IRC log for #asterisk on 20120109

00:00.23p3nguinhttp://pastebin.com/tER2jGnY
00:00.33SeRiawesome. Thanks
00:04.19*** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com)
00:08.01SeRifor the iax.conf acl I use my brothers ip under allowed correct?
00:08.29SupYoshiF2Knight :D Working on the readme now, I just want to say im a complete Idiot and you were absolutly right on every single line :P Thank you very much just needed a littl push here
00:09.13p3nguindeny=0.0.0.0/0.0.0.0
00:09.14p3nguinpermit=209.16.236.73/255.255.255.0
00:09.30p3nguinwhere permit contains his address and netmask.
00:09.40SeRiawesome thanks!
00:09.51p3nguinIf his address never changes, a mask of 255.255.255.255 would be best.
00:10.02*** join/#asterisk wdoekes2 (~walter@wjd.osso.nl)
00:10.20SeRican hosts be used?
00:10.23p3nguinno
00:10.27SeRiok.
00:10.57p3nguinBut since his address is static, you don't need to worry with a name anyway.
00:11.17SeRigot it. I was wondering for something like callcentric.com
00:12.37p3nguinThey suck.
00:13.47SeRiindeed
00:13.52SeRi:(
00:14.24SeRiI cant seem to let them go for some bull shit reason... :(
00:16.02p3nguinYou've got a dirt cheap DID with them, so I understand.
00:16.24p3nguinBut to keep them means that you will put up with their crap.
00:17.30SeRitell me about it. you are been nice by saying *crap*....
00:18.49*** join/#asterisk vinhdizzo (~vinh@pool-74-100-182-10.lsanca.fios.verizon.net)
00:19.13*** join/#asterisk wdoekes2 (~walter@wjd.osso.nl)
01:10.37p3nguinWe finally got 100 Mbit cable service available to the consumer.  It has been testing in the lab for many months.
01:13.41*** join/#asterisk coppice (~coppice@m121-202-3-210.smartone.com)
01:16.17SeRinice
01:16.24SeRi3.?
01:16.29SeRi3.0*
01:29.25*** join/#asterisk bandroidx (~bandroidx@2607:f358:1:fed5:5:0:558:12)
01:33.02*** join/#asterisk bandroidx (~bandroidx@2607:f358:1:fed5:5:0:558:12)
01:40.08*** join/#asterisk TheBigS (~TheBigS@c-69-255-105-123.hsd1.va.comcast.net)
01:40.43TheBigSAnyone have any experience setting up Polycom 331 <--- complete asterisk noob
01:46.32*** join/#asterisk sixohquad (~Cody@184.65.142.249)
01:51.11sixohquadhey guys, is anyone able to help me out by telling me more about the errors im receiving here? im trying to dial out when this happens, was working fine, stopped working now and delivers that series of errors
01:51.14sixohquadhttp://pastebin.com/kNaxgquw
01:53.10WIMPyBefore you broke it? The sip peer seems to be missing.
01:54.18sixohquadok
01:55.38*** join/#asterisk SupYoshi (SupaYoshiL@ip51cc8577.speed.planet.nl)
01:56.18SupYoshiHi :) Does anyone know good dutch voices for asterisk?
01:57.01sixohquadmmm, 0 sip registrations
01:57.05sixohquadwtf
01:58.30*** join/#asterisk bandroidx (~bandroidx@2607:f358:1:fed5:5:0:558:12)
02:10.47*** join/#asterisk bandroidx (~bandroidx@2607:f358:1:fed5:5:0:558:12)
02:20.04sixohquadim receiving other errors now. http://pastebin.com/E1Qbrgr3
02:20.17sixohquadthat is my sip.conf, extensions.conf, and the errors are at the bottom
02:21.39[TK]D-Fendersixohquad: Your sip.conf has no [general] section and is completely broken because of it
02:21.48sixohquadthank you :)
02:22.22[TK]D-Fendersixohquad: I also se no register statements at all... so I don't see why you would expect even a hope of seeing an attempt let alone a success in "sip show registry"
02:22.32[TK]D-Fendersixohquad: that is for* registering outbound like to an ITSP, etc
02:22.40[TK]D-Fendersixohquad: not to check phones registering to *
02:22.43sixohquadi had one in there
02:22.48sixohquadi had a sip register line
02:22.48[TK]D-Fender"had"
02:22.51[TK]D-Fendermaybe
02:22.54sixohquadlol
02:22.57sixohquadkthx
02:23.12[TK]D-Fenderfirst [general] then registrations, THEN your peers
02:23.15[TK]D-Fenderorder matters
02:23.20[TK]D-Fendermissing bits = fail
02:26.09carrarhey
02:26.14carrarisn't there a book that covers this?
02:26.16[TK]D-FenderI'm off for  a bit...
02:26.26[TK]D-Fender~osmosis
02:26.26infobot[~osmosis] Osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ...
02:26.28[TK]D-Fender;)
02:26.34carrarheh
02:26.45carrar~book
02:26.45infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
02:33.04sixohquadi have the book
02:33.06sixohquadim just a n00b
02:33.18sixohquadi have it open beside me
02:33.38sixohquadbut when your a n00b l0zer sometimes you dont know what to look for to solve your problem :)
02:35.19*** join/#asterisk Yourname` (~Yourname@unaffiliated/yourname/x-837320)
02:38.29sixohquadshould the register line go in the general context?
02:40.31*** join/#asterisk bandroidx (~bandroidx@2607:f358:1:fed5:5:0:558:12)
02:41.07p3nguinIt is required to be in the general settings section.
02:42.28F2KnightSupYoshi, np, glad you got it.
02:47.14sixohquadok, now im running into this: http://pastebin.com/431zKZ9U
02:47.17sixohquadthanks btw p3nguin
02:50.08*** join/#asterisk luckman212 (~do-not-re@pool-108-41-8-176.nycmny.fios.verizon.net)
02:55.35p3nguinYour ITSP probably require you to be registered before you can make calls.
02:58.21sixohquadi am registered now
02:58.46sixohquadstate: registered
02:59.12p3nguinWithout adequate SIP debug, there's not much else I can guess about.
03:00.29*** join/#asterisk LostyJai (~blah@202.171.190.130)
03:02.02sixohquadwhat would you like me to do that will help you more?
03:02.38p3nguincore set verbose 3
03:02.39p3nguinsip set debug on
03:02.40p3nguinMake a call, pastebin the entire output.
03:03.25p3nguinTo turn sip debug back off, sip set debug off
03:06.40sixohquadhttp://pastebin.com/5ArEJqg1
03:07.15LostyJaihey all
03:07.29LostyJaigot a weird problem, when i dial 990 to log an agent on or off
03:07.54*** join/#asterisk roxdragon (~gianni@unaffiliated/roxdragon)
03:07.56roxdragonhi all...
03:08.05roxdragonhow to load ALL module on asterisk?
03:08.07LostyJaiwhen it says enter a new extension, i type in the agent's extension... it'll say agent is logged in, then "I'm sorry, that's not a valid extension"
03:08.21roxdragonNo such command 'sip show peers' (type 'core show help sip show' for other possible commands)
03:12.23p3nguinroxdragon: What does your modules.conf look like?
03:12.52p3nguinsixohquad: I don't see any problem indicated in your pastebin.  It looks like a successful call to me.
03:15.36*** join/#asterisk bandroidx (~bandroidx@2607:f358:1:fed5:5:0:558:12)
03:16.04roxdragonp3nguin, solved... but
03:16.05roxdragonERROR[2560]: rtp_engine.c:253 ast_rtp_instance_new: No RTP engine was found. Do you have one loaded?
03:16.18p3nguinroxdragon: What does your modules.conf look like?
03:16.59roxdragonp3nguin, http://paste.ubuntu.com/797842/
03:17.26roxdragonhow to load all modules?
03:18.01p3nguinLine 8 of that file takes care of doing that.  All you have to do is provide a conf for each module.
03:18.44roxdragonyes but don't load
03:19.04p3nguinWhat does your rtp.conf look like?
03:19.19roxdragon2 modules loaded
03:19.21roxdragonone moment
03:19.48roxdragonhttp://paste.ubuntu.com/797845/
03:20.24p3nguinThat one also looks okay.  Have you tried restarting asterisk recently?  core restart now
03:20.59roxdragonthat's ok
03:20.59roxdragon:D
03:21.02roxdragonthank u
03:21.34roxdragonit's work
03:21.41roxdragon:D thank youuuuuu goodnight
03:24.02LostyJaihttp://pastebin.com/7zGup75U
03:24.06LostyJaithis is the log of my error
03:25.38p3nguinThere is no extension by the name of 'return' in that context.
03:25.49p3nguinline 8
03:26.21LostyJaihow do i fix that?
03:26.43p3nguinCreate the extension.
03:26.54LostyJaicalled return?
03:26.55LostyJaiok
03:26.57LostyJaithanks
03:27.26p3nguinI'm more concerned with the reason the call is being sent to that extension in the first place.
03:27.39LostyJaino idea, just did a restore
03:27.48LostyJaiwhich config file would that be set in
03:27.50LostyJaiand what should ti be
03:28.09p3nguinNever heard of "a restore."  Sounds strangely like a Windows term.
03:28.12*** join/#asterisk sixohquad (~Cody@184.65.142.249)
03:28.29LostyJaitrixbox
03:28.31sixohquadp3nguin, it seems to connect the call properly now, but hangs up immediately after connecting.
03:28.42p3nguinYeah, we don't support trixbox here.
03:28.44p3nguin~trixbox
03:28.44infobotTrixbox is unable to be supported here.  It is a closed-source distribtuion of Asterisk and FreePBX containing proprietary code that its users do not have access to, making it difficult to support.  Try joining #trixbox and asking your questions there.
03:30.14LostyJaithought agent login/logoff was an asterisk thing
03:30.17LostyJaibut okay, thanks
03:30.44*** join/#asterisk brah (c82b2429@gateway/web/freenode/ip.200.43.36.41)
03:30.45p3nguinIt is, but it's too bad that YOU are not in control of asterisk.
03:31.06LostyJaiyeah i don't know my asterisk
03:31.23LostyJaisystem was build before my time, was just given to me
03:31.28p3nguinThere's a book for that.
03:31.30p3nguin~book
03:31.31infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
03:33.05sixohquadp3nguin, do you see any indication in the debug as to why it would hang up the call on me right away/
03:33.34p3nguinI didn't notice anything like that.  What I saw was a successful call.
03:34.18sixohquadhm, strange
03:34.22p3nguinHave you made calls to asterisk which stop at asterisk?  Have you made calls through your ITSP which originate at asterisk and not from a phone?
03:35.25sixohquadi can call other extensions internally just fine
03:35.32sixohquadhow do i originate a call from asterisk?
03:36.00sixohquadmake an extension that dials that number?
03:36.01p3nguinOn the CLI, use the "channel originate" command.
03:36.09sixohquadok will give it a shot
03:36.11p3nguinYou obviously already have the extension.
03:36.20p3nguinOtherwise it would not have made the call that you showed me.
03:36.23sixohquadyeah
03:36.24sixohquadtrue
03:36.43sixohquaddont quite have the full concept of dialplans yet, but i do see what you're saying
03:37.49p3nguinchannel originate SIP/call-labs/1604-your-number application Playback tt-weasels
03:43.30sixohquadughhh, putty keeps jumping my scrolling and im having trouble copy/pasting it because of it lol
03:44.21p3nguinI'm not sure what you mean.  If you can better describe the issue with PuTTY, maybe I can help you solve that so you can continue troubleshooting.
03:44.50sixohquadwhen i try to scroll back, something else hits the CLI and makes me jump back to the bottom of the log
03:45.17p3nguinsip set debug off, core set verbose off
03:45.21sixohquadi cant nice
03:46.04p3nguin*shrug*
03:46.30sixohquadoops, i only meant to type nice\
03:46.31sixohquadsorry
03:48.30sixohquadugh, i damn near pasted it all in here by accident
03:48.32sixohquadhttp://pastebin.com/nVPa6RkJ
03:50.01p3nguinLooks successful.
03:50.22sixohquadit didn't call my phone at all though?
03:50.40p3nguinIt had to have done something.
03:50.45sixohquadlol
03:50.59sixohquadeverything except call me :)
03:51.26p3nguinI'm not seeing the verbose messages in with the sip debug.
03:52.09p3nguinIf it's there, you'll have to point it out.  If it isn't there, you'll have to fix that.
03:52.24sixohquadit should have played back the weasels thing shouldn't it?
03:52.35p3nguinIt played it.
03:52.50p3nguinLine 184 says so.
03:53.15p3nguinBut if I had the verbose messages in here like I originally asked for, I could see the dialplan progression.
03:53.44*** join/#asterisk Zert (~zert@l49-17-157.cn.ru)
03:53.46Zerthellp
03:53.49Zerthello
03:53.50sixohquadim sorry i must have missed you asking for a verbose messaGE
03:54.07Zertis there any method to read DTMF in AGI during STREAM FILE execution?
03:54.13p3nguin(2102.37) <p3nguin> core set verbose 3
03:54.13p3nguin(2102.39) <p3nguin> sip set debug on
03:54.13p3nguin(2102.40) <p3nguin> Make a call, pastebin the entire output.
03:54.23sixohquadoh, its on verbosity 9
03:54.25sixohquadyou want me to lower it?
03:54.35p3nguin9 is just silly.
03:54.46ZertI don't want to wait until playback completed
03:54.51sixohquadoh ok lol
03:55.00p3nguinBut 9 would still show me the messages... which I do not see.
03:55.11p3nguinIf they are there, I need you to point them out.
03:55.59p3nguinWait...
03:56.05p3nguinDisregard.
03:56.16p3nguinYou aren't using any dialplan in this test.
03:56.23p3nguinThat's why it doesn't appear.
03:56.29sixohquadhttp://pastebin.com/SWMw2GG3 is the new pastebin
03:56.52p3nguinYou're connecting directly to SIP/your-itsp/your-number without dial plan.
03:56.53resist0rhaha 9 is just silly
03:57.27sixohquadp3nguin, im not sure i understand, how should i do it differently?
03:58.04p3nguinThe test does not get much more basic than this.
03:58.33p3nguinAsterisk originates a call to the destination specified.  When there is an answer, it runs the application specified.
04:00.06sixohquadyeah, i understand that, i thought thats what you were doing with the originate command?
04:00.19p3nguinThat's what I'm talking about.
04:00.27sixohquadwhenever i make a call from the softphone, it says connection established, and then says call ended
04:00.33p3nguinNote the part about the answer.
04:01.00p3nguinIf the channel does not go into state "Up" (answered), the application should never execute.  Yours executed.
04:02.08sixohquadhttp://pastebin.com/ZfKANcsx
04:02.15sixohquadthere's my extensions.conf
04:02.36p3nguinThis originate does not utilize extensions.
04:03.25sixohquadso what should i do?
04:03.34p3nguinYour default context is pretty useless there.
04:03.35sixohquadhow do you suggest i work through this?
04:03.47sixohquadi copied that context out of the book ;)
04:03.49p3nguinFind out how your phone is answering the call.
04:03.54sixohquadits not
04:03.56sixohquadguaranteed
04:03.59sixohquadits calling my cellphone
04:04.04p3nguinIt is.  Guaranteed.
04:04.04sixohquadand it doesn't even ring
04:04.11sixohquadno, there is something else going on
04:04.13p3nguinBecause:  (2201.00) <p3nguin> If the channel does not go into state "Up" (answered), the application should never execute.  Yours  executed.
04:04.15sixohquadmy phone is not answering the call
04:04.26p3nguinThere's an answer.
04:04.40p3nguinOtherwise the app would not execute.
04:04.52sixohquadfantastic
04:04.58sixohquadso there's something answering the call, and its not me
04:05.07p3nguinCall some other number.
04:07.35sixohquadphone never rang
04:07.37sixohquadhttp://pastebin.com/bPD5hPV2
04:07.42sixohquadcalled a different number
04:10.41p3nguinI've had another idea.
04:10.52p3nguinWhat is this call-labs thing?
04:12.13sixohquadjust a context
04:12.16sixohquadsip trunk
04:12.18p3nguinuh no
04:12.23p3nguin~trunk
04:12.23infobotsomebody said trunk was a word with varying definitions.  In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN.  There is no such thing as a "SIP Trunk" -- Don't use the term. the nose of an elephant
04:12.54sixohquadok, my sip provider?
04:12.58Yourname`If a SIP trunk is actually a GSM gateway - how does the DTMFing situation work if the GSM gw isn't sending proper DTMF tones?
04:12.59p3nguinIs it an ITSP?  Is it something of yours?  A company?  Need Details!
04:13.11sixohquadITSP :)
04:13.15sixohquadsorry i dont know what details to give yet
04:13.17sixohquadi'll get there
04:13.18sixohquadlol
04:13.43p3nguinMy thought is that they are poorly configured and they are answering calls when they should not be.
04:13.50sixohquadits vdi
04:14.04p3nguinI don't know what that is.
04:14.11sixohquadoh ok
04:14.30sixohquadits a company
04:14.35*** part/#asterisk Zert (~zert@l49-17-157.cn.ru)
04:14.47p3nguinCan you use another provider for a test?
04:14.56sixohquadsure
04:17.57sixohquadomg
04:17.59sixohquadit works fine
04:18.05sixohquadshakes his head
04:18.18sixohquadi'll find out whats going on with that provider tomorrow
04:18.26p3nguinOkay, so your provider is teh suck.
04:18.36sixohquadyeah, something's configured wrong
04:18.46p3nguinThey are not supposed to be answering your calls that you are sending out.
04:19.08sixohquadyeah thats no good
04:19.46sixohquadim sure its just a configuration error
04:19.54p3nguinI'm wondering if it would ever make it out ever.
04:19.59sixohquadyeah
04:20.07sixohquadi think the billing panel has just given me some wrong information
04:21.23sixohquadanyways, thanks for your help, sorry for running you in circles needlessly, you can imagine how bad it was confusing a nub :)
04:21.34p3nguinTry another test through call-labs.  Replace "Playback tt-weasels" with "Dial SIP/1001"
04:21.58sixohquadthrough the provider that didn't work?\
04:22.07p3nguinYou tried this already?
04:22.14sixohquadno
04:22.22p3nguinI'll wait.
04:23.41p3nguinAs soon as asterisk gets the answer indicator, your SIP phone should ring.  Go ahead and pick it up and wait a few seconds to see what happens next.  I'm curious what they are doing with your calls after they incorrectly answer.
04:24.44sixohquadok, this provider hasn't given me a proper register line i dont think, if i exclude it, will that ruin everything?
04:24.48sixohquadi have a context for it
04:25.07p3nguinYou can't have a "context" for it.  Contexts are something else.
04:25.45p3nguinRegistration is for telling the other side how to reach you.  Additionally, many providers will require you to register first before you are allowed to send calls inbound to them.
04:29.33sixohquadhrm
04:29.35sixohquadyou know what
04:29.41sixohquadi think i need to talk to this provider tomorrow
04:29.44sixohquadand see what is happening
04:30.13sixohquadafter we sort out how its supposed to be properly configured, i'll debug like this again and see if my calls get through
04:30.21sixohquadi think its likely a config problem, im probably missing something
04:30.27p3nguinWhen you tried that last test, did you end up talking to yourself?
04:30.34sixohquadthe fact that they haven't provided a register line could be the key to all of this
04:30.39sixohquadno
04:31.09p3nguinRemember, registration is to tell them how to reach you so they can send calls to you.
04:31.38sixohquadok
04:31.47sixohquadso outgoing should work fine without it
04:32.10p3nguinUnless they require registration first, outgoing has nothing to do with registration.
04:32.23p3nguinBut you said earlier that it was registered successfully, anyway.
04:34.21sixohquadhttp://pastebin.com/SDt0ejC1
04:34.48sixohquadyeah, its complicated and we're not too worried about incoming calls at this point so i'll confirm that part with my ITSP tomorrow
04:35.52sixohquadthere are multiple servers, and i dont think i was given the right information
04:36.03p3nguinThat gave a forbidden response.  THAT is what I would expect to see from a provider who requires registration before you can send calls in.
04:36.10sixohquadok
04:36.13sixohquadso i need that register line
04:36.19*** join/#asterisk bandroidx (~bandroidx@2607:f358:1:fed5:5:0:558:12)
04:36.42p3nguinIn addition to that, your application probably would have failed to run if the call would have been accepted.
04:38.27sixohquadok, i'll check with them tomorrow
04:39.39sixohquadannoying to find that this was caused by improper information, but im glad to know my asterisk works
04:39.43sixohquadwith the other provider
04:39.59sixohquadi'll go back to the book now :)
04:40.09sixohquadthanks again for your help p3nguin, i really appreciate your time
04:41.07p3nguinI hope it works out for you.  Let me know what they have to say about answering your outgoing calls.
04:41.53sixohquadyeah i'll let you know
04:41.58sixohquaddo you know chigambamukoko
04:42.00sixohquad?
04:42.08sixohquadirc nick
04:43.11sixohquadi gotta go get some grocery shopping done. ttyl, thanks again
04:43.36luckman212i'm curious about Asterisk's TRANSFER_CONTEXT
04:46.10luckman212there seems to be no mention of it in "the" ~book
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05:45.50[TK]D-Fenderluckman212: And "The Book" can't include absolutely everything without requiring a forklift
05:47.26Maliuta[TK]D-Fender: if it came with a forklift that'd be cool! ;)
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07:41.25F2KnightQ: Does anyone know a way to tell what functions/applications/commands are loaded by what modules? Some are easy but other modules load several additional tools.
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07:43.49schmidtsgood morning
07:44.01F2Knightmorning schmidts
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07:47.22kaldemarF2Knight: egrep -r "function name=|application name=" *
07:47.42IsUphello
07:47.55kaldemarF2Knight: the source files have XML documentation for the functions and applications they provide.
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07:51.14F2Knightkaldemar, should I run that in the source dir?
07:52.25F2Knightkaldemar, nvm running it in source gave what I was looking for .. Thanks!
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07:56.54mirelabQ: Does anyone know if queue member status shouldn't change if member of a queue is for example SIP/333@<ip> or SIP/333@<peer defined in sip.conf>
07:57.36MariusAgonMorning, guys. I have to send an e-mail from my dialplan, doing it with System() app. Maybe it's more linux  CLI based question, but how can I break line in my e-mail body? Thank you in advice.
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07:58.48F2Knightmirelab, sorry but can you explain your question better?
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07:59.41F2KnightMariusAgon, System() will execute your shell statement.. (I like mutt as an email client) \n is the standard new line char for linux systems
07:59.50mirelabF2Knight: I've posted it on Asterisk::forum => http://forums.digium.com/viewtopic.php?f=1&t=81197&sid=97269b2de52a547237f4203bfb7dbd36
08:00.29MariusAgonF2Knight thank you.
08:00.44F2KnightMariusAgon, you may want to consider some more thought in the logic of that call though.. You may find creating a simple shell or python script that you can pass variables to will work better to populate a templated email
08:02.17mirelabF2Knight: can you take a look on forum> i can explain here too if needed :)
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08:04.15F2KnightYour member line looks wrong...
08:04.32F2Knightyou have SIP/4003@4003
08:04.51F2KnightThat would dial extension 4003 using the sip account 4003
08:05.15F2Knightyour member should just be SIP/4003
08:06.35F2Knightor you may want to have the extensions in a context ... say  [myqueued-agents]
08:06.55F2Knightand then do a Local/4003@myqueued-agents/n
08:07.39F2Knightthe /n is magic on a LOCAL channel it maked the calls paths flow like other channels,.. normially local does not stay in the call paths.
08:07.58mirelabF2Knight: i tried with SIP/4003 what was logic to me too, but than I always get "busy noone has answerd..."
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08:08.33F2Knightare you trying to dial from the same phone?
08:09.38mirelabF2Knight: no, just want that [4003] to be dialed agent that will ring when member is in queue
08:09.58mirelabF2Knight: but status get stuck on Ringing
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10:07.25dj_hamstais there something similar to Digium Academy - Asterisk Essentials Online Video Training Course 6 month but free?
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10:12.06MariusAgonHello, guys. Trying to execute an php script from my dialplan, but seems i can't pass or catch variables(I'm poor scripter). From my dialplan i'm passing them like this : System(php /usr/queue.php ${number} ${time} ${queue}) (thoese set before in dialplan). Is this correct way? Thank you in advice.
10:13.28jacc0@MariusAgon: why are you using System() and not AGI?
10:13.43MariusAgonI'm not very familliar with AGI :/
10:13.54jacc0just rename your php to agi
10:13.56jacc0:)
10:14.50MariusAgonHmz, going to try that, thank you jacc0! :)
10:14.53olliiyou dont need to rename it
10:14.58olliiAgi(foo.php,123)
10:15.07olliilocation should be var/lib/asterisk/agi-bin/
10:15.12jacc0put this line in top of your php:
10:15.17jacc0#!/usr/bin/php
10:15.18ollii#!/usr/bin/php
10:15.20olliiyeah
10:15.31olliiif this is your php binary path
10:15.32jacc0;)
10:15.45MariusAgonOk,thank you, going to try that.
10:16.04jacc0I need to re-wire my pc; bye, ttyal
10:16.09MariusAgonbye
10:16.20olliiMariusAgon: http://www.voip-info.org/wiki/view/Asterisk+AGI
10:16.27olliithat might help you
10:17.24MariusAgonOK, thanks
10:22.33mirelabCan anyone help me with queue member status, => http://forums.digium.com/viewtopic.php?f=1&t=81197&sid=dd4647da750735dceeb67dca9eb16578
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10:26.56MariusAgonollii: still can't catch sent arguments, trying to do that with $argv[1] etc. And getting error "ast_carefulwrite: write() returned error: Broken pipe". Maybe i still miss something..
10:31.23mirelabMariusAgon: try to catch response to a new variable which you don't need to use after that, that can remove Broken pipe error sometimes
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10:33.25olliiMariusAgon: http://pastebin.com/ePvvFgnz
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10:34.32olliithis is simple
10:34.46olliithere already exists a phpagi class: http://phpagi.sourceforge.net/
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10:46.40DanFromUKhi, is there a quick way to end a sip channel from the cli?
10:47.12DanFromUKi can see the channel using core show channels, its an old channel that didnt end for some reason, but its blocking calls because of call-limit
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10:49.33olliidepends on your version...soft hangup [TAB]
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10:52.15DanFromUKthanks. that did the job
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11:31.27gegoHi. Is there a way to get a QuadBRI PCI card to work with dahdi? I have one runnning with misdn_v1 on an antique kernel but have to update this now.
11:32.13gegoI've tried with misdn_v2 which works but only in combination witch lcr - which I want to avoid
11:32.23WIMPyDepends on what you do with it. You might need a patch or you can use misdn v2.
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11:33.25WIMPydidn't like the ide of an extra application, but it has been the most stable solution so far.
11:34.03WIMPyAlthough I just discovered a bug in the transmission of dialling information between lcr and Asterisk.
11:34.05gegoWIMPy: I use it as an interface to euro-isdn - and to an analog fax - so both TE and NT mode are needed
11:34.39WIMPyDoes your card have 4 or 5 jumpers per interface for TE/NT selection?
11:35.20gegolet me have a look. Oh no wait: let me have a cigarette instead ...
11:37.03tzafrir_laptopgego, which card?
11:37.27tzafrir_laptopwhat does dahdi_hardware (from dahdi-tools) have to say about that card?
11:37.56tzafrir_laptopIf you don't have dahdi-tools installed and don't want to fully install it,
11:38.13tzafrir_laptopdownload the latest version and run from the source tree:  ./xpp/dahdi_hardware
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11:46.14gegotzafrir_laptop: it's a BeroNet BN8S0
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11:47.09tzafrir_laptopgego, so I suppose dahdi / chan_dahdi should support it
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11:47.57itsurkghello all, is there a repository for asterisk 1.8. I need to install asterisk1.8 with gui
11:48.03gegoI installed 2.6.0-rc1 - is that the latest?
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11:49.14joobiehey guys.. i usually use polycom cos the audio quality is great.. but i need a phone that has about 8 extensions (without adding an additional module to the polycom phones)
11:49.40joobiewhat would be the next best phone in terms of audio ?
11:49.51gegotzafrir_laptop, dahdi_hardware tells me: driver should be 'qozap' but is actually 'wcb4xxp' \n pci:0000:03:06.0     qozap+       1397:16b8 Generic OctoBRI ISDN card
11:49.54joobieneed to look at other brands cos polycom dont do one ofthose phones unless you buy their LED ones
11:49.56joobiewhich are pricey
11:50.12tzafrir_laptophmm... "generic" means it is probably not detected
11:50.20WIMPyjoobie: Snom 3x0 allow 12 accounts and have G.722 support.
11:50.25tzafrir_laptopMost likely will require a trivial patch
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11:51.05joobiegood audio WIMPy ?
11:51.30WIMPyjoobie: G.722 is wideband.
11:51.50joobieI dont use that
11:51.54joobieI use g729
11:51.57joobielow compression
11:52.06joobiehow does g722 compare in terms of b/w
11:52.20gegoWIMPy, it's got 4 jumpers each.
11:52.27WIMPy64kbps just like G.711
11:52.52joobieahh
11:52.59WIMPygego: In that case you will need the te_nt_override patch for wcb4xxp to get NT mode.
11:53.10joobieergh
11:53.11joobiegtg
11:53.14joobiebrb later
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11:53.36tzafrir_laptopWIMPy, are you sure? The card has no jumpers?
11:54.07tzafrir_laptop(feel free to comment on https://issues.asterisk.org/jira/browse/DAHLIN-203 )
11:54.11WIMPytzafrir_laptop: He said only 4. wcb4xxp needs the famous fifth jumper.
11:54.55tzafrir_laptopuses here a Junghanns card with just 4
11:55.10tzafrir_laptop(for testing, of course)
11:55.27WIMPytzafrir_laptop: he, I uploaded the updated patches there :-)
11:56.24tzafrir_laptopIIRC the problem with the patch there is that it breaks configuration with jumpers
11:56.30WIMPyWithout the patch you cannot set NT mode in software. It will only try to read the state of that fifth jumper.
11:56.42tzafrir_laptopThis is why that patch is on hold
11:56.56gegoWIMPy, where do I find these patches and to what do I have to apply em?
11:57.18WIMPyFrom what I read the configuration by jumper should still work. But I can't test that. My cards only have 4.
11:58.30WIMPyThe configuration ist very ugly anyway. You have to configure the same thing in three different places.
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11:58.59WIMPyEither (Jumper or module parameter), dahdi/system.conf and asterisk/chan_dahdi.conf.
11:59.30WIMPySo getting rid of at least one of the three places would be nice.
12:00.01WIMPygego: Follow the link tzafrir_laptop posted. The latest patch still works, even with some offset.
12:00.03gegomaybe I've just blacklisted/inserted the wrong modules (or maybe just in the wrong order) -> why does dahdi_hardware say: driver should be qozap ?
12:01.32WIMPyqozap is an older addon. Not sure how current that is.
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12:11.59WIMPyhas a feeling that WaitExten is eating the digits.
12:13.14schmidtsWIMPY WaitExten is allways hungry ;)
12:13.42WIMPyyes, but I can see the digits arrive, but the exten does not include all of them.
12:15.52schmidtsWIMPy how can you see it? dtmf debug?
12:16.13WIMPy"dtmf", yes.
12:16.47WIMPytries to add additional debug output.
12:23.30WIMPyUgh. Does someone know how WaitExten works? It seems to exig after the first digit is received. Where is the rest received?
12:29.33schmidtsWIMPy imho you can give a parameter how many extension you want to get
12:30.25WIMPyThe work seems to be done somewhere else.
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12:36.33dj_hamstai need a quick guide on how to add users and setting up sip phones within asterisk now 1.7
12:36.44dj_hamstai found an old one but the config is different
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12:38.49fobus912Hi All
12:39.03mirelabDeas anyone know why queue member status is not changing from (Ringing) if member is SIP/<ext>@<sip_peer> or (Unknown) if member is SIP/<ext>@<IP adress> and could it be changed?
12:39.06fobus912Does anybody know how to configure the dial plan to send DTMF after the call is bridged
12:39.31jacc0@dj_hamsta: edit sip.conf and ad a phone. edit extensions.conf and add a dialplan for that extension
12:39.31fobus912I have configured the dialplan as the following
12:39.32fobus912exten => _1XXX,1,Dial(SIP/${EXTEN:1}@FXO,,D(wAwwwwBwCwDwAwBwCwDwAwB))
12:39.42fobus912but the DTMF are sent before the call is bridged
12:39.52dj_hamstajacc0, i am using the gui portion freepbx
12:40.05jacc0then you are in the wrong channel
12:40.39jacc0join #freepbx
12:40.41fobus912can please someone advise the correct configuration
12:43.02jacc0@fobus912: create a contex in extensions,conf like: extens = _X.,1,sendDTMF(abcdabcdabc)
12:43.21jacc0and then use originate to connect a phone to this dialplan when it picks up
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12:45.18jacc0originate(sip/${EXTEN},exten,yoursenddtmfcontext,${EXTEN},1)
12:46.58fobus912Let me try this jacc0 Thanks for the suggestion
12:47.26kaldemarfobus912: maybe with U() or M() that uses app SendDTMF. D() will send the digits between answer and bridge, so that won't work for you.
12:49.28mirelabDeas anyone know why queue member status is not changing from (Ringing) if member is SIP/<ext>@<sip_peer> or (Unknown) if member is SIP/<ext>@<IP adress>
12:50.24fobus912kaldemar i just tried with Macro and usign M is not working
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12:56.43fobus912jacc0 can't get it working
12:56.46fobus912as you suggested
12:58.52fobus912is there any other way i can achieve it please
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13:02.47jacc0it works for sure; please show me your extensions.conf
13:03.12jacc0and your cli output
13:03.13jacc0~pb
13:03.13infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude.
13:04.59fobus912jacc0 here is my extension conf
13:05.00fobus912[test]
13:05.00fobus912extn=_1XX,1,SendDTMF(ABCD)
13:05.15fobus912originate(sip/${EXTEN:1}@FXO,exten,test,${EXTEN:1}@FXO,1)
13:05.42fobus912*originate(sip/${EXTEN:1}@FXO,extn,test,${EXTEN:1}@FXO,1)*
13:06.07fobus912I'm really not sure how i should configure it
13:07.29jacc0originate(sip/${EXTEN:1}@FXO,extn,test,${EXTEN:1},1)
13:07.33jacc0next time
13:07.35jacc0~pb
13:07.35infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude.
13:07.55jacc0so i can edit it :)
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13:10.56fobus912jacc0 is it the correct way to do it ?
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13:22.48jacc0originate(sip/${EXTEN:1}@FXO,extn,test,${EXTEN:1},1)
13:23.12mirelabDeas anyone know why queue member status is not changing from (Ringing) if member is SIP/<ext>@<sip_peer> or (Unknown) if member is SIP/<ext>@<IP adress>
13:23.14mirelab<PROTECTED>
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13:27.21jacc0you should remove the second @FXO
13:30.08BenC[UK]Can anyone give me names of any tier1 uk voip carriers?
13:30.12[TK]D-Fendermirelab, Do not expect member status that way.  use a Local channel to do the dial instead
13:30.52*** join/#asterisk rgsteele (~rgsteele@apo.aweber.com)
13:30.58*** join/#asterisk hrolf (~hrolf@unaffiliated/hrolf)
13:31.13hrolfHow do I get the wait time for a call abandoned in queue?
13:31.40[TK]D-Fenderhrolf, Its in the queue log
13:32.08mirelab[TK]D-Fender: Local/.../n with n to be normal channel, so i was told, but still too bad that status is not changing :(
13:32.27hrolf[TK]D-Fender: Sorry I failed to mention that I need it through the Manager API.
13:32.33[TK]D-Fendermirelab, It probabkly will once you change
13:33.07hrolfLike the Leave event (when a call leaves a queue) only reports the call ID, channel etc. But not the wait time.
13:33.21[TK]D-Fenderhrolf, Then you alredy knew your answer
13:33.45hrolf[TK]D-Fender: Does that mean I can get it?
13:33.50hrolfs/can/can't
13:34.06[TK]D-Fenderhrolf, No, it means you can already see that it doesn't give it to you the way you want.
13:34.08mirelab[TK]D-Fender: i think i worked with Local member and as i can remeber it's working fine, but i hate adding contexts with 1 line for a feature like that lol
13:34.41hrolf[TK]D-Fender: Fine. So is there something that I can do about it?
13:36.19BenC[UK]hrolf, can't you monitor events and keep track of time that way?
13:36.25[TK]D-Fenderhrolf, You've got the source....
13:38.31hrolfBenC[UK]: Yep, I was just asking if there is any already built solution for this thing. I think I'll have to do it my self by keeping track of the events.
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14:00.12qakhanhi all
14:00.58qakhani have 4 agents in a queue, when call comes in and no agent take the call then all agents logoff automatically
14:01.05qakhanplz help
14:01.54[TK]D-Fenderqakhan, read the sample queues.conf .  It lists the features that will kick agents who don't answer
14:05.36*** join/#asterisk serafie (~erin@nat/digium/x-jpgmqyhvxouidofl)
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14:15.44qakhan[TK]D-Fender can u tell me?
14:21.40*** join/#asterisk jacco_7564 (~Jacco@195-241-134-176.ip.telfort.nl)
14:21.44jacco_7564Hello
14:22.36jacco_7564the database asterisk is using, is that mysql?
14:23.40Guggeif you set it up to use mysql yes
14:23.45[TK]D-Fenderjacco_7564, what "database"?
14:23.59schmidtsjacco_7564 what asterisk version?
14:24.10[TK]D-Fenderjacco_7564, Only DB * inherently uses is AstDB and that is either BDB pre *10 or SQLite
14:24.10leifmadsenjacco_7564: the built in AstDB is a key/value pair DB based on Berkeley DB
14:24.19leifmadsenin Asterisk 10 and later, the backend is SQLite
14:24.24leifmadsenbut it operates the same right now
14:24.32jacco_7564Using asterisk 1.8
14:24.38leifmadsenthen it is berkeley DB
14:24.43[TK]D-Fenderjacco_7564, Then BDB
14:25.03jacco_7564Ok, thanks!
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14:48.55*** join/#asterisk Assalino (53f49842@gateway/web/freenode/ip.83.244.152.66)
14:49.07AssalinoHey hey
14:49.19Assalinototal Asterisk noob here
14:49.35AssalinoI'm following a tutorial so I can setup an Asterisk server with FreePBX
14:49.46WIMPy~freepbx
14:49.47infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
14:49.47Assalinowhich seems to be popular amongst the ASterisk devs
14:50.08Assalinothanks WIMPy. My question isn't related to FreePBX, I think
14:50.12AssalinoI just wondered
14:50.16Assalinoonce i set it up
14:50.20WIMPyGUIs certainly aren't popular in here.
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14:50.26Assalinocan I make a call from the service?
14:50.41AssalinoI'm not sure if ASterisk supports call making out-of-the-box
14:50.42WIMPyWhat service?
14:50.49Assalinofrom the server, sorry
14:51.07[TK]D-FenderFrom the server to where?
14:51.13[TK]D-FenderDoing what?
14:51.15WIMPyYou have to configure at least one peer and at least one extension to make a call.
14:51.17[TK]D-FenderWhen?
14:51.33Assalinowell, ideally what I want to do is create a service where users can input their phone numbers and get a call from ASterisk
14:51.39Assalinothat plays a recorded message
14:51.41[TK]D-FenderAssalino, Yes you can do this
14:51.46WIMPyHmm. No only an extension if you accept guests.
14:51.59Assalinoright now I'm just playing with it to see if I can build a prototype
14:52.10[TK]D-FenderAssalino, And so far FreePBX does virtually nothing to assist this process
14:52.26Assalinobut my guess is that I'd need to subscribe to some form of service to be able to make calls, no?
14:52.36[TK]D-FenderAssalino, naturally
14:52.52Assalinois this something Asterisk provide, or do you use a 3rd party?
14:52.58*** join/#asterisk sekil (~sekil@78.24.104.82)
14:53.06[TK]D-FenderAssalino, we call them "The Telephone Company"
14:53.07AssalinoI read somewhere you could do it via Skype and Google Voice, but I'm not sure if it's the same
14:53.10WIMPyYou need some sort of connection to the PSTN. Either on some sort of line or via an ITSP.
14:53.15[TK]D-FenderAssalino, And no, Asterisk is not the telco
14:53.32Assalinoso you can't do it purely via VoIP?
14:53.38[TK]D-Fender~itsp
14:53.39infobot[~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs.
14:53.40[TK]D-Fender^^^
14:53.59[TK]D-FenderYes, and "Asterisk" is not a service provider.  It is software that can talk to one however
14:54.06AssalinoI see
14:54.20AssalinoI knew that you could get hardware to connect an asterisk server to phone lines
14:54.27Assalinobut I thought we could use VoIP otherwise
14:54.33Assalinowhich seems ideal to me
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14:54.43sekildigium b410p is fully functional today? no need for misdn?
14:55.29[TK]D-Fendersekil, Been DAHDI compaible for years AFAIK
14:55.34WIMPysekil: Sort of. Depending on your card and needs, you might need a patch.
14:56.03WIMPyOr is it specifically about the b410p?
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14:56.20[TK]D-FenderWIMPy, Were there multiple hardware releases of that card for which a patch is required?
14:56.27*** join/#asterisk fobus912 (~fobus912@41.143.31.230)
14:56.31[TK]D-FenderWIMPy, Otherwise he did just tell you the model..
14:57.01Assalino@[TK]D-Fender, so if I set up a server, without having the hardware to connect to the ITSP, I won't be able to make calls?
14:57.20WIMPyI wan't sure if the question was about the card or the driver. As the obvious choice for that card would be dahdi.
14:57.30[TK]D-FenderAssalino, If your server has internet connetivity then you have the "hardware"
14:57.50[TK]D-FenderAssalino, And you need something to make calls.  What that is depends on what you want to use
14:57.50Assalinoyup, that's what I thought
14:58.05Assalinoright, so now I need to find a VoIP provider in the UK then
14:58.08WIMPyI've been told the b410p does not work reliably with misdn2.
14:58.09Assalinothat works with ASterisk
14:58.15[TK]D-Fender~itsplist-uk
14:58.15infobotextra, extra, read all about it, itsplist-uk is UK based ITSps include http://www.voiptalk.org/  http://www.voipon.co.uk/  http://www.gradwell.com/ and a few other tinpot companies you can dig up with google.
14:58.58Assalinoawesome
14:59.00Assalinoyou guys rule :)
14:59.54Assalinothank you
15:00.34sekilI'm still on * 1.2 and misdn
15:00.56WIMPyOuch
15:01.53sekilhm...looks like libpri is all I need nowadays..
15:02.34WIMPymisdn1 is also still maintained.
15:02.39WIMPyfor some reason.
15:03.03*** join/#asterisk irroot (~gregory@197.170.58.6)
15:03.20sekilI'd rather go for libpri and chan_zap/dahdi if possible
15:03.35sekilmisdn needed a lot of kernel patching etc.
15:04.13WIMPyYou don't need to patch, but the old version is no longer included, either.
15:04.16*** part/#asterisk irroot (~gregory@197.170.58.6)
15:04.26sekilold version?
15:04.34sekilmisdn1 you mean
15:04.36sekil?
15:04.42WIMPyyes
15:04.51sekilyou mean misdn is in the kernel nowadays
15:05.05sekil?
15:05.15WIMPyIt has been for may years.
15:05.31sekilbrushes the spider web
15:05.34WIMPymisdn1 up to 2.6.25.4 and misdn2 since 2.6.26.
15:05.59*** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com)
15:06.11sekilhm...
15:07.04sekilon 2.6.21 I had to patch
15:07.08sekilgood to know
15:07.33*** join/#asterisk voipeng (~voipeng@75-150-128-2-Philadelphia.hfc.comcastbusiness.net)
15:07.44WIMPyUnless you needed special features or some hardware ID update, you shouldn't have needed to patch.
15:08.15voipenghello, we turned up an additional server and it is replying to register requests on port 3:3 .... any idea where there is a static assignment for all 5060 traffic in asterisk?
15:08.32WIMPyOr actually, you never need to patch, as you can build the modules standalone, just like you would do with dahdi.
15:08.43voipengtcpdumps show it try to connect but the port connection is refused as its not a real registration port
15:10.03voipenghttp://pastebin.com/0pXd4H2d
15:10.19sekilyeah true...but I'm pretty sure misdn wasn't in 2.6.21 so I had to do something..
15:11.27sekildahdi doesn't support hfc-s single port cards also as I recall..
15:11.46WIMPyNot without 3rd party addons, no.
15:12.13WIMPyAnd no USB support at all.
15:12.15sekilyah
15:12.22sekilpci ones..
15:18.37*** join/#asterisk mirelab (~mirko@212.200.146.253)
15:19.32fobus912Hi all
15:20.03fobus912Is there anyone who can tell me if it's possible to send DTMF with SendDTMF application after the call is bridged
15:20.20fobus912using the D() option does send the DTMF but before the call is bridged
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15:23.51jacco_7564On outbound routes, is the prefix or prepend used to let asterisk know that it is dialing outside? for example i want to have to dial 00, and then a normal telephone number
15:27.41kaldemarjacco_7564: prefixes are one choice. read up on dialplan patterns and dialplan structure in general to get an idea on how decisions can be made in dialplan.
15:28.23olliigego: ask beronet ... they've a live support on beronet.com ... very friendly...
15:30.23[TK]D-Fenderjacco_7564, #freepbx <-
15:30.24gegoollii: beronet have a very friendly support for their berofix cards/boxes, but not for the pci-cards which have reached EOL ... for them ...
15:30.36olliioh okay :(
15:31.20WIMPyLuckily most of them are the same.
15:31.48olliiby now they're using hfc chips on their cards...maybe this old card is also hfc based
15:36.13voipengany suggestions why our registration server would reply like this? http://pastebin.com/0pXd4H2d
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15:43.43r0m|uis dijib in jail?
15:44.32Qwellwho?  probably.
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15:45.11r0m|ulol I havent seen him arround for a while.... all ways talking about getting high and what not....
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16:32.43p3nguinr0m|u: Maybe he's in jail for recording all his phone calls without consent.
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16:41.10r0m|up3nguin: lol
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17:01.26roxdragonhi all.. i recive this message when arrive incoming call
17:01.26roxdragondial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
17:01.31roxdragonwhat's this?
17:01.43Qwellroxdragon: more context needed
17:02.23roxdragoncan you help me?
17:02.34Qwellnot without more context, no
17:03.10p3nguinI would say that the phone you are trying to Dial() is not available.  Show us the sip debug.
17:03.32roxdragonQwell, extensions.conf?
17:04.07fobus912hi all
17:04.12roxdragonthe phone is registred
17:04.19*** join/#asterisk skrusty (~ksrusty@62.252.24.138)
17:04.22p3nguinShow us the sip debug.
17:04.29p3nguincore set verbose 3
17:04.33p3nguinsip set debug on
17:04.34fobus912can please someone advise how i can cheive the following, Dial DTMF from the dial plan after the channel is bridged
17:04.38p3nguinMake a call, pastebin the entire output.
17:05.08fobus912i was able to do it with D() but it dial the DTMF before the channel is bridged
17:05.29r0m|uman we have a nasty storm down here
17:05.58fobus912i'm fighting all day long to make it work
17:06.07fobus912and i can't find the solution for this
17:07.40roxdragonok
17:08.32fobus912anyone please ?
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17:22.45*** join/#asterisk odin917 (~Gavin_Sil@66.246.94.130)
17:23.06odin917Hi All! I am having an issue with bandwidth.com SIP trunks for incoming calls (outgoing works fine). when i call into my DID for my pbx i can see the traffic hitting my server (via tcpdump -s0 -n -v udp port 5060 and host 216.82.224.202) but i get no output in asterisk CLI.
17:23.07odin917here is the tcpdump wher ei can see the traffic http://pastebin.com/TFgHbA5c
17:23.48[TK]D-Fender<fobus912> i was able to do it with D() but it dial the DTMF before the channel is bridged <- I seriously doubt that.  Show us.
17:23.57[TK]D-Fender~pb
17:23.57infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude.
17:26.48*** join/#asterisk anthm (~anthm@freeswitch/developer/anthm)
17:27.01*** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com)
17:27.10SuperNullHey guys, anyone know of a commercial grade CDR billing software?
17:38.35QwellDoes "commercial grade" mean it has to be Java, and crash a lot?
17:39.04carrarI means it only runs in a DOS window
17:39.25[TK]D-FenderSuperNull, MS Excel
17:39.27WIMPyA Macro for Excel would be fine, I guess.
17:39.31Qwellha
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17:52.48paulcSuperNull: How about A2Billing - have you looked at that?
17:53.44*** join/#asterisk Troyblackman_ (~samuelmat@65-36-42-216.static.grandenetworks.net)
17:54.15Troyblackman_hello
17:55.19Troyblackman_anyone in here?
17:56.00[TK]D-FenderNope.
17:56.09carrarshhh
17:57.05p3nguinNow look what you've done.
17:57.11p3nguinNow he's going to ask things.
17:57.11*** join/#asterisk sekil (~sekil@78.24.104.82)
17:58.16Troyblackman_Ill just use forums
17:58.49[TK]D-Fenderlol
17:59.12[TK]D-FenderOr if you have a question you could just ask....
17:59.57Troyblackman_im just stuck on getting my did to accept incoming calls
18:00.13Troyblackman_ill research more on my own and see what I can figure out
18:00.20[TK]D-FenderTroyblackman_, PASTEBIN the failed attempt so we can see what's wrong
18:00.22[TK]D-Fender~pb
18:00.22infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude.
18:00.24[TK]D-Fender^^^
18:00.54Troyblackman_ok
18:01.58*** join/#asterisk sekil (~sekil@78.24.104.82)
18:01.58Troyblackman_Ill be back later on after I do a bit more research and if I cant figure out I will pastebin it
18:02.04Troyblackman_Thanks again
18:03.07*** part/#asterisk Troyblackman_ (~samuelmat@65-36-42-216.static.grandenetworks.net)
18:04.51drmessanolol
18:15.11jayteeI keep getting this message, "Windows has encountered an error and needs to close". I can't tell if it's an error or a feature though.
18:16.33[TK]D-Fenderjaytee, Ones that turn your screen blue are SPECIAL FEATURES
18:19.58p3nguinOh, I see the problem.
18:19.59*** join/#asterisk slidesinger-lt (~jtatum@c-174-57-5-70.hsd1.nj.comcast.net)
18:20.21p3nguin*Windows* has encountered ...
18:28.11*** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file)
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18:30.57r0m|ulol
18:31.26WIMPyWindows _was_ encountered.
18:36.12*** join/#asterisk longhorn (~ady@netaware.ro)
18:37.05longhornhi. I have a asterisk sistem going for 2 years now. I just changed the voice provider and now I get the following notice in logs: [Jan  9 20:36:47] NOTICE[2894] chan_sip.c: Registration from '<sip:1002@192.168.111.1>' failed for '192.168.111.3' - No matching peer found
18:37.24longhornand I can;t receive calls only place calls outside
18:37.42longhorncan someone help me with some advices
18:37.45longhornthanks
18:37.56beeklonghorn: Did you change sip.conf?
18:38.11longhornbeek no i change nothing
18:38.19beekWell, you're going to need to do so.
18:38.23longhornjust the cables from one device to another
18:38.31longhornplease guide me
18:38.47longhorna friend did the configs
18:38.48WIMPyWhat exactely did you change?
18:38.50beekWhat do you mean by "voice provider"?
18:38.51longhornand i don;t know much
18:39.05longhornfrom one company to another
18:39.26WIMPyCompanies of/for what?
18:39.29beekWell, you'll need credentials for the new company and make the appropriate changes in sip.conf to register with them.
18:39.32longhorncable from one provider device to another nothing to do with the astersk
18:39.53longhorni have access to all system
18:40.00longhorntell me what I have to do ?
18:40.02WIMPyAr you talking about phone lines? Or internet connection? Or what?
18:40.03beeklonghorn: look in sip.conf for the old registration information.  You'll need to replace it with the new.
18:40.17longhornWIMPy phone lines
18:41.00WIMPyOk, that shouldn't matter. Any other changes?
18:41.12longhornWIMPy none
18:41.17WIMPyLike e.g. new router or something?
18:41.25longhornbeek what in /etc/asterisk/sip.conf ?
18:41.29longhornno
18:41.45beeklonghorn: What exactly did you change?  Your telco provider?
18:41.48p3nguinsip.conf isn't going to be involved in the phone line configuration...
18:41.53WIMPyChanging phone lines will not affect your SIP clients. There must have been another change.
18:42.01longhornbeek yes
18:42.02p3nguinHowever, the NOTICE you received is regarding your SIP phone.
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18:42.16longhornlet me tell you mi configuration
18:42.18beeklonghorn: And these are real analog POTS lines and not a VoIP provider?
18:42.32longhornit is a VOIP provider
18:42.36longhorni'm from Romania
18:42.39longhornit is called UPC
18:42.55WIMPySo NOT phone lines?
18:43.27longhornthe cable that gets in MP-114 FXO it is normal phone cable
18:43.40beekWTF?
18:43.50longhorn192.168.111.3 device is a MP-114 FXO
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18:44.08longhornbeek maibe i can't explain this very right ...
18:44.46WIMPyAh, so somehting on that device has changed, apart from the other cable connected to it, it seems.
18:44.46p3nguinYou don't need to have your gateway device registering to asterisk.  Configure it for static.
18:45.25beeklonghorn: Let's start again.   Disregard the company for a moment... are you connecting to the PSTN via POTS (analog line) or via sip?
18:45.26longhornp3nguin and how i do that ?
18:45.54p3nguinDo you have access to the device?
18:46.34longhornp3nguin yes i have
18:46.37p3nguinIf you do not have access, asterisk may have to be changed instead.  I would prefer to configure the device, though.
18:46.56longhornbeek sip
18:47.45beeklonghorn: Pastebin your sip.conf file (***** out any password information)
18:48.54longhornhttp://pastebin.com/kZmGAkj6
18:49.42beekCrap....  FreePBX.   You'll need to pastebin all of those included files as well longhorn.
18:50.05WIMPyOr go to #freepbx
18:50.06p3nguin~freepbx
18:50.06infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
18:50.12p3nguinWe don't do FreePBX.
18:50.15p3nguinAnd we don't do Windows.
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18:50.42longhornp3nguin i have linux
18:50.51r0m|upukes
18:51.22beeklonghorn: pastebin those include files so I can see how you're actually connecting.
18:51.39p3nguinI suppose you aren't familiar with the phrase.  "I don't do windows" means that I do not wash window glass.  I was merely playing with the phrase.
18:51.42p3nguinCarry on.
18:52.42p3nguinneeds to call Deutschland.
18:52.43longhornhttp://pastebin.com/1DBq6Fpi
18:53.14p3nguinI suppose it is too late to be in the office, though.
18:53.30p3nguinI should have called them at 6 am here.
18:53.31WIMPy19:53
18:53.36WIMPyProbably
18:53.43p3nguin12:53 here
18:54.25beeklonghorn: Pastebin sip_registrations_custom.conf and sip_registrations.conf
18:54.48longhornbeek they are empty
18:55.27beekHow about sip_custom.conf and sip_additional.conf?
18:55.30beekhates FreePBX
18:57.48longhornhttp://pastebin.com/veAebDJX
18:58.13longhornbeek only sip_additional.conf has content
18:59.18beekbrb
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19:03.05adeel|worki'm experiencing call quality issues terminating calls to a few NPA's and was looking for an alternative SIP provider, than my current one (voip.ms)....any suggestions?
19:03.26adeel|workpreferably US based
19:03.28p3nguinflowroute
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19:22.19beeklonghorn: Just to confirm... you moved analog phone cables from one place to another and made no other changes to anything?
19:23.07longhornbeek right
19:23.34longhornthat is why I can;t explain ... and I don;t know much about this sistem
19:24.12beekAnd to confirm... it's     <asterisk> --- <MP-114> --- PSTN ?
19:25.08longhornyes
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19:26.23beekAnd what part did you physically change?
19:26.57longhornbeek just the wire
19:27.20beekFrom the <MP-114> to the PSTN...
19:27.22beek?
19:27.34longhornno
19:27.42longhornsorry
19:27.43longhorntes
19:27.44longhornyes
19:27.56WIMPymaybe, finally...
19:27.57WIMPyd'oh
19:28.26WIMPyThat alone wouldn't change anythin on the Asterisk side.
19:28.41WIMPySomethign else must have happend.
19:28.59beekDefinitely.  Something else has changed.
19:29.15[TK]D-Fenderlonghorn, pastebin "sip show peers"
19:30.30longhornhttp://pastebin.com/2JzESYi6
19:32.15[TK]D-Fender<longhorn> hi. I have a asterisk sistem going for 2 years now. I just changed the voice provider and now I get the following notice in logs: [Jan  9 20:36:47] NOTICE[2894] chan_sip.c: Registration from '<sip:1002@192.168.111.1>' failed for '192.168.111.3' - No matching peer
19:32.33[TK]D-FenderI know *I* don't see a "1002" in that list anywhere...
19:33.24WIMPyNo, it's 800.
19:33.27[TK]D-FenderAlso odd that so many other's are pointing to the same IP.
19:33.44[TK]D-Fenderthat reg attempt doesn't seem to think "800"
19:35.04[TK]D-FenderOh.. actually... there is the key bit..
19:35.09[TK]D-Fenderit CAN'T register.
19:35.12[TK]D-Fenderthe host is fixed
19:35.17[TK]D-Fenderand shouldn't be
19:35.48[TK]D-FenderAnd fromuser=800 is the wrong parm to make the names match.  That is for * calling out on it,
19:35.52WIMPyWell, maybe that message has always been there.
19:35.55[TK]D-Fenderit should have been "username=800"
19:36.10[TK]D-FenderSeveral minor silly breakages
19:36.23[TK]D-FenderFix the host, restrict the IP's fix the username
19:40.45*** join/#asterisk pdtpatrick_ (~pdtpatric@12.249.4.226)
19:41.29pdtpatrick_Question - is there anyway to see channel count with sip? sip show channels  ..something?
19:42.14akrohn'core show calls' could be close to what you're looking for
19:45.00pdtpatrick_that does it - thanks :)
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20:06.21p3nguinI would prefer to configure MY on-premise devices to use static entries in asterisk instead of registrations.  Registrations are for things that move around and/or change.
20:17.13*** join/#asterisk nny (~Scott@174.107.223.14)
20:17.54nnyI feel stupid for asking this, but does asterisk have a voicemail feature to allow access when the voicemail message is played as the user (example press * and login to hear messages). I can't recall and the voip-info doc is lacking
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20:18.47WIMPyYes, look at the a and o extensions.
20:19.28nnyWIMPy: ahh thanks, got it
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20:40.17p3nguinIs there a sound file that says "You do not need to dial a 1 to call this number"?
20:53.16malcolmdnot that i remember, no
20:53.36malcolmdeach sounds package includes a txt file that includes written descriptions of each prompt
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21:11.02p3nguinI looked there first, and I didn't see it.  Then I asked, hoping someone knew something about it that I didn't.
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21:12.37[TK]D-Fenderp3nguin, A very worthy commision for Allison
21:13.33p3nguinAlso one that says, "There is no need to dial a 9 before the phone number.  Please stop doing it."
21:14.03[TK]D-FenderKeep it up.. you'll hit a volume rebate level ;)
21:14.14p3nguinI know her hourly rate, but it would only take a few minutes to record a few sentences... does she have a minimum?
21:15.13_Corey_Digium used to have a one-off type of product on their website for these kinds of prompts... not sure if it's still around
21:15.20[TK]D-FenderDunno... pretty sure her rates are posted by hr site, otherwise fire off an e-mail...
21:15.30p3nguinMaybe we could get her to do it for the asterisk project and not for me.
21:15.38p3nguinI'll email her.
21:16.14[TK]D-Fenderp3nguin, If you're worried about usage rights You could hand them over.. but asking her to do it "for the poject" ... this is her living.  What she did alerady was billed
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21:43.26sixohquadIf you guys were doing offsite asterisk pbx's for companies, would you run multiple companies in one asterisk install, or would you virtualize your servers and run one company per virtualization?
21:44.02fenrusi'd give each company a vm somewhere
21:44.17fenrusamazon or some other cloudprovider
21:46.30*** join/#asterisk joesuffceren (~chatzilla@ip68-104-167-226.ph.ph.cox.net)
21:47.54joesuffcerenis anyone using a Dell R210ii server with digium hardware (TE121 specifically)? I am having trouble with it (two separate R210ii servers and two separate TE121 cards and no joy) that digium support (thus far) cannot sort out, and Dell support has told me "it's not a supported card"
21:48.52WIMPyIt might help if you tell us what the issue is.
21:49.10joesuffcerenI'm just wondering if it might be software related and/or if anyone else has done it before. I'm running the FreePBX distro on both boxes. I'm thinking of installing asterisk/dahdi/libpri  from source as a test.
21:49.23joesuffcerenWIMPy: getting there. :-)
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21:50.08joesuffcerenthe card throws PCI read/write errors in dmesg, and dahdi won't restart (gives an error from dahdi_cfg about span not existing).
21:50.18sixohquadSorry got disco. Did anybody answer my previous question?
21:50.24joesuffcerendigium tech SSH'd into the system and said the config is good
21:50.37WIMPyThat sounds evil.
21:51.46fenrusi have no really clue about theese matters
21:52.08fenrusbut i'd go to my server bios and try disabling stuff i dont need
21:52.15fenrusand enableing legacy mode
21:52.16fenrusetc
21:52.26fenruscould be some "feature" that fucks stuff up
21:52.29p3nguinIt seems much easier to keep things separated if each company has its own asterisk system.
21:52.37WIMPyMaybe latencies can be configured.
21:52.54WIMPyOr disable spread spectrum.
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21:54.44sixohquadP3Nguin, you're probably right, but the financial side is much more lucrative if you can fill a server with more clients :)
21:55.21sixohquadCan you include different files from inside sip.conf and extensions.conf?
21:55.23WIMPyAnd much easier to offer free calls between them :-)
21:55.45WIMPySure
21:56.26sixohquadSo keeping it clean would be easy. Keep each company in separate files.
21:56.47fenrusmysql backend perhaps?
21:56.54joesuffcerenWIMPy/fenrus: I'll look into those. thanks. I was just hoping for the silver bullet of someone saying "we had that same problem and resolved it by ___________
21:56.57joesuffceren:-)
21:57.25WIMPy____________ = Get a cheaper server.
21:57.28p3nguinYou could certainly use one system for multiple companies.  I didn't think that was the question.
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21:58.13sixohquadYeah, I was mostly fishing for pro's and cons of doing either way
21:58.26joesuffcerenyeah, corporate policy is to use dell for everything and use their financing, etc., so my hands are tied on that one
21:58.45fenrusoften reffered to as 'dellhell'
21:58.48sixohquadThat makes one hard drive failure epic though
21:58.54WIMPySo what cards do Dell support?
21:59.17joesuffcerennothing from digium. They do sell Cisco T1 cards, but I wasn't interested in going there
21:59.54WIMPyThey certainly won;t offer drivers for Asterisk.
22:02.33*** join/#asterisk macroevolve (~macroevol@c-98-234-125-202.hsd1.ca.comcast.net)
22:02.35p3nguinsixohquad: Sure, but you'll have a nice backup plan and disaster recovery plan to minimize downtime in the event of failure, so that would almost be a non-issue.
22:04.47macroevolvehey guys - i was curious for anyone that has started a outbound calling company - if one of your telemarketers is calling on behalf of Company A (Customer A) and the prospect want to be put on the DNC list, does this prevent you from calling that same prospect even on behalf of other companies?
22:07.05[TK]D-Fendermacroevolve: I cannot image it would
22:08.02macroevolveTKD:  thx!
22:08.33p3nguinMaybe if there is a certain significant common factor of the companies... and I don't mean that they use the same phone company.
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22:12.38amaninacanHey guys, I've got a dumb question about VoIP in general(let me preface this with the fact that I'm trying to win an arugment)
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22:13.15[TK]D-FenderSo what you're saying is you want us to tell you what you want to hear.  Gotcha.  Go on...
22:13.37sixohquadP3nguin, thx for the input. Ill keep it in mind as I learn more here
22:13.37sixohquadHaha
22:13.41amaninacanBasically, we've got a hosted VoIP provider that says in order to get their shit to work, we have to use port triggering. However, in every experince I've had with VoIP (I've managed setups from 3-300 phones) we've used something like STUN/ICE.
22:13.48amaninacanWhich is preferred?
22:14.09p3nguinYou don't NEED either of those options.
22:14.25p3nguinI don't use port triggering and I don't use STUN.
22:15.05amaninacanThat makes even more sense.
22:15.54*** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com)
22:16.46amaninacanSince I've been out of VoIP for a while now, remind me again why someone would need any of those?
22:18.40p3nguinSome people use STUN when they have a very nasty NAT implementation that SIP/RTP do not work with.
22:19.24p3nguinI've never had an occasion to use port triggering.  Ever.
22:19.30amaninacanYeah.
22:21.07amaninacanWell, thanks p3nguin. That was exactly what I needed to hear. Now I can go tell these idiots to shut the fuck up and fix their shit, or lose my business.
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22:46.15Dovidhi all. my OpenSipS server sends INFO packets to see if asterisk is alive. asterisk gets the info packets but does not respond. any idea why asterisk would not do that? running asterisk 1.8.X
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22:52.23CGMChrisOne of my customers (the owner of his organization, with about 8 phones), he wants to dial an extension and have everybody's phone ring and everyone all jump into a conference.  He's using a 4 line phone, so I'm guessing if this is possible it must us a function in the dial plan.  Anyone ever tried to do this?
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22:53.23WIMPycore show application Page
22:54.24CGMChrisWIMPy: That throws an error -> WARNING[17650]: app_meetme.c:800 build_conf: Unable to open pseudo device
22:54.47WIMPyYes, you need dahdi for that.
22:55.06pabelangerCGMChris: dial all the extensions then redirect the channels to a local conference call when they answer
22:55.14CGMChrisWIMPy: So, I need ztdummy since I don't have Zaptel hardware?
22:55.40WIMPythat would be dahdi today.
22:56.02WIMPyOr you do it manually from the dialplan and with ConfBrige.
22:56.06CGMChrispabelanger: Wouldn't that have to be linear...with one person having to answer before it could move on to the next Dial() request?  Dial is not async right?
22:56.33CGMChrisWIMPy: I'll go for the dahdi approach.  Thanks.
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22:58.02paulcWhat's the highest number of accounts/line appearances you've seen on a softphone? Most seem to max out at 5 or 6 - anyone know of a softphone with more?
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23:05.10pabelangerCGMChris: you could work some dialplan magic to blast them out at the same time using local channels
23:05.51CGMChrisI got it.  Thanks for help, I just needed to do some yum magic to install "dahdi-tools", updated dahdi-firmware and dahdi-linux and reboot and its good to go.
23:07.47p3nguinpaulc: 12
23:08.04paulcp3nguin: That'd do nicely - what product?
23:08.09p3nguinpaulc: Wait, you said soft phone.  Disregard.
23:08.13paulcDOH!
23:08.29paulc(which phone with 12? I've got 9 on an Aastra something or other and it works great for what I want to do)
23:08.39p3nguin5 or 6 seems about right for a soft phone.
23:08.58*** join/#asterisk digilink (~digilink@unaffiliated/digilink)
23:10.51p3nguinThe SPA509G is a 12-line SIP phone.  But with many phones, you can add expansion modules and have a crap-load more.
23:10.57*** join/#asterisk Micc (~Micc@c-24-19-33-189.hsd1.wa.comcast.net)
23:11.07CGMChrisCisco has the SPA504G with 4 lines, and another 9 series with 8 lines, and they both support the 32 line attendant console...so if you need lines/appearances and can get away from the softphone into a physical device, that's the ultimaqte
23:12.28[TK]D-Fenderattendant console != LINES
23:12.33[TK]D-FenderOn Cisco...
23:12.52[TK]D-FenderPolycom IP601+ support 12 w/ and extra attendant module
23:12.58paulcYeah.. no major problems using a physical device.. doing a bunch of testing with queues and stuff.. I like the Aastra cos it gives me 9 registrations
23:13.16p3nguin12 > 9
23:13.59paulcBut $["12" >  "9"] is false (as discovered today, randomly)
23:14.32p3nguinAlways nice to compare number as text strings.
23:15.59[TK]D-Fenderpaulc: Without quotes it will parse as numeric properly
23:16.01paulcThe clue was when the web service returned 6, it was tested against   > "50"   and returned true and I'm like "Hello WTF?" - then I realised what he'd done
23:16.32paulc[TK]D-Fender: Yup yup, did a quick bit of testing and realised "ah, it's properly strict" - so I'm fixing it all over the place now.
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23:25.29p3nguinpaulc: Need more lines?  http://www.digiumcards.com/images/digium_cards_images/sidecar_for_601.jpg
23:26.27paulcHaha lines to the max! Do the Polycom sidecars give you a registration per button then? I thought they were more BLF type functionality than registrations?
23:29.37p3nguinYou can use them for all sorts of things, including registrations for multiple accounts.
23:30.52Miccin 1.8.8.1 when I park a call and it rings back, its ringing back the sip trunk that the caller is on, not the one that parked the call. Is this how it's supposed to work now? Its kind of hard deciphering how its supposed to work by reading features.conf
23:31.06p3nguin"configurable as a line registration, call appearance, or speed dial"
23:31.52p3nguinParking should ring back to the device that parked the call.
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23:35.10Miccp3nguin, yeah it used to work in 1.6, but in 1.8 its ringing back to the trunk that the caller is on.
23:35.28p3nguinYou know the drill: show me.
23:35.48Miccok, but I gotta get some taco bell first. brb
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