simple, it is a server with fax/modem cards, I'm not an expoert of AT commands! |
| 13:06.16 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 13:07.46 | rbd | hey guys. I'd like to run asterisk (with meetme, etc) on blade servers...unfortunately they have no expansion slots (obviously)...so I couldn't use an X100P for timing....looks like it'd have to be ztdummy... any alternatives here? would this keep me from going to blades (is ztdummy that much worse? I'd have a few hundred folks in meetme rooms per box) |
| 13:09.02 | Mimmus | ztdummy is OK fro me |
| 13:10.05 | [TK]D-Fender | rbd: no cards = ztdummy. All there is to it |
| 13:10.12 | riddlebox | Mimmus, I am trying to read up and see, cant promise anything |
| 13:11.54 | *** join/#asterisk mtx2 (n=mtx@66.226.228.204.cpe.speedyquick.net) |
| 13:12.28 | *** join/#asterisk stimpie (n=stimpie@84-104-5-14.cable.quicknet.nl) |
| 13:12.28 | *** join/#asterisk ToTo (n=ToTo@207.176.6.64) |
| 13:13.08 | andrewyager | hi, the problem i'm having is that in my queue log the total call time is recorded, as well as channel information, but the actual length of call with the operator is reported as 0 seconds, and the log seems to suggest the call started and ended at the same time |
| 13:13.09 | *** join/#asterisk mtx (n=mtx@66.226.228.204.cpe.speedyquick.net) |
| 13:13.12 | andrewyager | any suggestions? |
| 13:13.22 | Mimmus | riddlebox: thanks, if I have only channel-bank on the asterisk board, how have I to set clock? |
| 13:13.27 | rbd | [TK]D-Fender: yeah I'm just seeing if ztdummy is good enough for meetme in 2.6...it looks like with recent updates, it is |
| 13:13.34 | rbd | meetme, audio streaming, etc |
| 13:13.42 | mtx | D-Fender: http://pastebin.com/d32c058c2 - does this look right? |
| 13:14.37 | *** join/#asterisk wscholar (n=wayne@214.sub-75-208-249.myvzw.com) |
| 13:18.29 | *** part/#asterisk MindTheGap_ (n=MindTheG@201.80.60.227) |
| 13:19.30 | *** join/#asterisk l2trace99 (n=jr@75.112.133.235) |
| 13:20.55 | *** join/#asterisk AndyMillar (i=freenode@andymillar.co.uk) |
| 13:21.17 | riddlebox | bw [TK]D-Fender I called trisys the other day, and heard the friendly voice of allison while I was transfered to the queue |
| 13:21.32 | AndyMillar | mh, does anyone have any good indications of the system requirements for asterisk? |
| 13:21.37 | tzafrir_laptop | Mimmus, how is that channel bank connected? zap card? if so it can provide timing |
| 13:22.05 | AndyMillar | i.e. how I can work out what hardware I'd need to run 10/50/100 simultaneous calls? |
| 13:23.17 | tzafrir_laptop | 100 ulaw calls? the PC you connect from |
| 13:23.20 | BiG_NoBoDy | OS:debian asterisk 1.6, hylafax, and avantfax |
| 13:23.28 | Mimmus | tzafrir_laptop: cb is connected by a PRI E1 interface to a Sangoma PRI card |
| 13:23.33 | BiG_NoBoDy | what else is needed ? |
| 13:23.43 | AndyMillar | tzafrir_laptop: 100 different sip handsets |
| 13:23.56 | *** join/#asterisk tobias (n=tobias@user-0c8hj2l.cable.mindspring.com) |
| 13:24.40 | tzafrir_laptop | AndyMillar, 100 handsets on the LAN? No recordings? No conferences? Any recent PC will be much more than enough |
| 13:25.30 | AndyMillar | tzafrir_laptop: hopefully no conferences, all recording |
| 13:25.47 | tzafrir_laptop | Mimmus, yes, that card provides zaptel timing. You can have your meetme. |
| 13:26.02 | *** join/#asterisk CtRiX (n=CtRiX@aretha.navynet.it) |
| 13:26.02 | tzafrir_laptop | To test that you do have zaptel timing, try running zttest |
| 13:26.25 | riddlebox | BiG_NoBoDy, a way to send emails from the asterisk box |
| 13:27.16 | riddlebox | tzafrir_laptop, Mimmus is having issues with faxing on his system |
| 13:27.16 | AndyMillar | tzafrir_laptop: i'm thinking something along the lines of a quad-core 2GHz (or more) xeon with 4-8GB RAM as a box for this |
| 13:27.45 | EmleyMoor | Is there (in the book, perhaps) a good reference to logical operations in asterisk? |
| 13:27.46 | BiG_NoBoDy | riddlebox > at the time it is sending avantfax (but numbers it has in contact list) over postfix (as i understand |
| 13:28.56 | tzafrir_laptop | AndyMillar, if what you describe here are all the tasks of that PC, than a quad core is a gross overkill. |
| 13:29.01 | *** join/#asterisk bbryant (n=Brett_Br@adsl-153-42-209.chs.bellsouth.net) |
| 13:29.25 | tzafrir_laptop | Likewise I believe anything beyond 2GB |
| 13:29.40 | BiG_NoBoDy | i have tried kirtsana.info/2007/11/08/trixbox-2303-with-postfix-iaxmodem-hylafax-and-avantfax-working-perfectly/ manual but it does not send |
| 13:29.43 | EmleyMoor | Are you trying to run a village PSTN on it? <g> |
| 13:29.45 | tzafrir_laptop | Unless you want to make a large ramdrive and reocrd stuff to there |
| 13:29.55 | [TK]D-Fender | AndyMillar: I would advise a fast raid 5 HD system though |
| 13:30.23 | riddlebox | BiG_NoBoDy, on the avantfax website it says you can forward faxes via email from avantfax |
| 13:30.25 | tzafrir_laptop | AndyMillar, spend a bit more money on reliability of the system than on pure power, maybe |
| 13:30.48 | rbd | hey guys...anyone have any idea of the # of channels that one could support on a 2x quad core using g711 (half in IVR, half on meetme)? looked at asterisk dimensioning page, but I was wondering if anyone else had any info. for now I am assuming around 500 channels... |
| 13:31.39 | BiG_NoBoDy | riddlebox > only numbers you know that are in contact list, but others stay in avant fax |
| 13:32.40 | [TK]D-Fender | rbd: IVR people = irrelevent, Meetme will be the issue |
| 13:32.55 | tzafrir_laptop | rbd, test for yourself |
| 13:33.00 | BiG_NoBoDy | i made a FaxDispatch file as it is written in manual i provided erlier |
| 13:33.20 | tzafrir_laptop | take a number of mighty servers and bombard it with calls |
| 13:33.52 | *** join/#asterisk coolthreads (n=shane@203-97-238-71.cable.telstraclear.net) |
| 13:35.01 | tzafrir_laptop | while true; do asterisk -rx 'originate SIP/tested-peer/ivr application Pleayback 1-minute' sleep 1; done |
| 13:35.05 | riddlebox | BiG_NoBoDy, so you want it to be automatically emailing faxes to someone when they come in? |
| 13:35.43 | tzafrir_laptop | (where '1-minute' is a sound file that takes 1 minute, the length of it and the sleep time are parameters for the bombardment) |
| 13:36.10 | AndyMillar | tzafrir_laptop: overkill is better that getting any jitter though (in this case) |
| 13:36.28 | BiG_NoBoDy | riddlebox > yes! that would be very greate! |
| 13:36.40 | riddlebox | BiG_NoBoDy, would all faxes go to one person? |
| 13:36.45 | BiG_NoBoDy | yes |
| 13:36.49 | rbd | tzafrir_laptop: I would if I had the hardware :) ... doing some cost estimations now |
| 13:37.49 | rbd | [TK]D-Fender: so even if meetme is not doing any significant transcoding (just 711 to pcm and back I'd assume), it will still impose a significant load? |
| 13:37.53 | *** join/#asterisk Chris-NB (n=chris@nfw.ecos.at) |
| 13:38.34 | [TK]D-Fender | rbd: cumulatively, yes |
| 13:39.35 | *** join/#asterisk ToTo (n=ToTo@207.176.6.38) |
| 13:39.55 | rbd | [TK]D-Fender: ok, I did look into other conference apps, like app_conference... anything viable meetme alternatives on the horizon that you know of? I think app_conference has fallen a bit behind (not in the trunk and all) |
| 13:40.02 | Kobaz | is there a way to check for a valid agent login without using AgentLogin() |
| 13:40.20 | Kobaz | i'm doing a custom login and using AddQueueMember |
| 13:40.26 | Kobaz | but i wanna check for a valid password |
| 13:40.29 | [TK]D-Fender | Kobaz: Write your own code. |
| 13:40.54 | Kobaz | that's what i thought |
| 13:42.06 | *** join/#asterisk mog (n=mog@nat/digium/x-b3b99c39ffa766ad) |
| 13:42.06 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:42.26 | riddlebox | BiG_NoBoDy, you can always get into asterisk and edit your incoming context so that after it is done receiving the fax then, use a System() command to email it to the person |
| 13:43.23 | BiG_NoBoDy | riddlebox > could you say where ? because i am noob to avant fax and beginner user to linux |
| 13:43.39 | BiG_NoBoDy | i know that some file from /etc/asterisk/* |
| 13:44.09 | riddlebox | BiG_NoBoDy, yes you would probably edit /etc/asterisk/extensions.conf |
| 13:45.21 | AndyMillar | tzafrir_laptop: so you suspect that for ~100 users or so, a quad core xeon with ~8GB RAM and 8 SAS disks in RAID5/6/10 is overkill? |
| 13:45.22 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177581966.dsl.bell.ca) |
| 13:46.26 | BiG_NoBoDy | riddlebox > em what option ? sould i search? |
| 13:46.53 | riddlebox | BiG_NoBoDy, NvFaxdetect or something like that |
| 13:47.02 | jameswf-home | so when does *zap* become fully depricated I noticed it is no longer in the 1.4 trunk |
| 13:47.19 | Katty | [TK]D-Fender: i got pictures of the pups. shall i forward them to you? |
| 13:47.36 | tzafrir_laptop | jameswf-home, zaptel is fully supported in 1.4 . chan_zap has been renaned chan_dahdi, though |
| 13:47.51 | tzafrir_laptop | zaptel is not supported in 1.6.0 and beyond |
| 13:47.54 | riddlebox | BiG_NoBoDy, http://www.hylafax.org/man/current/faxrcvd.1m.html |
| 13:48.25 | [TK]D-Fender | Katty: When you've actually got them. |
| 13:48.26 | jameswf-home | tzafrir_laptop: so in a near release we will have to switch configs |
| 13:48.33 | Katty | [TK]D-Fender: k |
| 13:48.46 | AndyMillar | tzafrir_laptop: as for what i'm doing, i like overkill |
| 13:49.09 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
| 13:49.54 | BiG_NoBoDy | riddlebox > as i understand |
| 13:50.04 | BiG_NoBoDy | riddlebox > can i pm you ? |
| 13:50.16 | riddlebox | yeah its fine |
| 13:50.53 | tzafrir_laptop | jameswf-home, in 1.4.x: no . But zapata.conf is officially deprecated as of 1.4.22 |
| 13:51.41 | mtx2 | D-Fender: switching to span 3 worked with the only differences as you see them in the paste bin. http://pastebin.com/d32c058c2 |
| 13:52.49 | Kobaz | hmm |
| 13:54.06 | Mimmus | thanks for your support, there is no chance to get a decent percentage of success without T38 |
| 13:55.27 | *** join/#asterisk c4t3l (n=root@74.95.210.124) |
| 13:55.51 | c4t3l | yodel! |
| 13:56.19 | [TK]D-Fender | ay-he-hooo |
| 13:56.40 | c4t3l | hows the world of asterisk today? |
| 13:56.41 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 13:56.54 | Assid | crap my e61 doesnt want to connect my asterisk box |
| 13:57.22 | c4t3l | astricon ends today huh? |
| 14:04.46 | c4t3l | does anybody here know where one could acquire some old AT&T (Westinghouse?) swtiching equipment |
| 14:05.08 | c4t3l | for collector's purposes... |
| 14:05.26 | riddlebox | hrmm told Mimmus that earlier |
| 14:05.50 | c4t3l | I'd like to have an old school switchboard in my garage :) |
| 14:06.37 | BrianR___ | Heh. My mom worked on one of those when I was a kid. |
| 14:07.01 | c4t3l | cool |
| 14:07.06 | BrianR___ | a lot of answering service switchboards from the pre-bell breakup era actually belonged to the CO |
| 14:08.33 | c4t3l | man. i just want to see one in person. I dont think photographs do them justice |
| 14:11.35 | *** join/#asterisk Blackvel (n=blackvel@dslb-084-057-068-147.pools.arcor-ip.net) |
| 14:14.14 | Blackvel | hi all. when programming an IVR, what programming techniques do you prefer depending on some ivr parts? goto (context), gosub or Macro? |
| 14:14.36 | Blackvel | Is there any rule what to prefer over the other? e.g Macro vs gosub? When its best to to program with "goto" instead of building the specific ivr part with gosub? |
| 14:15.33 | c4t3l | i think thats really up to the individual doing the programming |
| 14:16.06 | c4t3l | subjective perhaps |
| 14:16.11 | [TK]D-Fender | Blackvel: this has nothing to do with IVR's |
| 14:16.12 | *** join/#asterisk seanmh (i=HydraIRC@216.31.101.83) |
| 14:16.19 | [TK]D-Fender | Blackvel: All dialplan is jsut dialplan. |
| 14:17.01 | [TK]D-Fender | Blackvel: Macros are for performing similar process with different args. Gosub is a macro without args where everything is determined by existing vars, if even needed. |
| 14:17.18 | c4t3l | If you wanna go for the gusto, you should look into agi programming |
| 14:17.39 | [TK]D-Fender | Blackvel: Everything depends on the flow of what you actually need to do. You wouldn't even use either unless there was redundency savings to be had. |
| 14:18.05 | c4t3l | [TK]D-Fender: very true |
| 14:18.12 | [TK]D-Fender | Blackvel: And AGI is only for things you can't do in standard logic |
| 14:18.57 | Blackvel | so far it looks like I can live without AGI |
| 14:19.05 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-ba2d1eb84878494b) |
| 14:19.05 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:19.07 | Blackvel | i dont want to overcomplicate things |
| 14:20.49 | [TK]D-Fender | Blackvel: "ask your doctor if Moderately complicated" may be right for yoU! |
| 14:20.54 | c4t3l | Simple dialplans are best. For many reasons. Especially for parsing cli output if there's a wierd issue. So keep that in mind as well |
| 14:21.13 | Blackvel | already used a Macro with one args. ahh I see. so you use gosub over Macro when there is no need for arguments (auto return). |
| 14:21.35 | Blackvel | just trying to break the big ivr thing into smaller parts (therefore need for different contexts) with gosub or goto |
| 14:22.13 | Blackvel | to me it just looks like that I could either implement two ivr parts either as gosub (with jump back and continue in first part) or goto (no return) |
| 14:22.39 | c4t3l | I once had to troubleshoot someone elses dialplan. They had one call iterate through 65 instructions before the call was connected. This is not a joke |
| 14:23.07 | Blackvel | connected with answer or dial? |
| 14:23.17 | c4t3l | both |
| 14:24.17 | c4t3l | I'm not gonna name names (cuz its a commercial product based on *), but the calls took like 3 seconds to connect after pickup due to the hoops |
| 14:24.33 | Blackvel | just trying to put this ivr in front of my business phone so I can provide some informations before (dont want to answer each time the same questions on phone for nothing) |
| 14:24.40 | *** join/#asterisk mchou (n=mchou@c-76-103-44-118.hsd1.ca.comcast.net) |
| 14:24.58 | jksM | anyone using DECT-to-SIP gateways with DECT headsets have solved the problem of getting a "dial tone" when using the "answer" button on the headset? |
| 14:25.03 | Blackvel | whats okay? 1 sec? |
| 14:25.09 | c4t3l | IVRs are pretty easy once you get the hang of dialplan programming |
| 14:25.52 | Blackvel | i am trying to build it as modular as possible (different contexts etc...) |
| 14:26.09 | *** join/#asterisk moy (n=moy@63-255-103-7.ip.mcleodusa.net) |
| 14:26.15 | Blackvel | okies...thanks for your recommendations, guys |
| 14:26.21 | Blackvel | back to work :)# |
| 14:26.49 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:27.40 | *** join/#asterisk feqma (n=warriner@24.75.56.66) |
| 14:27.56 | *** join/#asterisk qwertyfcuker (n=psetti1@www.acepayroll.com) |
| 14:28.19 | qwertyfcuker | hello folks |
| 14:29.00 | jaytee | if I'm using SIPAddHeader to add an Alert-info do I have to do it on the line right before the Dial app or can I use it in another context right before the call uses Goto to jump to it? |
| 14:29.38 | qwertyfcuker | quick question: I'd like to connect asterisk to an NEC ksu at a remote locations. The NEC unit has a digital station card.....What hardware do I need on the asterisk box to connect to the NEC box? I have been trying to do it w/ a Sangoma card w/ 4 FXO modules, but it doesn't seem to work |
| 14:31.02 | [TK]D-Fender | jaytee: Anytime before the dial |
| 14:31.36 | jaytee | [TK]D-Fender, thanks man! |
| 14:31.37 | [TK]D-Fender | qwertyfcuker: You cannot connect to digital trunk lines unless its a T1/E1/J1 signalled link |
| 14:31.53 | Assid | http://assid.pastebin.com/d27dadaa0 -- my e61 doesnt want to register - please help |
| 14:31.59 | [TK]D-Fender | qwertyfcuker: Which is quite likely not the case |
| 14:32.07 | c4t3l | NEC digital station card sounds very proprietary to me. Chances are you wont find any easy answer for this one... |
| 14:32.52 | qwertyfcuker | i'm sure it's proprietary, generally only their own phones work |
| 14:33.12 | [TK]D-Fender | qwertyfcuker: At which point there is nothign to be done for it |
| 14:33.19 | c4t3l | :( |
| 14:33.24 | qwertyfcuker | :( |
| 14:33.36 | qwertyfcuker | i could probably use analog lines though right? |
| 14:34.09 | [TK]D-Fender | qwertyfcuker: with those cards, yes |
| 14:34.39 | qwertyfcuker | thank you guys very much |
| 14:35.39 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
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| 14:36.22 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:37.11 | Assid | hrmm nv,.. got it working. |
| 14:37.20 | Assid | needs qualify= apparently |
| 14:38.36 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 14:41.18 | *** join/#asterisk Defraz (n=T0tal@fw.fuzecore.com) |
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| 14:50.53 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:51.58 | *** join/#asterisk furet (n=furet@clubnix100.esiee.fr) |
| 14:52.01 | furet | hello |
| 14:52.30 | EmleyMoor | Is there a way to find lines in a dialplan that end in similar ways? |
| 14:53.21 | *** join/#asterisk CunningPike (n=arodgers@64.251.77.9) |
| 14:53.41 | *** join/#asterisk mihinomenest (i=Cgah431y@66.255.220.17) |
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| 14:54.50 | furet | Does someone know how to set up asterisk to make calls with a CISCO 12 SP+ phone ? |
| 14:55.24 | furet | I can make call from my cisco phone, but i cannot call the phone from a softphone |
| 14:57.53 | furet | (this phone uses skinny protocol) |
| 14:59.34 | [TK]D-Fender | furet: pastebin your call attempt at verbose 10 |
| 14:59.36 | [TK]D-Fender | ~pb |
| 14:59.37 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 14:59.38 | [TK]D-Fender | ^^^^^^^^^^^ |
| 14:59.47 | [TK]D-Fender | Katty, another one here for you, gt him! |
| 15:00.03 | tzanger | hmm |
| 15:00.57 | tzanger | if I have a TDM card (say a T1 card) there are two ways I could loop back. one way is that anything I receive from the cable I just loop back on the transmit side, and the other is anything I'm about to transmit, I loop back into my own receiver (internally) -- what are the actual names for these two types of loopback? |
| 15:01.25 | tzanger | I want to say the first is remote loopback and the second is local loopback but that's not quite right, becuase remote loopback is when you request the far end to loop back for you so you can test the line itself |
| 15:02.42 | *** join/#asterisk casix (n=casix@pbxedifici.adamvozip.es) |
| 15:02.44 | casix | hello |
| 15:03.06 | EmleyMoor | I'm wondering if I can convert any more of my dialplan into macros... but working that out will probably take me some time |
| 15:03.09 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 15:03.25 | [TK]D-Fender | EmleyMoor: My rates are very accessible ;) |
| 15:03.49 | casix | if I specified a timeout option for the Dial command when will start it to count? after the execute of dial or after reciving the ringing packet? |
| 15:04.35 | [TK]D-Fender | casix: should be from ringing. |
| 15:04.46 | casix | ok, thx :) |
| 15:04.48 | EmleyMoor | I just got rid of all my n+1 priorities and combined a few ExecIf and GotoIf lines together |
| 15:05.12 | EmleyMoor | (oh, and my "false only" GotoIfs) |
| 15:07.14 | [TK]D-Fender | Holy fuck .... http://www.ireport.com/docs/DOC-93628 |
| 15:13.12 | furet | http://rafb.net/p/Gh4Xb926.html (when i call ekiga from cisco phone) |
| 15:13.28 | *** join/#asterisk netsecur (n=netsecur@ippp-hmg147.bgnett.no) |
| 15:14.27 | furet | http://rafb.net/p/vDavTJ74.html (when i call a cisco phone from another cisco phone) |
| 15:16.00 | furet | A call from cisco phone to ekiga does not give me informations |
| 15:16.18 | netsecur | i have a problem: when i have two (or more) trunks from the same provider, Asterisk always treats it as if all calls are coming in on one of the two trunks, therefore again not allowing me to set up separate inbound routes for each trunk |
| 15:16.23 | furet | from ekiga to cisco (sorry) |
| 15:16.48 | furet | may be you need my skinny.conf ? |
| 15:17.23 | netsecur | http://www.aussievoip.com.au/wiki/How+to+get+the+DID+of+a+SIP+trunk suggests a change in extensions.conf and so on, however my system uses users.conf to configure trunks |
| 15:19.10 | netsecur | anyone who can suggest how i would go about solving this while still using users.conf for trunks? |
| 15:19.54 | *** join/#asterisk tobias (n=tobias@user-0c8hj2l.cable.mindspring.com) |
| 15:21.43 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:22.25 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 15:24.59 | *** join/#asterisk jon79 (n=manning_@97.66.98.181) |
| 15:25.45 | jon79 | ~thebook |
| 15:25.46 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 15:27.10 | c4t3l | ~dance |
| 15:27.15 | c4t3l | darnit! |
| 15:27.18 | Assid | err who was ist here who runs flowroute? |
| 15:29.48 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 15:31.15 | netsecur | jon79: was that aimed at me? |
| 15:31.41 | netsecur | oh never mind |
| 15:35.58 | *** join/#asterisk LiNeTuX (n=LiNeTuX@63-255-103-7.ip.mcleodusa.net) |
| 15:36.25 | LiNeTuX | Good morning from Astricon... |
| 15:36.56 | *** join/#asterisk chrisM8 (n=chrisM8@manila.tarsus.co.uk) |
| 15:37.00 | eric2 | I've peered 2 * servers and in sip.conf I have 'fromuser=myname' so the calls will flow back and forth |
| 15:37.01 | jaytee | [TK]D-Fender, man I love Polycoms, now I've got different ring tones depending on whether the call is from an external or an internal number. My boss is ecstatic. |
| 15:37.28 | jeev | jaytee, have you figured out how to make it light up and not ring? without volume touching ? |
| 15:37.33 | eric2 | when I set the caller id, I get number as showing 'myname' as its set as fromuser=myname |
| 15:37.36 | eric2 | how do I set the number? |
| 15:37.41 | jaytee | jeev, yep |
| 15:37.47 | jeev | how! |
| 15:37.51 | chrisM8 | hi, I got a strange problem. LAN with around 50+ SIP phones, all but two are getting WMI via SIP NOTIFY OK. Only two phones are not? Any idea what the problem might be? |
| 15:37.54 | jaytee | I read the manual |
| 15:37.56 | eric2 | setting it with CALLERID(num) isn't working |
| 15:38.04 | jeev | i read the manual too |
| 15:38.25 | jaytee | obviously not as throughly as you should have :-) |
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| 15:44.11 | ReDNeQ | morning/noon/afternoon |
| 15:46.38 | *** part/#asterisk elred (i=sauron@fucksheep.org) |
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| 15:54.02 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:54.17 | troubled | hey guys, I have an extension that calls my console (call over speakers/intercom), and I am wondering if there is a dial switch to allow the caller to use features.conf bindings to trigger actions or if I should just use some fancy bridge to console with a menu setup to trigger sounds |
| 15:56.22 | [TK]D-Fender | troubled: same as for any call. |
| 15:57.04 | *** join/#asterisk raz (n=raz@unaffiliated/raz) |
| 15:57.11 | raz | hmm. how can i adjust the volume when using format_mp3? |
| 15:57.20 | troubled | [TK]D-Fender: which is? |
| 15:57.36 | [TK]D-Fender | troubled: Go read up on features.conf on the WIKI. |
| 15:57.44 | [TK]D-Fender | raz: You don't |
| 15:57.58 | [TK]D-Fender | raz: playback is fixed |
| 15:58.03 | raz | ewww |
| 15:58.06 | [TK]D-Fender | raz: regardless of format. |
| 15:58.06 | troubled | [TK]D-Fender: ive read it a few times, but i just cant get it to trigger any of the features from the line im calling the console from |
| 15:58.25 | [TK]D-Fender | raz: it is up to your source recording to be properly normalized as well as your endpoints & interfaces |
| 15:58.36 | troubled | [TK]D-Fender: wW seems specific to automon which isnt what I want, unless it allows any feature defined to work |
| 15:59.11 | *** join/#asterisk therproject (n=mries@h-64-105-53-130.mclnva23.covad.net) |
| 16:00.55 | troubled | [TK]D-Fender: ideally, I want to call the console, it auto answers and then be able to speak over the speakers and at anytime just hit a DTMF and playback certain sounds in addition to my call over the speakers, but using a menu seems like it would block inside the Dial() the way I was thinking originally which is why I opted for using features.conf, if you follow me |
| 16:01.18 | adr3nalin3 | Guys I am having trouble with an analog card. It seems as though asterisk isn't receiving the disconnect signal from the telco when a caller hangs up. Is there a setting I can change maybe fix this issue? I have enabled call progress. |
| 16:02.25 | adr3nalin3 | I have also checked with the telco and forward disconnect is enabled on the telco side. |
| 16:02.42 | *** part/#asterisk chrisM8 (n=chrisM8@manila.tarsus.co.uk) |
| 16:05.14 | Kobaz | [Sep 25 12:04:47] WARNING[7470]: app_queue.c:3014 try_calling: The device state of this queue member, Local/2608@standard, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. |
| 16:05.23 | Kobaz | is that bad? |
| 16:12.04 | *** join/#asterisk Carlos_PHX (n=Carlos@63-255-123-105.ip.mcleodusa.net) |
| 16:14.45 | km- | How do you pass multiple arguments to an application in manager API? |
| 16:14.53 | km- | i.e., if you're doing Application: Festival |
| 16:15.04 | km- | do you have a single Data: line with ("arg1","arg2") |
| 16:15.09 | km- | or is it multiple Data: lines? |
| 16:15.46 | tzafrir_laptop | km-, in originate? |
| 16:15.48 | *** join/#asterisk jeev (n=email@unaffiliated/jeev) |
| 16:16.12 | km- | yep. |
| 16:16.30 | *** join/#asterisk Nasra (n=maxshipp@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 16:16.31 | km- | It'll be an originate that's terminating to the application Festival -- trying to pass multiple arguments to that app. |
| 16:16.39 | tzafrir_laptop | I'd expect that in 1.4 they would be seperated with '|' and in in 1.6 with ',' . |
| 16:16.46 | tzafrir_laptop | But jeev should know :-) |
| 16:17.40 | Qwell | jeev knows all |
| 16:17.50 | jeev | i wont tell |
| 16:18.21 | km- | I dont have much to offer you for the help except for a hearty pat on the virtual back and a thank you |
| 16:18.32 | jeev | you guys just have to check svn every day to see if it i put it in. |
| 16:18.53 | tzafrir_laptop | adr3nalin3, at which country are you? Is there any sort of disconnect supervision on that line? |
| 16:19.04 | *** join/#asterisk hfb (n=hfb@pool-96-247-116-5.lsanca.dsl-w.verizon.net) |
| 16:20.54 | tzafrir_laptop | adr3nalin3, I'm not sure if enabling call progress actually helps |
| 16:21.09 | adr3nalin3 | tzafrir_laptop: I am in the US. |
| 16:21.20 | *** join/#asterisk ManxPower (n=manxpowe@39.sub-70-221-226.myvzw.com) |
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| 16:21.38 | adr3nalin3 | tzafrir_laptop: Call progress was a recommendation by a digium tech. but I am not sure either. |
| 16:21.45 | *** part/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 16:22.08 | tzafrir_laptop | do you use fxsks signallling there? |
| 16:22.15 | *** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net) |
| 16:22.40 | jeev | km-, dont worry, your thanks is what makes me continue helping everyone throughout the asterisk world. |
| 16:23.24 | ManxPower | callprogress=yes is just an alias for randomlydisconnectmycalls=yes |
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| 16:24.06 | *** part/#asterisk pyite (n=pyite@63-255-103-7.ip.mcleodusa.net) |
| 16:24.08 | *** join/#asterisk pyite (n=pyite@63-255-103-7.ip.mcleodusa.net) |
| 16:24.53 | [TK]D-Fender | Kobaz: Only if you care. |
| 16:26.33 | Kobaz | heh |
| 16:26.56 | *** join/#asterisk miguel3239 (n=elguero@ns1.nashuacs.com) |
| 16:27.03 | tzafrir_laptop | busydetect=yes likewise. |
| 16:27.04 | Kobaz | [TK]D-Fender: we're trying to do a new install and there are complaints of calls getting lost |
| 16:27.45 | Kobaz | not sure where they are getting lost |
| 16:28.47 | tzafrir_laptop | someone from QA asked me about an aparant bug of our system that disconnects when pressing '1' on the phone too many times in a row quickly |
| 16:29.16 | tzafrir_laptop | "fixed" by disabling busydetect :-( |
| 16:29.48 | Kobaz | it's really weird.... mr outside calls in on a pri, the call goes to a queue, agent A picks up. agent A get's a call from agent B, outside goes on hold, agent A goes back to outside, and can't get the call back |
| 16:30.25 | Kobaz | [TK]D-Fender: when agent A goes to pick outside back up, asterisk is still showing that channel is going to music on hold |
| 16:30.27 | ManxPower | Kobaz: that problem is almost ALWAYS a phone bug/problem/issue |
| 16:30.31 | Kobaz | k |
| 16:30.47 | ManxPower | how, exactly, does the user put the first call on hold? |
| 16:30.52 | Nasra | hi, I have a question: I am using voip services with ATA/Router Linksys...how can I expand my or add another # (1-800) can I add another ATA using same ip? ... Iam new to all this...thanks... |
| 16:30.56 | Kobaz | hits the ringing line 2 |
| 16:31.01 | Kobaz | line 1 auto goes on hold |
| 16:31.32 | Kobaz | and i was getting that error from app_queue, i was wondering if it was related somehow |
| 16:31.34 | ManxPower | Kobaz: and what phone is it? |
| 16:31.46 | Kobaz | aastra |
| 16:31.59 | Qwell | [TK]D-Fender: |
| 16:32.10 | Kobaz | i wish polycom made phones with a bunch of programmable buttons |
| 16:32.11 | ManxPower | Nasra: You don't. Your server/provider does. How is this related to Asterisk? |
| 16:32.14 | Kobaz | [Sep 25 12:04:47] WARNING[7470]: app_queue.c:3014 try_calling: The device state of this queue member, Local/2608@standard, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. |
| 16:32.26 | Kobaz | that's the error from the queue, but i dont think it's causing the calls to die |
| 16:32.30 | ManxPower | Kobaz: did you check UPGRADE.txt? |
| 16:32.32 | Kobaz | yeah |
| 16:32.59 | Kobaz | it doesn't say anything about that, other than some stuff about agentringbacklogin being depricated |
| 16:33.23 | Kobaz | which i've moved to using AddQueueMember/RemoveQueueMember instead |
| 16:33.40 | Nasra | ManxPower : thanks alot .... |
| 16:35.05 | ManxPower | Kobaz: have your users tried putting the first call on hold first? |
| 16:35.10 | ManxPower | like pressing the HOLD button |
| 16:35.17 | Qwell | [TK]D-Fender: psst |
| 16:35.19 | Kobaz | ManxPower: not yet |
| 16:35.27 | Kobaz | ManxPower: i can't even replicate the issue here in the office |
| 16:36.57 | Kobaz | i'm gonna use SIP directly and test some stuff rather than using Local |
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| 17:00.39 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 17:00.39 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:01.13 | russellb | skype for asterisk w00t |
| 17:02.26 | *** join/#asterisk rvhi (n=chatzill@udp255518uds.hawaiiantel.net) |
| 17:04.11 | anonymouz666 | russellb: where? |
| 17:04.15 | anonymouz666 | how? |
| 17:04.17 | anonymouz666 | what happened? |
| 17:04.18 | russellb | just announced |
| 17:04.26 | russellb | at the keynote at astricon |
| 17:04.29 | russellb | :-D |
| 17:04.30 | ph0enix | awesome! |
| 17:04.33 | MikeJ | russellb: one client per client logged in? |
| 17:04.40 | russellb | no |
| 17:04.50 | anonymouz666 | I don't need to open the X server? |
| 17:04.50 | russellb | you can have as many users logged in as you want |
| 17:04.52 | MikeJ | they actually finally released the pres api? |
| 17:04.53 | [TK]D-Fender | ~skypeforasterisk |
| 17:04.54 | jbot | skypeforasterisk is, like, [~skypeforasterisk] is a Digium-made channel driver that allows Asterisk to connect to the Skype network using a Skype-supported connection method. Unlike all other solutions it does not require X11, kernel modules, or a Windows machine. See http://www.astricon.net/skype for beta details. |
| 17:04.56 | [TK]D-Fender | ~skype |
| 17:04.57 | jbot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, with most solutions being complex, complicated, and hack-ish . Digium's SkypeForAsterisk (see ~SkypeForAsterisk) is a new solution that is a cleaner non-dependent option. |
| 17:04.57 | russellb | and you can have as many calls per user as you want |
| 17:05.25 | MikeJ | I thought it was pay per user model? |
| 17:05.31 | russellb | nope. |
| 17:05.34 | [TK]D-Fender | ~skypeforasterisk |
| 17:05.35 | jbot | [~skypeforasterisk] is a Digium-made channel driver that allows Asterisk to connect to the Skype network using a Skype-supported connection method. Unlike all other solutions it does not require X11, kernel modules, or a Windows machine. See http://www.astricon.net/skype for beta details. |
| 17:05.41 | [TK]D-Fender | there we go |
| 17:05.42 | russellb | skype for asterisk is a pay per channel module. |
| 17:05.51 | implicit | hello |
| 17:06.00 | MikeJ | russellb: what is the distinction? |
| 17:06.04 | MikeJ | concurrent call? |
| 17:06.08 | russellb | right |
| 17:06.11 | MikeJ | cool |
| 17:06.27 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 17:06.27 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 17:06.31 | [TK]D-Fender | MikeJ: distinction : you can be 10 identities but limited to 5 calls. |
| 17:06.44 | *** join/#asterisk pyite (n=pyite@63-255-103-7.ip.mcleodusa.net) |
| 17:06.47 | MikeJ | [TK]D-Fender: yeah.. already got that |
| 17:06.55 | [TK]D-Fender | MikeJ: Same concepts as DID's VS B-Chans on PRI |
| 17:06.59 | anonymouz666 | russellb: open source? |
| 17:07.11 | MikeJ | [TK]D-Fender: thanks.. not an idiot.. already covered it |
| 17:07.44 | russellb | anonymouz666: part of it ... the channel driver itself is source available, not to be confused with open source. it's glue to the skype for asterisk library |
| 17:07.46 | [TK]D-Fender | MikeJ: Ok, was typing at the same time you were ack-ing |
| 17:07.46 | MikeJ | russellb: did skype actually publicly release that pres api now.. |
| 17:08.01 | russellb | MikeJ: no |
| 17:08.04 | MikeJ | they announced like 2 years ago |
| 17:08.16 | MikeJ | lame |
| 17:09.04 | russellb | but skype presence will be exposed to asterisk. |
| 17:10.53 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 17:10.53 | *** mode/#asterisk [+o file] by ChanServ |
| 17:10.56 | *** join/#asterisk rgsteele||work (n=rgsteele@75.147.74.137) |
| 17:11.16 | [TK]D-Fender | I've never actually used Skype before... it does have a DTMF interface, right? |
| 17:11.16 | russellb | err ... file |
| 17:11.21 | russellb | [TK]D-Fender: yes |
| 17:11.32 | [TK]D-Fender | russellb: Ok, so it is quite usable.... |
| 17:11.37 | russellb | indeed :) |
| 17:11.51 | russellb | I use skype a good amount with family |
| 17:11.58 | [TK]D-Fender | russellb: And general background question : do you need to auth people adding you as a contact? |
| 17:12.10 | russellb | [TK]D-Fender: the auth policy is configurable. |
| 17:12.15 | file | hola |
| 17:12.25 | russellb | you can have it automatically auth people, or a number of other settings. i don't remember them exactly |
| 17:12.28 | [TK]D-Fender | russellb: Ok, I'll simply have to get off my butt and test it out. |
| 17:12.37 | [TK]D-Fender | russellb: but thatnks for the quick confirm. |
| 17:12.46 | russellb | no problem |
| 17:12.49 | [TK]D-Fender | russellb: I'll likely implement a gateway for it. |
| 17:12.54 | russellb | yay |
| 17:12.56 | [TK]D-Fender | (for use here at work) |
| 17:12.58 | russellb | right |
| 17:13.03 | russellb | Hopefully lots of people will! :-D |
| 17:13.45 | [TK]D-Fender | russellb: Oh, and Skype does video as well, no? Can * pass-through or re-pack? |
| 17:13.59 | Qwell | russellb: ! |
| 17:14.14 | russellb | [TK]D-Fender: Skype does video and it will eventually be supported. It is not supported in the initial beta |
| 17:14.19 | rgsteele||work | Hey folks. I've got an Asterisk box which has been humming along great for awhile. But today, I found that zaptel wasn't working. Inbound calls get a busy signal, outbound calls get staticky silence. I can't restart zaptel (some of the modules are in use by other things on the system), but I was able to test that plugging a standard phone in to the POTS line allows me to make outbound... |
| 17:14.20 | rgsteele||work | ...calls. I can see in the asterisk logs that the number passed to zaptel for outbound calls is normal, so my only recourse at this point seems to be rebooting the box (as trying to shut down dependent services one at a time would probably take longer than just a reboot). Anybody have an idea of what else it could be besides the zaptel services just being hosed and needing a restart? |
| 17:14.36 | russellb | Qwell: w00t |
| 17:14.39 | Qwell | russellb: I see you made that change for me. <3 Did you do the second change too? |
| 17:14.43 | Qwell | s/do/see/ |
| 17:14.52 | russellb | Qwell: i don't know what the 2nd change was. pm me |
| 17:14.57 | Qwell | jabber? |
| 17:15.18 | russellb | jabber is being lame |
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| 17:38.43 | angryuser | i got some siemens sip phones, (C470IP 650IP 675IP) sometimes the called party hears voice but the calling person dont, it does not matter if it's internal/external call, pretty desperate here, any ideas why is it happening ? |
| 17:39.01 | angryuser | i am sure about ports |
| 17:39.11 | C4away | canreinvite=no globally or on each phone |
| 17:39.15 | angryuser | they are in range |
| 17:39.31 | angryuser | checking |
| 17:39.42 | *** part/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 17:40.34 | eric2 | Failed to authenticate on INVITE ??? |
| 17:41.05 | eric2 | my proxy server has all the sip accounts on it, my other server has the TDM card |
| 17:41.17 | eric2 | why do calls come in, but I cannot place them out? |
| 17:42.49 | angryuser | C4away: your answer was for me ? |
| 17:43.46 | *** join/#asterisk legis (i=estar@unaffiliated/legis) |
| 17:44.53 | legis | Is there some softphone that supports sending fax? |
| 17:45.07 | c4t3l | softphone? fax? |
| 17:45.30 | legis | I want to send a fax from my PC (SIP) to the pstn |
| 17:45.34 | jjshoe | legis that would be interesting |
| 17:46.08 | c4t3l | legis: you can use asterisk + iaxmodem + hylafax for that |
| 17:46.12 | angryuser | legis: iaxmodem ? for linux |
| 17:46.23 | jaytee | I want to send a fax from my toaster to my microwave |
| 17:46.40 | legis | jjshoe: easy 2.4GHz :P |
| 17:46.42 | c4t3l | no no... a fax from my toilet to my other toilet |
| 17:46.51 | legis | that was to jaytee i mean |
| 17:47.13 | *** join/#asterisk pirulo (n=pirulo@70.56.223.76) |
| 17:47.29 | legis | Ok, I'll check out iaxmodem |
| 17:48.02 | legis | how about fax_machine -> ATA/sip -> asterisk -> pstn |
| 17:48.14 | c4t3l | eww no! |
| 17:48.17 | legis | would asterisk support that? |
| 17:49.02 | c4t3l | legis: no use in circling the world to go accross the street. |
| 17:49.28 | merlinn | legis: it does technically work |
| 17:49.37 | merlinn | providing you're running G.711 end to end |
| 17:49.45 | merlinn | but it's flakey as shit |
| 17:49.53 | c4t3l | you put g729 in there and good luck buddy |
| 17:49.58 | legis | lol |
| 17:50.00 | merlinn | even if you dont |
| 17:50.10 | c4t3l | it aint gonna work... reliably |
| 17:50.12 | merlinn | it's pretty iffy |
| 17:50.25 | merlinn | we had a lot of ATA's ruunning fax machines in the field |
| 17:50.34 | merlinn | and each time we moved a version of asterisk |
| 17:50.38 | merlinn | the ATA's had to be reconfigured |
| 17:50.41 | c4t3l | merlinn: which field? |
| 17:50.49 | merlinn | field being with customers |
| 17:50.53 | merlinn | in production |
| 17:51.11 | c4t3l | merlinn: i know that. which geographical region |
| 17:51.15 | MikeJ | if you put g729 in there is no luck involved.. it simply will not work |
| 17:51.16 | merlinn | not like in the field belonging to farmer giles that he keeps pigs in behind the chicken shed |
| 17:51.16 | c4t3l | :) |
| 17:51.28 | merlinn | I'm in the UK |
| 17:51.29 | c4t3l | lol |
| 17:51.36 | legis | ah iaxmodem is a software modem :) |
| 17:51.49 | c4t3l | ahh, I went through the same issues here in the states |
| 17:52.04 | c4t3l | merlinn ^^^ |
| 17:52.25 | merlinn | we've just had to bite the bullet and install about 10 trillion PSTN lines |
| 17:52.29 | merlinn | for our customers |
| 17:52.36 | merlinn | and take the sting financially |
| 17:52.43 | c4t3l | merlinn: that is the best way to go |
| 17:52.52 | merlinn | as a result I now have about 10 billion ata's |
| 17:52.55 | merlinn | that nobody will buy on ebay |
| 17:53.09 | merlinn | I notice that every month or so someone puts a stockpile of about 50 used ATAs on ebay |
| 17:53.21 | ManxPower | There's a reason for that. |
| 17:53.26 | merlinn | I think it's all the voip companies out there doing what we did and tyring to get shot of their stock pile |
| 17:53.56 | ManxPower | The mistake was to use large numebrs of ATAs in the first place. |
| 17:54.06 | legis | MikeJ: heh yeah, no fax will fit in 8Kbit |
| 17:54.31 | merlinn | thanks ManxPower you've just recapped more or less what everyone just said in a manner that is totally useless |
| 17:54.34 | merlinn | but still somehow annoying |
| 17:54.43 | c4t3l | hehe |
| 17:55.09 | ManxPower | merlinn: it's one of my many talents |
| 17:55.14 | merlinn | perhaps you should become a school teacher - there's always space for self satisfied pricks in that market |
| 17:55.42 | c4t3l | wow ^^^ |
| 17:55.46 | ManxPower | That's a great idea! |
| 17:56.30 | ManxPower | There is always the /ignore option |
| 17:56.31 | *** join/#asterisk jplank (n=GBove@243.sub-75-209-215.myvzw.com) |
| 17:56.48 | c4t3l | Does prick mean the same thing in the UK as it does in the US?? |
| 17:56.59 | merlinn | yeah |
| 17:57.03 | merlinn | I'm actually american |
| 17:57.11 | c4t3l | ahh haa! |
| 17:57.17 | c4t3l | me too |
| 17:57.24 | merlinn | I just happen to be a UK resident currently |
| 17:57.28 | [TK]D-Fender | merlinn: It's OK... we accept you regardless ;) |
| 17:57.37 | merlinn | thanks buddy! |
| 18:01.07 | merlinn | has anyone got any experience using asterisk/freeswitch/whatever for mass termination |
| 18:01.34 | merlinn | essentially just intelligent routing of calls |
| 18:01.40 | merlinn | rather than any complicated processing |
| 18:01.58 | merlinn | skype are doing interconnects with sip companies now |
| 18:02.03 | c4t3l | stay away from skype!! |
| 18:02.11 | c4t3l | they are the devil |
| 18:02.19 | merlinn | so that joe bloggs incorporated can call skype users from their handsets |
| 18:02.20 | ManxPower | c4t3l: Even noobs know that. 8-) |
| 18:02.27 | c4t3l | :D |
| 18:02.39 | merlinn | but so that their users can be charged to call xyz company when its' really free |
| 18:04.47 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 18:06.24 | [TK]D-Fender | merlinn: These aren't the WMD's you're looking for... |
| 18:06.31 | merlinn | WMD? |
| 18:06.37 | *** join/#asterisk citywok (n=andrewp@65.249.42.130) |
| 18:06.42 | c4t3l | haha |
| 18:07.05 | merlinn | is that a joke becuse I'm americna |
| 18:07.06 | merlinn | lol |
| 18:07.31 | [TK]D-Fender | "mass termination" |
| 18:08.49 | merlinn | ah |
| 18:09.00 | merlinn | no I need to route like 10,000 simultaneous calls |
| 18:09.08 | merlinn | and I'm looking for the lowest cost solution |
| 18:09.19 | merlinn | because my current setup doesn't scale |
| 18:09.22 | merlinn | or won't scale to that numer |
| 18:10.30 | *** join/#asterisk Blackthorn (i=blacktho@76.77.160.10) |
| 18:11.33 | jaytee | every time I hear the name joe bloggs I think of Firefly "Trainjob" |
| 18:11.41 | Blackthorn | I have a number of sipura adapters connected to our * server and they do not ring like a normal phone. It does two quick rings then three rings. Do you know what this is ring is telling me? |
| 18:12.30 | citywok | 10,000 calls? are you a CO? |
| 18:13.12 | tzafrir_laptop | ManxPower, seems like the major announcement of every Astricon must be "Digium: Asterisk is not free software" |
| 18:13.38 | StephenF | How do you guys normally handle fax machines? Just keep them seperate from *? or use an ATA..., some other solution? |
| 18:13.49 | tzafrir_laptop | Major announcement of last Astricon was about proprietary software. Likewise of this time |
| 18:14.07 | merlinn | citywok: I don't even knoiw what that acronym means |
| 18:14.27 | merlinn | so I guess not :( |
| 18:14.29 | citywok | Central Office |
| 18:14.35 | merlinn | urm no |
| 18:14.43 | merlinn | we're an ISP I guess |
| 18:14.50 | citywok | thats a lot of freaking calls |
| 18:14.54 | merlinn | yeah |
| 18:14.56 | citywok | oh, okay yea that makes sense |
| 18:15.18 | russellb | tzafrir_laptop: come on ... clearly, we make all of asterisk open source that we can. with skype, it's not our choice. However, providing connectivity options to over 300 million skype users _IS_ a bit announcement |
| 18:15.18 | merlinn | it's not really telephony in the conventional sense |
| 18:15.50 | tzafrir_laptop | MAking it a a major announcement is Digium's choice |
| 18:16.15 | *** join/#asterisk swampwork (n=rew@64.238.252.218) |
| 18:16.30 | ph0enix | well i think its pretty cool. |
| 18:16.44 | ph0enix | no open source snobbery here. |
| 18:19.01 | Kobaz | http://pastebin.com/m4d26b7a9 |
| 18:19.36 | Kobaz | i'm having problems with a queue.... someone dials in, it goes through some cycles of the queue, and then the queue hangs up the caller |
| 18:20.46 | EmleyMoor | I understand from 1.6 Gosub will be preferred over Macro - does that mean Macro will likely be deprecated in 1.8? |
| 18:21.09 | Blackthorn | When someone calls my sipura ata's it does two quick rings and then three rings, know how to just get it to ring normally? |
| 18:21.44 | [TK]D-Fender | EmleyMoor: I wouldn't worry about that |
| 18:21.55 | EmleyMoor | Indeed, it's a long way off yet |
| 18:22.44 | AndyMillar | hmm, how much fun is it getting asterisk to fo faxes? |
| 18:26.50 | Kobaz | [TK]D-Fender: so i think i found my problem with random calls getting dropped |
| 18:27.02 | Kobaz | [TK]D-Fender: i did a pastebin... http://pastebin.com/m4d26b7a9 |
| 18:27.10 | Kobaz | [TK]D-Fender: :) |
| 18:28.38 | Kobaz | the queue will just decided to stop working after 3 cycles |
| 18:36.44 | EmleyMoor | AndyMillar: Depends on what you want to achieve - it can be "interesting" |
| 18:37.06 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 18:46.04 | *** join/#asterisk asterisker (i=asterisk@d54C4AE81.access.telenet.be) |
| 18:46.21 | asterisker | hi |
| 18:47.34 | asterisker | anyone seen the following yet? |
| 18:47.36 | asterisker | [Sep 25 20:46:58] NOTICE[30330]: chan_sip.c:15236 handle_request_register: Registration from '"Jiri"<sip:jiri@86.39.162.34>;transport=UDP' failed for '84.196.174.129' - Not a local domain |
| 18:47.43 | *** join/#asterisk leif[astricon] (n=Leif@63-255-123-206.ip.mcleodusa.net) |
| 18:47.49 | asterisker | I've got it working on 1.2 |
| 18:47.58 | asterisker | then upgraded to 1.4 for the gui |
| 18:48.09 | asterisker | but still getting this error |
| 18:48.27 | asterisker | I have a server on the internet and I want to register to it from home |
| 18:48.35 | asterisker | behind nat |
| 18:49.24 | ManxPower | asterisker: you have a [jiri] section in sip.conf? |
| 18:49.47 | asterisker | yes |
| 18:49.49 | asterisker | w8 |
| 18:50.31 | asterisker | [jiri] |
| 18:50.32 | asterisker | username=jiri |
| 18:50.32 | asterisker | type=friend |
| 18:50.32 | asterisker | secret=test |
| 18:50.32 | asterisker | regexten=1000 ; When they register, create extension 1234 |
| 18:50.32 | asterisker | callerid="xxxx" <1000> |
| 18:50.34 | asterisker | host=dynamic ; This device needs to register |
| 18:50.36 | asterisker | nat=yes ; X-Lite is behind a NAT router |
| 18:50.38 | asterisker | ;canreinvite=no ; Typically set to NO if behind NAT |
| 18:50.40 | asterisker | disallow=all |
| 18:50.42 | asterisker | allow=gsm ; GSM consumes far less bandwidth than ulaw |
| 18:50.44 | asterisker | allow=ulaw |
| 18:50.46 | asterisker | allow=alaw |
| 18:50.48 | asterisker | qualify=500 |
| 18:50.50 | asterisker | callerid=1000 |
| 18:50.52 | asterisker | context=tutorial |
| 18:50.54 | *** kick/#asterisk [asterisker!i=north@pdpc/sponsor/digium/Qwell] by Qwell (pastebin) |
| 18:51.05 | *** join/#asterisk asterisker (i=asterisk@d54C4AE81.access.telenet.be) |
| 18:51.06 | Qwell | ~pastebin |
| 18:51.09 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:51.19 | asterisker | oops |
| 18:51.21 | asterisker | sorry |
| 18:53.31 | tristanbob_ | <PROTECTED> |
| 18:53.39 | tristanbob_ | what does that error indicate? |
| 18:54.08 | Qwell | that the call was rejected |
| 18:54.13 | tristanbob_ | thanks Qwell ! |
| 18:54.16 | Qwell | any time |
| 18:54.23 | tristanbob_ | No Authority Found? |
| 18:54.24 | Qwell | sorry, I'm bitter :p |
| 18:54.36 | Qwell | umm, lemme see |
| 18:54.43 | tristanbob_ | I'm trying to setup an IAX trunk |
| 18:55.14 | Qwell | it's returning the cause code AST_CAUSE_FACILITY_NOT_SUBSCRIBED |
| 18:55.17 | Qwell | so...are you registered? |
| 18:55.41 | Qwell | (that's a semi-educated guess) |
| 18:56.33 | jeev | exten => 250,1,SIPAddHeader(Alert-Info: Visual) |
| 18:56.33 | jeev | exten => 250,2,SetVar(ALERT_INFO="Visual") |
| 18:56.38 | jeev | setvar is gone, right ? |
| 18:58.12 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 18:58.12 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-rc6, 1.4.22-rc5 (2008/09/09), *-Addons 1.6.0-rc1 (2008/09/03), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0-rc4+2.0.0-rc2 (2008/09/08), Zaptel 1.4.12.1 (2008/09/09), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev. AstriCon! #astricon |
| 18:58.46 | moy | Qwell: hey, yeah, it was good I think |
| 18:59.01 | Qwell | moy: great |
| 18:59.51 | Kobaz | you wouldn't think a queue would just randomly give up |
| 19:00.43 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 19:01.30 | asterisker | for those interested in the "not a local domain issue": it is solved. |
| 19:01.38 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 19:02.35 | asterisker | for some reason when you try to put "domain/localdomain = range/mask" and he doesn't want to accept it, he doesn't want to accept registers |
| 19:02.42 | tristanbob_ | Qwell, how do you debug iax in 1.4? |
| 19:02.51 | tristanbob_ | iax2 debug |
| 19:03.52 | ManxPower | asterisker: why not just leave those options off? |
| 19:04.29 | jeev | jaytee was helping me with "answer" where the lamp only blinks and there isn't sound.. i did this. |
| 19:04.35 | asterisker | I started fidling with them after the upgrade to 1.4 (from 1.2) |
| 19:04.53 | asterisker | register failed to work (because default config change?) |
| 19:05.10 | jeev | http://aosdada.pastebin.com/d15387a30 |
| 19:05.13 | jeev | anyone know what the problem be |
| 19:05.30 | StephenF | Is there a document or site somewhere that goes through configuring all the basic features of a normal PBX in *? |
| 19:05.36 | asterisker | as the message was "no local domain" and i was not on the local lan I thought that I needed to specify the range |
| 19:05.49 | asterisker | as described on the parameters |
| 19:05.50 | asterisker | ... |
| 19:06.12 | asterisker | Can anyone tell me when I should use domain ? |
| 19:07.07 | asterisker | i used ip address now but i would like to use domain names. |
| 19:07.39 | jaytee | jeev, I just modified mine to add the Visual and tested it and it works fine. |
| 19:09.37 | ManxPower | asterisker: everything you should need to know is in UPGRADE.txt and UPGRADE-1.2.txt in the 1.4 source tree |
| 19:10.22 | asterisker | OK thanks, I will look at it. sorry didn't have to much time to spend |
| 19:10.27 | jeev | crap |
| 19:13.54 | angryuser | hm, what is 'Activate annexe B for g729' is ? |
| 19:13.57 | *** join/#asterisk wiscados (n=mint@81.25.184.155) |
| 19:14.00 | *** join/#asterisk bpgoldsb (n=bpgoldsb@spatialdata2-gru-gw.customer.gru.net) |
| 19:14.00 | *** join/#asterisk BobPierce (n=BobPierc@216.36.132.162) |
| 19:14.09 | angryuser | annexe=option ? |
| 19:14.29 | angryuser | or some kind of signalling |
| 19:14.42 | bpgoldsb | Is loading every available module that you don't tell Asterisk to not load the normal behavior? |
| 19:15.06 | Qwell | bpgoldsb: yes |
| 19:15.17 | Qwell | see the autoload=yes in your modules.conf? |
| 19:15.20 | angryuser | ah it is g729b :) |
| 19:15.26 | bpgoldsb | Yes, I understand thats the default config |
| 19:15.28 | *** join/#asterisk SpeedDragon (n=SpeedDra@sm4-84-90-136-254.netvisao.pt) |
| 19:15.34 | bpgoldsb | It just seems... Error prone/bad to me |
| 19:15.42 | bpgoldsb | I figured I'd ask the people who know |
| 19:15.56 | Qwell | bpgoldsb: then fix your config to not do that |
| 19:16.52 | bpgoldsb | Qwell: I was asking more for a general census from people who know more about Asterisk than I do if thats acceptable, or if people turn off autoload and specifically load modules for better stability/preformance. |
| 19:20.56 | jaytee | bpgoldsb, you strike me as a person who might find this of interest: http://www.voip-info.org/wiki/view/Asterisk+Slimming |
| 19:20.56 | [TK]D-Fender | bpgoldsb: In general we only noload channel drivers and interface that might be a security issue. |
| 19:22.45 | *** join/#asterisk af_ (n=getsmart@88-149-241-240.dynamic.ngi.it) |
| 19:23.15 | bpgoldsb | jaytee and [TK]D-Fender: Thanks. |
| 19:23.44 | jaytee | I need a nap |
| 19:24.22 | jeev | :< |
| 19:24.26 | jeev | jaytee, we need to fix this! |
| 19:24.28 | jeev | yes, we!!! |
| 19:25.24 | jaytee | no, you need to grow a brain and fix it yourself. If I can get mine working then you should be able to get yours working too or it means I'm better than you are. Me! a low-life faggoty rollerblader!!! Neener neener!!! |
| 19:25.43 | jeev | hahah |
| 19:25.52 | jeev | you're lucky it worked |
| 19:25.56 | jeev | mine is giving me a hard time! |
| 19:26.13 | jeev | jaytee, i bet if we were at astricon and you could, you'd throw your rollerblades at me |
| 19:26.24 | jaytee | that's cuz your local-settings.cfg file is butt ugly compared with the grace and style of my local-settings.cfg |
| 19:26.28 | jeev | lol |
| 19:26.30 | jeev | bastard |
| 19:27.10 | jaytee | jeev, your problem or error is what we in the industry have long termed a PEBKAC |
| 19:28.19 | *** join/#asterisk funxion (n=x@63.214.236.169) |
| 19:28.33 | [TK]D-Fender | jaytee: Or another eye dee ten tee error;) |
| 19:28.51 | funxion | I just installed* 1.4 from branches and cant get asterisk to start im getting "load_module: Unable to create H323 listener." |
| 19:29.05 | funxion | lol |
| 19:29.13 | jaytee | [TK]D-Fender, yup and the usual RUTOK procedure will fix it most often I've found |
| 19:30.06 | BrianR___ | grr... pasting into a web browser textbox produces neither an onchange nor an onkeyup event :( |
| 19:31.48 | Qwell | BrianR___: real browsers do |
| 19:32.12 | BrianR___ | Qwell: D'oh.. Entered that in the wrong window... |
| 19:32.30 | BrianR___ | But pasting with the mouse in IE or firefox seems to not produce the desired event until the box loses focus... |
| 19:32.37 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 19:32.38 | *** join/#asterisk Talirk81 (i=434e2716@gateway/web/ajax/mibbit.com/x-7cc4c65678ba0f3b) |
| 19:33.08 | jeev | lol |
| 19:33.08 | funxion | [TK]D-Fender do you knwo what this "load_module: Unable to create H323 listener." could be? |
| 19:33.12 | jeev | you guys are assholes |
| 19:33.19 | jeev | but i still love you jaytee, in a very non-gay way |
| 19:33.19 | Talirk81 | <PROTECTED> |
| 19:33.41 | Talirk81 | look=loop |
| 19:33.57 | jaytee | jeev, and I still have you picture on my vanity mirror right next to Bobby Sherman's |
| 19:34.36 | [TK]D-Fender | Talirk81: Yes |
| 19:34.58 | Talirk81 | can you point me to where or explain how i would do it |
| 19:35.22 | Talirk81 | I know i could use a get to get but it looks like it stores it to a varible so im not sure how to loop on that |
| 19:40.18 | jeev | lol i dnuno who bobby sherman is, i think it's time for your nap |
| 19:40.40 | *** join/#asterisk Greg25c (n=chatzill@72.20.130.205) |
| 19:41.10 | Greg25c | We have an issue where users are dialing a conference room with an IAXy and then redialing - so Asterisk think the same user/extension is in the conference twice and the audio goes to heck for every one in the conference. Looking for away to configure the conference room such that an extension can only be present once in a conference room. |
| 19:41.31 | [TK]D-Fender | Talirk81: "core show application mysql". Its all in there. |
| 19:47.29 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 19:48.42 | infinity1 | if using users.conf, do i need to setup sip.conf? |
| 19:48.49 | infinity1 | for the phones? |
| 19:51.26 | tzafrir_laptop | funxion, port already in use? |
| 19:51.37 | tzafrir_laptop | listen() failed |
| 19:52.39 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 19:53.01 | Katty | [TK]D-Fender: ! |
| 19:53.05 | Katty | [TK]D-Fender: i have picked a pup |
| 19:53.11 | Katty | [TK]D-Fender: i get him sunday |
| 19:53.14 | Katty | [TK]D-Fender: would you like to see? |
| 19:53.16 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 19:53.18 | [TK]D-Fender | Katty: w00t |
| 19:53.33 | [TK]D-Fender | Katty: Told you... when its yours, in hand :) |
| 19:53.38 | Katty | [TK]D-Fender: it is mine. |
| 19:53.48 | [TK]D-Fender | Katty: Half way there! |
| 19:53.49 | Kobaz | hmm |
| 19:54.06 | Katty | well would anyone else like to see my new bundle of fluff |
| 19:54.09 | Kobaz | so i've narrowed down my queue problem to be related to the polycom in some way |
| 19:54.39 | Kobaz | aastra calls into a queue, polycom calls into a queue... after ring cycles in the queue, the polycom will get hung up |
| 19:54.43 | Kobaz | the aastra will keep ringing |
| 19:54.49 | Kobaz | [Sep 25 15:52:18] WARNING[9674]: channel.c:2557 ast_prod: Prodding channel 'SIP/2608-b7d075e8' failed |
| 19:55.00 | Kobaz | that's the polycom... and then asterisk hangs it up |
| 19:55.30 | Kobaz | er, after 3 ring cycles, i meant.... the polycom will get hung up |
| 19:56.04 | Kobaz | asterisk just decides to give up on the polycom |
| 19:56.43 | Kobaz | i'll have a sip debug for everyone's enjoyment in a sec |
| 19:57.40 | c4t3l | Katty: sure lets see the pup |
| 19:58.28 | Talirk81 | is there a substr type function inside asterisk so i could split the incoming caller id of a caller in area/exchange/branch varibles? |
| 20:00.53 | Kobaz | [TK]D-Fender: any ideas... our asterisk zen master? |
| 20:00.54 | *** join/#asterisk Daviey (n=Daviey@ubuntu/member/daviey) |
| 20:01.22 | Daviey | geez, never thought i'd see an offical Skype channel driver! |
| 20:01.31 | [TK]D-Fender | Kobaz: From what little you'v shown, no. |
| 20:01.59 | [TK]D-Fender | Talirk81: channelvariables.txt <- check it out in the doc folder |
| 20:07.17 | Kobaz | [TK]D-Fender: http://pastebin.ca/1210855 |
| 20:07.37 | Kobaz | [TK]D-Fender: the channel.c:2557 ast_prod: Prodding channel 'SIP/2608-b7d05e10' failed is where the phone gets hung up |
| 20:08.35 | Katty | c4t3l: ksec |
| 20:09.30 | *** join/#asterisk kalib (n=kalib@201008225158.user.veloxzone.com.br) |
| 20:09.41 | Katty | c4t3l: http://i36.tinypic.com/29apnb7.jpg |
| 20:10.25 | Kobaz | this is just so strange |
| 20:10.27 | c4t3l | what a qute puppy |
| 20:10.29 | *** part/#asterisk kalib (n=kalib@201008225158.user.veloxzone.com.br) |
| 20:11.13 | c4t3l | Katty: whats his name? |
| 20:11.15 | Kobaz | http://pastebin.ca/1210856 |
| 20:11.22 | Kobaz | and that's my queues.conf by the way |
| 20:11.43 | Kobaz | i dont see anything obvious that would cause the queue to just end after 3 ring periods |
| 20:12.11 | tzanger | c4t3l: askem |
| 20:12.55 | c4t3l | the puppy's name is askem? What does that mean? |
| 20:13.11 | Katty | c4t3l: Kaiser Riddick der Kleine Hobbit mit Waggytail |
| 20:13.19 | Katty | c4t3l: Riddick |
| 20:13.25 | c4t3l | cool |
| 20:13.35 | *** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) |
| 20:13.36 | Katty | Possibly Riddick mit Waggytail |
| 20:13.47 | c4t3l | lol |
| 20:14.35 | *** join/#asterisk zydoon (n=zydoon@41.225.153.114) |
| 20:14.41 | jaytee | awwww, I want a puppy too!!! |
| 20:14.49 | *** part/#asterisk zydoon (n=zydoon@41.225.153.114) |
| 20:15.48 | *** join/#asterisk maxxim (n=maxxxim@93-96-96-106.zone4.bethere.co.uk) |
| 20:16.02 | *** join/#asterisk Linuturk (n=Linuturk@fluxbuntu/developer/Linuturk) |
| 20:17.13 | maxxim | Hi. I have such scheme: gnugk->(h323)->asterisk->(sip)->addpac(gateway). When i make a call from gnugk clients, the gateway is trying to comunicate with gnugk directly, bypassing the asterisk. Call failed. How can i solve it? |
| 20:17.40 | *** part/#asterisk l2cache (n=l2cache@adsl-75-21-128-203.dsl.rcfril.sbcglobal.net) |
| 20:17.56 | *** join/#asterisk leif[astricon] (n=Leif@63-255-123-206.ip.mcleodusa.net) |
| 20:18.57 | *** join/#asterisk TheBonsai (n=bonsai@unaffiliated/thebonsai) |
| 20:19.42 | Katty | c4t3l: i gotta go get em a lil sweater :> |
| 20:20.10 | StephenF | Anything out there like this: http://icall.com/iphone/ |
| 20:20.15 | StephenF | but that is actually released? |
| 20:20.25 | TheBonsai | hi. is there a way to limit the codecs used (SIP) when a specific extension is dialled? e.g. have an extension with a conference behind, and influence the SIP setup to allow/disallow codecs xy? |
| 20:21.44 | kaldemar | TheBonsai: try setting variable SIP_CODEC in your dialplan. |
| 20:22.22 | kaldemar | maxxim: is it sending RTP to gnugk or what? be more specific. |
| 20:22.23 | *** join/#asterisk Dr-Linux|home (n=Nothing@221.132.117.17) |
| 20:22.40 | *** join/#asterisk shido6 (n=shido6@209.114.208.111) |
| 20:23.30 | Kobaz | [TK]D-Fender: strange isn't it? |
| 20:23.40 | TheBonsai | kaldemar: that doesn't work for some reason. i still send a completely different codec with my client. also i don't want to limit it to one codec, if possible. |
| 20:25.50 | maxxim | kaldemar> yes, RTP traffic. here is the asterisk log: http://rafb.net/p/BIZ6pZ20.html |
| 20:26.01 | maxxim | kaldemar> thanks for your time |
| 20:29.23 | kaldemar | is the RTP from the gnugk coming to asterisk? |
| 20:31.07 | maxxim | i can call from gnu to asterisk, withou problem. The issue is when i'm calling from gnugk to gateway that i connected to asterisk.... so, i can't dial to (let day 9999999 number) from gnugk to pstn (via asterisk and gateway) |
| 20:31.37 | maxxim | so, the connection between gnugk and asterisk is fine. i can hear 'backgroud' music |
| 20:31.57 | *** join/#asterisk iotashan (n=shan@adsl-71-150-254-145.dsl.mdsnwi.sbcglobal.net) |
| 20:32.32 | maxxim | i've created a sip peer for gateway. and i'm trying to put a call from asterisk using exten => s,1,Dial(SIP/99999999999@100) |
| 20:32.48 | maxxim | [100] is the name of gateway peer |
| 20:32.51 | kaldemar | have you tried setting an answer to an incoming call from gnugk before sending the call to the gateway? |
| 20:33.20 | maxxim | pacphone -> gnugk->asterisk->gateway(addpac) |
| 20:33.39 | maxxim | connection between pacphone->gnugk->asterisk works perfect |
| 20:34.09 | maxxim | the problem is that i can't put a call on gateway |
| 20:34.31 | maxxim | using 'trafshow' util, i've noticed, that gateway is sending some pakets directly to gnugk |
| 20:34.41 | maxxim | I think it shouldnt |
| 20:35.00 | maxxim | may be you can give me some documentation how to properlu configure asterisk for that... |
| 20:35.23 | kaldemar | it definitely shouldn't. try to aswer the call in asterisk to terminate the leg before making a new leg to the gateway. |
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| 20:37.04 | *** mode/#asterisk [+o russellb] by ChanServ |
| 20:37.07 | maxxim | kaldemar> how can I answer and after that to put the call to gateway? can you tell me please? |
| 20:37.27 | maxxim | just to add the Aswer step befor the dial one? |
| 20:37.38 | kaldemar | exten => s,1,Answer - exten => s,2,Dial... |
| 20:37.46 | kaldemar | yes |
| 20:37.57 | maxxim | thanks, let me try :P |
| 20:38.52 | *** join/#asterisk Havokmon (n=User@fw.vfemail.net) |
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| 20:42.53 | Talirk81 | I know you guys have helped alot with a substr type funciton and mysql looping, but can you do the same if you wanted to fire a php script with exec() or system() calls |
| 20:43.00 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:43.15 | maxxim | kaldemar> didn't help :( look please at the log http://rafb.net/p/lDWBTI31.html |
| 20:44.27 | maxxim | kaldemar> sniffing with trafshow, i still see that gateway is trying to send packets to RTP ports |
| 20:44.34 | maxxim | directly to gnugk |
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| 20:48.44 | kaldemar | maxxim: make a call with sip debugging enabled (sip set debug) and paste it. |
| 20:49.12 | maxxim | *CLI> sip set debug |
| 20:49.13 | maxxim | No such command 'sip set' (type 'help' for help) |
| 20:49.31 | maxxim | *CLI> sip debug |
| 20:49.32 | maxxim | ip peer |
| 20:49.37 | maxxim | wich one? ip of perr ? |
| 20:49.41 | kaldemar | you're running a pre 1.4 version of asterisk... |
| 20:49.54 | kaldemar | sip debug would be the older command. |
| 20:50.12 | maxxim | kaldemar> wich version do you recommend? |
| 20:50.41 | kaldemar | the newest in 1.4 |
| 20:51.05 | maxxim | *CLI> show version |
| 20:51.06 | maxxim | Asterisk 1.2.27 built |
| 20:51.26 | kaldemar | 1.4.21.2 is the current stable release. |
| 20:51.27 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:52.15 | maxxim | kaldemar> so you recommend me to put 1.4.21.2. instead of 1.2.27 ? or can i run debug for 1.2.27 now? or it is better to chang it to 1.4.x ? |
| 20:52.22 | maxxim | sorry fow such much questions |
| 20:52.57 | kaldemar | by all means, show the debug for 1.2, but there's always a chance that something in 1.2 is causing that. |
| 20:53.10 | leif[astricon] | 1.2 and 1.4 are entirely separate animals |
| 20:53.33 | maxxim | i don't mind with one to install, just to be a good one :) |
| 20:53.48 | maxxim | k, i'll play around to change it to 1.4 |
| 20:53.52 | maxxim | thanks for advices |
| 20:54.45 | kaldemar | soo, looks like 1.6 isn't going to be released in astricon.. or have i completely missed something? |
| 20:55.38 | *** join/#asterisk xenonex (n=xenonex@89.218.233.88) |
| 20:55.56 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 20:56.01 | russellb | kaldemar: probably not. still have a blocking issue in DAHDI ... |
| 20:56.11 | russellb | btw, skype for asterisk on digg ... http://bit.ly/1qldTZ |
| 20:56.57 | kaldemar | had to rub my eyes a few times reading that headline. :) |
| 20:58.31 | C4away | that won't last long |
| 20:58.34 | *** join/#asterisk implicit- (n=bayan@unaffiliated/implicit) |
| 20:58.40 | C4away | I'm sure someone at skype reads dig |
| 20:58.49 | russellb | why would it not last long? |
| 20:59.41 | russellb | C4away: it's not fake ... |
| 21:00.27 | kaldemar | is that planned to be an open source channel type or something else? |
| 21:00.54 | russellb | Only the channel driver part will be source available (not necessarily a traditional open source license) |
| 21:01.05 | russellb | the library that does all the skype magic will not be open source. |
| 21:01.14 | kaldemar | figures. |
| 21:01.54 | russellb | but it's a real skype connector, not some insane hack that doesn't really work ... |
| 21:01.55 | kaldemar | as much as i don't like skype, that sounds neat anyway. |
| 21:02.49 | Havokmon | Ok I'm stumped. I have a TDM400P. 1 fxs, 3 fxo. Not a month ago all 4 ports worked. Now I get no dialtone on my fxs port. Calls still come in on the fxo's. if I dial the fxs extension from a sip phone, it does not ring, but if I pick up the phone a connection is made. I've rerun genzaptelconf, and ztcfg -vv looks fine. Upgraded to latest zaptel drivers as well. Restarted box. Bad hardware maybe? |
| 21:03.04 | Havokmon | It seems like it's not plugged into power, but I just reseated everything.. |
| 21:03.12 | maxxim | kaldemar> done. http://rafb.net/p/d8bvdo60.html |
| 21:03.23 | *** join/#asterisk Pagautas (n=bigman@ns.voip.ktu.lt) |
| 21:04.08 | russellb | Havokmon: i would try support@digium.com |
| 21:04.47 | Havokmon | k - thanks |
| 21:05.13 | *** join/#asterisk seanmh (i=HydraIRC@216.31.101.83) |
| 21:05.24 | c4t3l | later all! |
| 21:07.42 | kaldemar | maxxim: nice. it's doing a reinvite. put canreinvite=no in your sip.conf. |
| 21:08.27 | maxxim | k, let me try :) |
| 21:08.44 | kaldemar | didn't think it would try reinvites with different channel types. |
| 21:09.13 | maxxim | should I add this parameter under the [100] configuration, or globally? |
| 21:10.28 | kaldemar | either way should do it. |
| 21:10.46 | kaldemar | if setting it on the peer only doesn't solve it, try it globally. |
| 21:12.08 | *** join/#asterisk mahlon (i=mahlon@martini.nu) |
| 21:13.05 | rwaite | stupid freaking echo |
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| 21:13.25 | maxxim | kaldemar> didn't help. new log: http://rafb.net/p/xqkX3k34.html WHat i've noticed: i can hear for a 0.5 second the tone into the pstn line... |
| 21:13.40 | maxxim | the same behavior was prior too... |
| 21:14.02 | *** join/#asterisk rasterix (n=IceChat7@80.177.176.254) |
| 21:15.08 | rasterix | if i find errors in the core show application help files where should i report them? |
| 21:15.45 | [TK]D-Fender | rasterix: Mantis, just like any other bug |
| 21:15.53 | [TK]D-Fender | rasterix: What did you find? |
| 21:16.05 | rasterix | im only on A |
| 21:16.08 | maxxim | kaldemar> now, i see that gateway is not trying to connect to gnugk... but there is no sound at all (just first 0.5 seconds) |
| 21:16.13 | kaldemar | maxxim: seems like you have yourself another issue now. the canreinvite prevents the gateway sending audio to gnugk, but thats something else. |
| 21:16.17 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 21:16.34 | rasterix | not too much so far... the help on AppendCDRUserField is a complete mess... but then its deprecated i guess |
| 21:16.39 | maxxim | kaldemar> can you find the problem? what logs/test can i do? |
| 21:17.11 | rwaite | skype for asterisk eh |
| 21:17.34 | kaldemar | i've never used ooh323 so i'm kinda lost with that one. hopefully someone else can help you. |
| 21:17.43 | rwaite | ugh you have to pay for it? |
| 21:18.08 | maxxim | kaldemar> it is related to translation between ooh323 and sip ? |
| 21:19.06 | kaldemar | probably. |
| 21:19.28 | *** join/#asterisk jplank (n=GBove@242.sub-75-209-159.myvzw.com) |
| 21:19.31 | jplank | skype for asterisk |
| 21:20.03 | maxxim | kaldemar> i'll try to connect to astersik via Xlite, and to place a call to gateway |
| 21:20.14 | maxxim | avoiding the gnugk and h323 |
| 21:20.26 | maxxim | kaldemar> thanks for you help!!! |
| 21:24.32 | EmleyMoor | jplank: What of it? |
| 21:25.19 | tzafrir_laptop | yet another non-free product by Digium |
| 21:26.54 | russellb | tzafrir_laptop: would you like some cheese with that whine? |
| 21:26.58 | russellb | :-p |
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| 21:33.51 | Talirk81 | With the mysql command is there a rowcount() type fuction to see the total number of returned rows before entering into a fetchrow loop? |
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| 21:41.02 | rasterix | found an error now Fender and im still on A |
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| 21:42.28 | rasterix | im going to catalogue them all and send them but I didnt really understand when you said who to report them to? |
| 21:45.24 | rasterix | core show application agi > syntax command|args > which must be wrong since args is optional? |
| 21:49.47 | [TK]D-Fender | rasterix: indeed, should be : [E|Dead]AGI(command[,args]) |
| 21:50.04 | rasterix | yup... like in the book |
| 21:51.21 | rasterix | and even that seems wrong to me AGI, EAGI and DeadAGI are separate apps |
| 21:51.59 | rasterix | also it seems weird that all the help files show | as the delimeter when it is about to be deprecated |
| 21:52.07 | *** join/#asterisk xenonex (n=xenonex@89.218.233.88) |
| 21:52.56 | [TK]D-Fender | rasterix: Actually as each other app has its own instructions [E|Dead] is also inappropriate |
| 21:53.17 | [TK]D-Fender | rasterix: "about to be" does not count. |
| 21:53.33 | [TK]D-Fender | rasterix: it is deprecated in 1 release, removed in another. |
| 21:54.26 | rasterix | i understand that... but , is now the preferred choice for delimiter and yet all newcomers to asterisk that use the help are being instructed to use a | = Madness in my opinion |
| 21:56.31 | rasterix | surely all it is doing is encouraging users to use a delimiter which wont work come 1.6 |
| 22:00.26 | *** part/#asterisk therproject (n=mries@h-64-105-53-130.mclnva23.covad.net) |
| 22:00.48 | Blackvel | any way to end/kill active sip channel? |
| 22:01.22 | Kobaz | soft hangup |
| 22:03.52 | Blackvel | ah works now |
| 22:04.45 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 22:04.45 | *** mode/#asterisk [+o russellb] by ChanServ |
| 22:07.11 | [TK]D-Fender | rasterix: as the next releases "upgrade.txt" will tell them the differences |
| 22:07.38 | maxxim | how can i place a call to an unregistered sip user? |
| 22:07.42 | [TK]D-Fender | rasterix: just because 1.6 is nearing release, 1.4 should not be modified in "psychic mode" |
| 22:07.52 | [TK]D-Fender | maxxim: Just dial it. |
| 22:08.17 | rasterix | so "," isnt already the preferred choice of delimiter and the one you see in most dial plans? |
| 22:08.23 | rasterix | my bad |
| 22:08.47 | [TK]D-Fender | rasterix: you mean MY preference? |
| 22:09.00 | [TK]D-Fender | rasterix: thats the heart of it you know. Preference. |
| 22:09.23 | maxxim | i'm just confused how asterisk works :) " Call from 'max' to extension '500' rejected because extension not found." |
| 22:09.48 | [TK]D-Fender | rasterix: and there are those who do : exten => exten,priority,app,args instead of exten => exten,priority,app(args) |
| 22:10.10 | [TK]D-Fender | maxxim: then clearly you do not have an exten to match in the proper place |
| 22:11.10 | rasterix | thats irrelevant fender... the fact is "," works now as a delimiter in 1.4 and if people were told to use it there would be less dial plans that needed this corrected come 1.6 |
| 22:11.12 | Blackvel | g'night |
| 22:12.14 | rasterix | there is no harm in encouraging people to start using the correct syntax for 1.6 especially if it works in 1.4 |
| 22:13.06 | maxxim | [TK]D-Fender> explain me please the basic steps. i have an sip user, registered to astersik. i want to call to an extension in order to (let say) play a music. Where should I add this extension? Under what context? |
| 22:13.21 | [TK]D-Fender | rasterix: And should your 2005 Ford Focus maintenance guide have instructions for the 2006 model year in it? Like "This isn't important for YOUR car but XYZ!!!) |
| 22:13.45 | [TK]D-Fender | rasterix: Just because they made the 2006 instruction book while a few 2005's were still in stock.. |
| 22:14.05 | *** join/#asterisk galeras (n=galeras@190.26.185.126) |
| 22:14.06 | rasterix | when you get you get your 2005 Ford Focus serviced does it turn into a 2006 model? |
| 22:14.09 | rasterix | no it doesnt |
| 22:14.23 | *** join/#asterisk AlexTO (n=alex@75.149.245.109) |
| 22:14.33 | [TK]D-Fender | rasterix: Provide instructions for what is, not what might be. 1.6 is not hit a solid release. That may be taken as "can change at any time) |
| 22:14.48 | rasterix | forget 1.6 |
| 22:14.52 | [TK]D-Fender | rasterix: And when does my * 1.4 install magically turn into 1.6? |
| 22:14.54 | rasterix | "," works in 1.4 |
| 22:15.11 | [TK]D-Fender | rasterix: Yes, it does. They both do. They both did since LONG before 1.4 |
| 22:15.33 | rasterix | so give me ONE reason why it is preferable to tell people to use | rather than , |
| 22:16.04 | [TK]D-Fender | rasterix: Since they were interchangeable, there is not point to preferring EITHER to the other. |
| 22:16.46 | rasterix | good grief there is a point to preferring , since it is the preferred syntax as demonstrated by the fact it will be the ONLY syntax in 1.6 |
| 22:16.54 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:17.49 | [TK]D-Fender | rasterix: rasterix Yes, and again you're expecting the 1.4 manual to care about the future. Does Apache 1.0 docs contain all the new rules for 2.0? |
| 22:18.01 | rasterix | yawn |
| 22:18.05 | galeras | Hello. Is possible to configure one E1/PRI and one E1 (unicall) in the same digium card (TE210p)? |
| 22:18.17 | [TK]D-Fender | rasterix: Attempting to rewrite the past to warn about the future is a waste fo time. Thats what upgrade.txt is for |
| 22:18.33 | rasterix | fender your wrong and your boring me |
| 22:18.39 | rasterix | lets move on |
| 22:18.41 | moy | galeras: of course |
| 22:18.56 | [TK]D-Fender | rasterix: Stop reading Ford Model T manuals to learn how to fix your 2008 Mustang. |
| 22:19.13 | [TK]D-Fender | Ok, off for qa while, back later |
| 22:21.07 | rasterix | shame he left... i was about to point out that THE BOOK for 1.4 uses "," as the delimeter |
| 22:22.45 | *** join/#asterisk thehar (i=thehar@xmission.xmission.com) |
| 22:23.38 | galeras | moy: i mean one E1 MFC/R2 and one E1/PRI, so there is no problem loading zaptel.conf and unicall.conf at same time? |
| 22:25.48 | moy | galeras: not at all, as long as they don't try to open the same zap device, and if you configure one E1 for PRI and one for MFC/R2, that means you will be using 1 span for PRI which chan_zap will take care of, and other for R2 which chan_unicall will take care of |
| 22:26.47 | moy | galeras: of course, you can also try libopenr2 and have everything in chan_zap :) |
| 22:27.08 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 22:28.07 | galeras | moy: is libopenr2 ready for production systems? |
| 22:28.34 | *** join/#asterisk joobie (n=joobie@201.023.dsl.mel.iprimus.net.au) |
| 22:28.58 | galeras | moy: any way, nice to know that is possible. Thank you |
| 22:29.32 | moy | galeras: okay, has been used for several people in Asterisk 1.2, 1.4 in production systems w/o issues, however is still under development and has not had a formal release yet, but is up to you, whatever you want to use |
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| 22:41.37 | jplank | h' |
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| 22:47.04 | edibrac | for voicemail, is the voicemail date based on the WAV file's filesystem date? |
| 22:47.33 | edibrac | i'm migrating over to a new box, and i'm not sure if i can just scp -r the /var/spool/asterisk directory over |
| 22:47.49 | edibrac | i suppose i can just be safe and tar it up instead |
| 22:48.01 | tzafrir_laptop | -r -p would be better (to keep dates) |
| 22:48.09 | tzafrir_laptop | rsync -a even better, I guess |
| 22:48.15 | edibrac | ah i'll do that |
| 22:52.13 | ManxPower | rsync -avvP -e "ssh" 8-) |
| 22:54.54 | outtolunc | you use quotes <G> |
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| 23:08.51 | drmessano | Skype? |
| 23:09.10 | ManxPower | drmessano: no thank you |
| 23:09.29 | drmessano | Was just reading the big announcement |
| 23:09.34 | thehar | Aye. |
| 23:09.35 | drmessano | All that comes to mind is |
| 23:09.40 | drmessano | "et tu, brute?" |
| 23:09.54 | ManxPower | drmessano: I suspect Skype will do for VoIP what AOL did for the internet. |
| 23:10.37 | ManxPower | I feel a disturbance in the Force, like 5 million totally computer illiterate people suddenly joining the Internet |
| 23:10.43 | drmessano | lol |
| 23:10.56 | drmessano | yeah |
| 23:12.31 | drmessano | I'm expecting lots of "So how do I install the asterisk PBX plugin in Skype?" |
| 23:12.37 | dlewis | drmessano: lol |
| 23:12.48 | galeras | Regarding linksys spa-400: someone has been able to select distinctive line (fxo port) for outbound call? |
| 23:13.07 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 23:13.13 | ManxPower | galeras: such a concept does not apply to calls going out an FXO |
| 23:14.20 | galeras | ManxPower: for us have meaning, because we have diffent pstn line in each spa400 fxo. |
| 23:14.43 | drmessano | Of course, this explains the moderately passive attitude of "We dont support skype here :)" to "We support all forms of VoIP here, troll someone for asking about skype and we'll ban you" in recent weeks |
| 23:14.45 | drmessano | :/ |
| 23:14.54 | galeras | MaxPower: *pstn line provider |
| 23:15.22 | ManxPower | galeras: the FXO port does not even generate ring voltage, how do you expect it to send out distinctive ring? |
| 23:16.25 | galeras | ManxPower: we don't want distinctive ring, we want to be able to choose the fxo port to make a call |
| 23:16.49 | ManxPower | galeras: I understand now. |
| 23:17.44 | galeras | so, no one here using spa-400 as a cheap fxo gateway? |
| 23:18.12 | boolean12 | I'm using an spa-3000. |
| 23:19.33 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:20.26 | drmessano | HAW!!! |
| 23:20.29 | drmessano | With over 100,00 installed systems, trixbox CE is the most popular full-featured, open source PBX distribution available. |
| 23:20.32 | drmessano | 100,00 ? |
| 23:20.46 | *** join/#asterisk xenonex (n=xenonex@89.218.233.88) |
| 23:20.48 | jaytee | is that comma in the right place? |
| 23:20.49 | dlewis | where are you reading that? |
| 23:20.59 | adr3nalin3 | How do install init scripts for centos? To have asterisk start automatically |
| 23:20.59 | drmessano | http://asteriskathome.sourceforge.net/ |
| 23:21.24 | jaytee | checkconfig asterisk on |
| 23:22.03 | drmessano | Im pretty sure their notice is in violation of TOS for Sourceforge since it also mentions a closed source commercial product |
| 23:22.19 | adr3nalin3 | jaytee: centos 5.2: error reading information on service asterisk: No such file or directory |
| 23:23.43 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 23:24.05 | jaytee | adr3nalin3, sorry, I meant chkconfig asterisk on and asterisk should not already be running. |
| 23:24.25 | jaytee | did you compile or install from packages |
| 23:24.37 | dlewis | drmessano: are there any stats that support for/against that claim? |
| 23:25.22 | jeev | anyone do sms gateway? |
| 23:25.55 | drmessano | dlewis: I could care less.. I was pointing out the extremely obvious mishap with the comma or lack of training zero |
| 23:27.35 | *** join/#asterisk mihinomenest (n=argh@66.255.220.17) |
| 23:30.02 | adr3nalin3 | jaytee: I used chkconfig (good ol tab completion) but same thing |
| 23:30.59 | *** join/#asterisk coppice (n=chatzill@184.204.17.210.dyn.pacific.net.hk) |
| 23:34.02 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 23:34.29 | andrewyager | adr3nalin3: You do have to manually install a file in /etc/init.d for auto start in CentOS 5 |
| 23:35.23 | andrewyager | cp /usr/src/asterisk-1.4.21.2/contrib/init.d/rc.redhat.asterisk /etc/init.d/asterisk |
| 23:35.24 | andrewyager | on my system |
| 23:35.33 | adr3nalin3 | ok thanks |
| 23:35.47 | andrewyager | then chkconfig asterisk on |
| 23:35.51 | andrewyager | service asterisk start |
| 23:36.55 | adr3nalin3 | thanks worked great |
| 23:37.01 | andrewyager | no probs |
| 23:37.33 | *** join/#asterisk bird_of_Luck (i=melifaro@secured.by.ipfw.ru) |
| 23:38.22 | bird_of_Luck | Hello people.. got quite a silly question: how to indicate ringging to user on answered channel ? |
| 23:38.48 | *** part/#asterisk raz (n=raz@unaffiliated/raz) |
| 23:38.49 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 23:38.51 | *** join/#asterisk Yourname` (i=Yourname@unaffiliated/yourname/x-837320) |
| 23:40.50 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 23:40.51 | *** mode/#asterisk [+o mog] by ChanServ |
| 23:41.47 | boolean12 | bird_of_Luck, Playtones(ring) |
| 23:42.04 | boolean12 | http://forum.voxilla.com/asterisk-support-forum/ringing-playtones-ring-comfort-ring-14558.html |
| 23:42.39 | *** join/#asterisk keith4_ (n=kbe2@216-164-144-176.c3-0.eas-ubr9.atw-eas.pa.cable.rcn.com) |
| 23:42.59 | bird_of_Luck | boolean12: tnx |
| 23:43.05 | keith4_ | am I crazy, or is the SPA942 backlit, but the SPA941 is not? |
| 23:43.22 | adr3nalin3 | spa941 is not backlit |
| 23:43.36 | jaytee | could that be what's behind the difference in model numbers? |
| 23:44.01 | boolean12 | There are a few more differences. |
| 23:44.09 | lanning | I think the major difference is 1 vs. 2 ethernets |
| 23:44.25 | adr3nalin3 | The SPA-942 is equivalent to the SPA-941, but adds a backlit LCD screen, second ethernet port and IEEE 802.3af power over ethernet (PoE). |
| 23:44.35 | keith4_ | okay. good. so i'm not crazy |
| 23:44.39 | boolean12 | Well said adr3nalin3 |
| 23:44.46 | adr3nalin3 | copy & paste |
| 23:44.47 | keith4_ | any other differences? |
| 23:45.03 | adr3nalin3 | http://images.google.com/imgres?imgurl=http://www.telephonyware.com/images/items/spa9xx.jpg&imgrefurl=http://www.telephonyware.com/telephonyware/tw00280.html&h=375&w=418&sz=27&hl=en&start=3&sig2=gSoH9c_1VwROZDJHtWhhbw&um=1&usg=__bO437CrJN0t-AfHoCvC251YXHWs=&tbnid=YkO0eJXGZGwmkM:&tbnh=112&tbnw=125&ei=LSLcSOzBBoigePTZhfgO&prev=/images%3Fq%3Dspa942%26um%3D1%26hl%3Den%26client%3Dfirefox-a%26rls%3Dorg.mozilla:en-US:official%26sa%3DN |
| 23:45.25 | keith4_ | ah, 4 sip registrations |
| 23:46.07 | boolean12 | the 941 and 942 can have 4 sip registrations. |
| 23:49.27 | keith4_ | before I waste any time, is running asterisk in a xen domU out of the question? |
| 23:50.05 | adr3nalin3 | I would not recommend running asterisk virtualized but I have no experience |
| 23:51.58 | boolean12 | I did it succesfully, but you'll need to run zaptel with a xen RTC patch |
| 23:52.07 | boolean12 | Since there is none :-p |
| 23:52.09 | keith4_ | ugh |
| 23:52.24 | boolean12 | Good luck :-p |
| 23:53.35 | scooby2 | Anyone know how to fix "stuck" calls in 1.4? |
| 23:53.37 | boolean12 | Or! You could patch asterisk to run async rtp intead of sync. |
| 23:53.45 | boolean12 | That would be far more complicated and useless. |
| 23:53.50 | keith4_ | heh |
| 23:53.51 | keith4_ | thanks! |
| 23:54.17 | boolean12 | ANYTIME! *gives you a thumbs up* |
| 23:54.32 | boolean12 | If you want help with running on xen, pm me. |
| 23:54.48 | *** join/#asterisk jks (n=jks@193.189.93.254) |
| 23:55.45 | andrewyager | I have a strange ongoing problem; our queue log is not reporting the time a call was answered while in a queue. We get total time (including hold time) but the log seems to think that the call ends as soon as it is answered. The calls aren't transferred and we are running 1.4.21.2 on CentOS 4.7 |
| 23:57.06 | keith4_ | do I necessarily need the latest zaptel with asterisk 1.4.whatever? |
| 23:57.14 | keith4_ | or can the versions be disjoint |
| 23:57.26 | andrewyager | It's buggy to run disjoint versions |
| 23:57.40 | andrewyager | usually they fix issues together, and later versions don't necessarily like earlier versions |
| 23:58.18 | keith4_ | that's what i figured |
