IRC log for #asterisk on 20161228

00:03.49PenguinInside the macro, if the user does not provide input, but instead hangs up, the call goes back to the context where the mobile phone was Dialed.  The g option to the Dial keeps the call alive.  Without the g option, the call dies instantly.
00:04.28PenguinThe call needs to stay open so that the original Dial may timeout and send the incoming call to voicemail.
00:05.06PenguinThe problem is that the call is dying in the subcontext instead of the Dial reaching timeout and going to VoiceMail().
00:05.43WIMPyDoes that requite sip/SipPhone to be unreachable?
00:06.01PenguinNo, it can be reachable or unreachable.
00:07.09PenguinIf I remove the branched Dial in the incoming context, it doesn't matter if the sip phone is alive or not, the call will go to voicemail.
00:07.10WIMPyErm.
00:07.27PenguinIt's just dialing two things.
00:07.31WIMPyI don't see a Return from the macro.
00:07.33PenguinBut the second thing is a mobile.
00:08.21PenguinThat mobile has voicemail.  The mobile voicemail will answer the call to the mobile sometimes.
00:08.42PenguinI don't want the caller to ever hear the mobile voicemail.
00:08.48PenguinHence the macro.
00:09.10PenguinThe macro makes the mobile provide user input to take the call.
00:09.28PenguinOtherwise, the call should go back to asterisk's voicemail in the incoming context.
00:10.05PenguinIf the mobile answers the call and presses 1 to accept the call, the caller is connected to the person on the mobile.
00:10.09PenguinNo problem there.
00:11.02PenguinIf the mobile rings too many times, exceeding the Dial() timeout value, the call goes to asterisk voicemail in the incoming context.
00:11.06PenguinNo problem there either.
00:11.42PenguinBut sometimes the mobile has no signal or has a dead battery, and the mobile voice mail answers, plays a short message, then hangs up the mobile.
00:11.56PenguinThat's where the call dies instead of staying in asterisk.
00:13.26*** join/#asterisk friedrich (~friedrich@aextron.de)
00:18.19WIMPyOk, I've got a dirty hack: What if you call an additional local channel that just waits int the Dail inside the macro?
00:19.23*** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com)
00:20.05PenguinI suspect the same thing will happen, but I'm willing to test it anyway.
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00:24.52Penguin<PROTECTED>
00:24.52Penguin<PROTECTED>
00:24.55PenguinDidn't help.
00:25.11PenguinSame result -- call ended after the mobile voice mail hang up.
00:25.27WIMPySo it kills all 3 calls?
00:27.22PenguinI guess it did.  Call #1 is me calling the phone number which goes to incoming context.  Call #2 is going to SIP phone and Local channel where there is another Dial with a macro.  Call #3 is the call to the mobile.
00:27.35PenguinCall #3 ends when the voice mail answers then hangsup.
00:27.48PenguinThat ends #2, which ends #1.
00:27.50WIMPyOk, so all 4 call.
00:27.55WIMPys
00:28.14WIMPyI think it's time to show us what's ahppening.
00:28.28PenguinWell, call #3 calls the mobile and the Wait() like you asked me to test.
00:28.53WIMPyYes, so that's a total of 4 calls.
00:29.02WIMPyOr channels.
00:29.15PenguinIt's at least four channels.  Maybe five.
00:29.24WIMPyPlus the local channel.
00:38.53Penguinhttp://termbin.com/m7js
00:39.13PenguinThe mobile voice mail answers the call...
00:39.32Penguinasterisk plays the sound file and then executes Read().
00:39.55Penguin<PROTECTED>
00:40.05PenguinThe voice mail is playing the short message.
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00:40.28PenguinThe voice mail is not able to provide input...
00:40.31Penguin<PROTECTED>
00:40.42PenguinThe Read() restarts, asking for user input again.
00:40.51WIMPyOk, it says the call was answered. That seems to be the issue.
00:40.54PenguinThen the mobile voice mail is done and hangs up the call.
00:41.00Penguin<PROTECTED>
00:41.11PenguinAs soon as the mobile voice mail hangs up, it's all over.
00:41.24WIMPyBut then it still goes to t. I don't get that bit.
00:41.37PenguinThe g option to the Dial keeps it active.
00:41.44PenguinWithout the g option, it dies on the spot.
00:42.41PenguinNow I realize that the g option allows the callee to hang up and have the dialplan progress...
00:43.03PenguinSo I gave it other things to do after the Dial...
00:43.35PenguinThat prevents it from returning back to the first Dial which needs to end so the call can go to asterisk VoiceMail().
00:44.39PenguinI gave it a Ringing() and a Wait(60) just for fun.  It rang for a full minute before it then executed exten t.
00:44.45PenguinSo I removed that.
00:44.50WIMPyOk, but I don't see why that would kill the first dial.
00:45.53PenguinLet me do one without the g option.
00:48.19Penguin<PROTECTED>
00:48.19Penguin<PROTECTED>
00:48.38PenguinAnd when that channel dies, the incoming channel also dies.
00:49.42PenguinI need to be able to keep the call alive after the mobile answers and hangs up without having provided input to the Read().
00:50.14WIMPyTry to set the MACRO_RESULT right at the beginning of the macro.
00:50.40WIMPyI still don't see why the first dial should terminate.
00:50.51WIMPyOr it actually doing so.
00:52.55PenguinWhat do you suggest I set the result to?
00:53.06PenguinBUSY?
00:53.08WIMPyBUSY
00:53.11PenguinOK, let me test.
00:53.21WIMPyJust to make sure you have a default.
00:53.41PenguinYeah, I'll set it default as BUSY and then if user input it 1, change it to CONTINUE.
00:53.46Penguinor unset it.
00:53.55WIMPyYeah.
00:57.00Penguin<PROTECTED>
00:57.00Penguin<PROTECTED>
00:57.10PenguinOff to VoiceMail() !!!!
00:57.14PenguinI think that's the solution.
00:57.29PenguinThe problem was that there was no macro result.
00:58.01PenguinI'm going to eat supper and then try to clean this up and test more.
00:58.07PenguinI think that's the solution, though.
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00:58.11PenguinThanks a bunch!
00:58.15WIMPyGreat.
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01:05.14*** join/#asterisk arturon (~Anecchi@187.143.164.101)
01:05.30arturonHello good afternoon
01:05.56arturondoes somebody know if reverse polarity can be done in a mgcp channel?
01:06.24arturonI googled it with no luck
01:06.38WIMPyThat makes no sense.
01:06.50WIMPyIP doesn't have polarity.
01:07.07*** join/#asterisk rwb (~Thunderbi@65-183-151-239-dhcp.burlingtontelecom.net)
01:07.15arturonI'm registering Emtas devices
01:07.46arturonother call agents can perfom that
01:08.46arturonthis message in a request S:XAL/rev(-)
01:08.54arturonwith this message in a request S:XAL/rev(-)
01:09.43SamotIs that even supported still in the current versions of Asterisk?
01:10.18arturonI beleive so
01:10.30arturonmy release of asterisk is not too old
01:10.39SamotWhat version?
01:10.40arturon3 months probably
01:10.49arturonlet me give you the rev
01:11.08arturonAsterisk 13.8.2
01:11.34arturonit still have the mgcp channel
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01:13.14SamotIs chan_ccs an Asterisk module or third party?
01:13.28arturonI found lines like this in the config for dahdi channels can make the reverse polarity
01:13.29arturonansweronpolarityswitch=yes
01:13.54arturonbut it doesn't work when I configure it in the mgcp.conf
01:14.53SamotYou can't use settings from other channel drivers.
01:14.56arturonI don't know
01:15.04arturondo you think it could work?
01:15.07SamotNo.
01:15.18SamotThat's what I said "You can't..."
01:15.20SamotCannot.
01:15.40SamotChannel driver settings are not universal.
01:17.23arturonyes I was just trying to see if was programed the same way
01:17.37arturonbut I see that's not possible
01:27.47arturonwho can tell is there is some code in mgcp module to do reversal polarity or not?
01:30.41WIMPyOk, so you all like SIP, right? Can someone tell me what "SIP/2.1 200 Minimal Impact" means?
01:48.41ChannelZWell, I wouldn't say I "like" SIP, I tolerate it :)
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01:59.43WIMPyYou must be a very tolerant person.
02:02.12SamotWIMPy, error codes are more important.
02:02.20SamotThe message can be modified.
02:02.38SamotGenerally SIP codes follow the same as HTTP codes. So a 200 is an OK.
02:03.02WIMPyI think it might have been the response to a BYE for a call tat was already cleared.
02:03.36SamotRight but don't pull your hair out of a 200 error with a unique message.
02:03.41Samot200 = OK
02:03.48Samot404 = No User Found
02:04.07SamotEtc. There are more error codes for SIP but they follow the same logic.
02:04.09WIMPyYes, but it didn't look OK at all.
02:04.14SamotHah.
02:04.29SamotOK is just the ACK.
02:05.10WIMPyMaybe it was just a "200 ignored - shut up".
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03:47.28nfsnobodyhey guys, building a new 11.25.1 box with lua support, and I'm getting the below error on startup:
03:47.33nfsnobody- /usr/sbin/asterisk: symbol lookup error: /usr/lib/asterisk/modules/pbx_lua.so: undefined symbol: luaL_loadbufferx
03:47.50nfsnobodyI've gone higher verbosity, but can't get any more information... this appears to indicate to me that it maybe can't find the lua library? Any suggestions?
03:50.28nfsnobodyI did build lua from source, maybe there's a compiler flag I needed to add to show the source directory
03:52.10nfsnobodyI did ensure it was enabled in menuselect, but do I need to specify some sort of location in the configure?
03:52.15nfsnobodyI can't find a list of configure options for asterisk
03:52.36SamotYou're missing something LUA.
03:52.48SamotA dev package something that has the symbols.
03:53.21nfsnobodyyeah I thought I might be, all from source, so no packages
03:53.35nfsnobodyI've just found the --with-lua=PATH flag on configure, I think maybe it's pointing to system lua instead of the one I've compiled
03:53.44nfsnobodywould that path be libs, or binary?
03:54.26SamotYou set the path for lua when you installed it.
03:54.39SamotI don't use LUA so I have no idea where it installs.
04:01.04nfsnobodythanks Samot, you've put me on the right track, system lua.h and my local lua.h were in different dirs, recompiling now
04:04.23nfsnobodydamn, same error :(
04:05.43ChannelZDid you build it as a shared lib? And installed such that it put all of its headers someplace interesting as well? (this is normally the "-dev" version of a package in packaged systems)
04:06.50nfsnobodybuilding from source ChannelZ, but I checked out /usr/local/includes/lua.h which it created, and it specifies the version I've compiled form source (5.3), so I ran ./configure --with-lua=/usr/local/includes
04:06.58nfsnobodyit built, so I assume it found that header file and ran with it
04:07.12ChannelZcheck the config.log
04:07.34nfsnobodystrings /usr/lib/asterisk/modules/pbx_lua.so | grep liblua
04:07.34nfsnobodyliblua-5.1.so
04:07.39nfsnobodythat's interesting, that infers the wrong version
04:07.42ChannelZand also that you ran ldconfig to update the linked library system
04:08.39nfsnobodyohhh, that could be my issue
04:10.04nfsnobodydev libraries are a bit beyond my normal skillset, I just need to run /sbin/ldconfig right?
04:10.36nfsnobodyldconfig -p | grep lua
04:10.37nfsnobodyliblua-5.1.so (libc6,x86-64) => /lib64/liblua-5.1.so
04:10.42nfsnobodyok, definitely referencing the wrong library
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04:13.27nfsnobodyweird, I can't find a liblua-5.3.so anywhere
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04:27.31nfsnobodyok, looks like lua from source doesn't compile a .so by default, so I had to manually patch the makefile
04:27.38nfsnobodyrebuilding asterisk now, fingers crossed
04:30.03WIMPyDid you ./configure --enable-shared?
04:32.06nfsnobodyI did not
04:32.09nfsnobodyI probably should do that :P
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04:33.20nfsnobodyoh do you mean on lua WIMPy? it doesn't come with a configure script, just a makefile
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04:33.35WIMPyHuh?
04:33.55nfsnobodylua source doesn't come with a configure script, it just ships with a Makefile
04:34.32WIMPyNo cmake or waf, either?
04:35.02nfsnobodyjust a Makefile and a src dir, which has another makefile and normal source files
04:35.11nfsnobodyhmm, it still appears to be trying to use 5.1
04:35.13nfsnobodystrings /usr/lib/asterisk/modules/pbx_lua.so | grep "5\."
04:35.13nfsnobodyliblua-5.1.so
04:35.30nfsnobodyI don't get it, I'm linking it to the right header file now
04:37.18nfsnobodywhat causes it toselect liblua-5.1.so as the correct module to use?
04:37.39nfsnobodyI have a liblua-5.3.so in the same directory, and the .h file I point the configure to specifies version 5.3
04:38.18WIMPypkgconfig?
04:38.28WIMPyAnyway, I'm AFK again.
04:40.07nfsnobodyI'm reading through the asterisk configure script, it references lua 5.2 and 5.1, but not 5.3
04:40.12nfsnobodyconfigure:27253:LIBS="-llua5.1 ${pbxlibdir} -lm $LIBS"
04:40.12nfsnobodyconfigure:27141:LIBS="-llua5.2 ${pbxlibdir} -lm $LIBS"
04:40.21nfsnobodymaybe asterisk 11 doesn't have support for lua 5.3
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04:58.47D30hello everyone... i came across with webRTC.. i read some docs online but havent to tried it yet... came accross a tutorial on asterisk + webrtc using sip5ml .. now im a little bit confused as to whether asterisk is need to implement webrtc?
04:58.54D30this is the tutorial i found https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
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05:25.10SamotYes, it's needed.
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06:39.01D30@Samot thanks, but can you explain why? I've seen some webRTC in youtube without asterisk sip server included and so I am considering whether to used Asterisk server or not. Can you explain benefits of using Asterisk for webRTC?
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06:43.29SamotWebRTC is a SIP client that uses Web Sockets
06:43.36SamotYou need a system that supports WS/WSS
06:46.33D30WSS means secure websockets right? sorry not so familiar with the abbrv..
06:46.43SamotYes
06:53.44D30okay thanks.. another question.. do webRTC and Asterisk supports video call?
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06:55.09SamotYes.
06:57.50D30great thank you :)
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08:17.02[[thufir]]can you use webrtc with a phone?
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08:46.42D30i dont know yet if we can use webrtc with a phone.. what i know is that you can call directly using your web browser so there is no need to use a soft phone..
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08:48.03D30i keep on searching about webrtc implementation and i came across with this https://appear.in/ i think they are using webrtc...
08:48.44D30so the question still pops up on my head if its really needed to have an Asterisk server in place..
08:49.19D30i have a local asterisk server now and i like to play it with webrtc..
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13:17.08Maver|cKHello
13:17.18Maver|cKAny Asteirsk Guru Here to Help ?
13:18.02WIMPy~ask
13:18.02infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
13:22.12pawieckiHi. For blind-transferred calls, that failed (busy, timeout), I want the caller to be connected back to the transferer (peer that transfered the call). I think of setting variable TRANSFER_CONTEXT for transferring peer, and then making custom context to handle this. Is this a good idea to do it that way?
13:25.40Maver|cKIax2 DOnt Carry Callerid Dnid Information
13:26.16Maver|cKwhat variable shud i use to DialPlan for tht
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13:36.00WIMPyWhat exactely is your question?
13:36.09WIMPyAnd why are you sending empty messages?
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15:08.09Maver|cKis there any function call dont answer and play file to user press any key if press key is correct call pass to desired dialplan
15:09.37SamotWhat are you trying to do?
15:10.43Maver|cKi m trying to authenticate a caller
15:10.53Maver|cKwheather it is geniune caller or robot dialer
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15:11.19SamotSo you want the call to go to a recording first?
15:11.26Maver|cKno
15:11.30Maver|cKi dont want recording
15:11.38Samot"play file"
15:11.42SamotThat would be a recordin.
15:11.48Maver|cKyeah
15:11.54SamotA pre-recorded file.
15:12.05SamotSo you want the call to be answered and playback a recording..
15:12.10WIMPy'core show application Read'
15:12.12Maver|cKi used background function while playing a file for example press 1 to proceed a call
15:12.33SamotIf they press the proper digit they can make it to a device?
15:12.34WIMPyThat's another way to do it.
15:12.37SamotThat's an IVR
15:12.38Maver|cKbut when i press 1 it change my extenstion
15:12.44Maver|cKyeah
15:12.50SamotYes, IVR's need a destination.
15:12.55WIMPyThat's why Read is easier.
15:12.56Maver|cKi want to dial a dnid
15:13.00SamotSo if 1 is the correct digit..
15:13.07SamotYou do something with it.
15:13.17WIMPyAnd you don't need an extra Application to play the file.
15:13.18Maver|cKyeah
15:13.25SamotThen when they press 1 your dialplan executes a Dial().
15:13.33Maver|cKyes
15:13.39Maver|cKi need this one
15:13.45SamotIt's an IVR
15:13.49SamotWith a dial command.
15:14.03Samot~book
15:14.03infobotAsterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
15:14.31Maver|cKi shud see read function ?
15:14.52WIMPyRead is an application, not a function.
15:15.03Maver|cKokay
15:15.09Maver|cKsorry i m noob in asterisk
15:15.10SamotMore importantly..
15:15.13Maver|cK:)
15:15.21SamotYou asked this question in #freepbx
15:15.27SamotAre you using FreePBX?
15:16.25Maver|cKno i m using asterisk
15:16.35SamotStraight Asterisk?
15:16.37SamotNo GUI?
15:16.37Maver|cKyup
15:16.40Maver|cKno gui
15:17.05SamotOK, then asking this question in the other room wasn't productive.
15:17.06Maver|cKall setting are made using cli mode
15:17.34SamotSo you send an incoming call to an IVR context.
15:17.43SamotThat IVR context has the digit you want to match..
15:17.47Maver|cKyes
15:17.57SamotIf that digit is pressed, you use Dial() to send the call out to another DID
15:18.04SamotOr to a device.
15:18.06Maver|cKyeah
15:18.08Maver|cKto device
15:18.25Maver|cKbut ivr shud no anwer
15:18.26SamotWell that's it.
15:18.34SamotWhat should answer then
15:18.41SamotWhat should answer the call?
15:18.42Maver|cKonly play file
15:18.45SamotSomething has to ANSWER
15:18.53WIMPyIt has to answer. Otherwise it can't work.
15:18.53Maver|cKcall will answer by the device
15:18.57SamotYou can't play a file on an unaswered call.
15:18.59SamotNo.
15:19.02SamotThat's not how it works.
15:19.06SamotThe call is answered by the PBX
15:19.11Maver|cKi did it using background
15:19.17WIMPyYou can play a file without answering, but you can't read DTMF without.
15:19.22SamotBecause an IVR answered the call.
15:19.25Maver|cKcall come i set background file with noanswer
15:19.29SamotOK.
15:19.38SamotYou have a PBX
15:19.41SamotThat's where the calls go.
15:19.45Maver|cKyeah
15:19.46SamotThe PBX decides what to do with it.
15:19.52Maver|cKcall will go to my pstn line
15:19.56SamotSo I call your DID, I get your PBX
15:20.02SamotOnce I press the proper digit
15:20.05Maver|cKyes
15:20.06SamotThe PBX calls your device.
15:20.12Maver|cKyeah
15:20.16SamotYou answer, the PBX bridges the channels.
15:20.22Maver|cKyes
15:20.29SamotSo the first thing that needs to answer is an IVR
15:20.33DovidHi. Does the Via need to stay the same in a dialog or can it change? I have an issue where a cisco sends an invite where the via is it's external IP when then send back a 100 trying, 183 with SDP then the cisco sends back a cancel but the via is the cisco's internal IP (and not external IP as before) so OpenSipS just passed it along to Asterisk instead of handling the cancel its self
15:20.34SamotBecause you want them to press digits.
15:21.07Maver|cKif i answer the call billing will charge to my client
15:21.26SamotYou can't have them press digits in a call that's not answered.
15:21.26WIMPyThat's the way it is.
15:21.54Maver|cKbut i tested in sip trunk i get it wht i want
15:21.57SamotIn fact, it will cost more to the client doing this.
15:22.05WIMPyYou can (technically) delay answering the call until the file has finished playing.
15:22.07SamotBecause the call is being handled longer.
15:22.18SamotOh he's trying to stop robo dialers.
15:22.19WIMPyBut if your provider allows it, is another question.
15:22.30SamotHe wants human interaction to continue the call.
15:22.30Maver|cKyup i want to stop robo dialers
15:22.44SamotWhich means the call needs to be answered by something that determines that
15:23.01SamotGenerally, the user.
15:23.34Maver|cKi did it in sip trunking
15:23.47SamotWhat are you talking about?
15:23.47Maver|cKbut issue is with iax
15:24.12Maver|cKsip client call to my asterisk server ... i transfer call to my another asterisk server using iax2 trunk
15:24.12SamotYour provider supports IAX?
15:24.14*** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta)
15:24.26SamotSee..
15:24.31SamotThat's an important piece of this.
15:24.41WIMPyIs it?
15:24.48SamotCalling between two servers
15:24.57SamotCall being on one and the call being sent to another..
15:24.58SamotYeah.
15:25.07SamotFor a n00b
15:25.08WIMPyWhat difference does that make?
15:25.23SamotIt's more than just PSTN -> PBX -> Phone
15:25.23Maver|cKnow
15:25.31SamotPSTN -> PBX ->PBX -> Phone
15:25.31WIMPyCalling a SIP device or an IAX device is not that different.
15:26.08SamotI didn't say IAX was the important part
15:26.19SamotI said TWO PBXes in the mix is the important part.
15:26.28Maver|cKissue is iax2 dont send callerid or dnid
15:26.39WIMPyDialling another PBX usn't any different from dialling a phone, either.
15:26.55SamotI know.
15:27.15Maver|cKi just want to know what i use instead of callerid
15:27.15SamotBut when you think the layout is PBX -> Phone, you work a certain way.
15:27.36WIMPyWell, there's two solutions: 1. don't mess with the extension. See above. Or 2: Save the original extension as you have to do in all cases where you mess with the extension.
15:27.39Maver|cKi bebug iax2 trunk is showing only called number
15:27.42SamotAn additional PBX adds to the equation.
15:27.59Maver|cKdebug*
15:28.05SamotWhy do you need to use the caller ID?
15:28.14SamotI asked what you were trying to do...
15:28.16Maver|cKdnid to dial out the number
15:28.29SamotWhat number?
15:28.29Maver|cKi want to dialout the dnid
15:28.41Maver|cKthe number hit to my dialplan
15:28.47Maver|cK1st
15:28.56SamotLet's try this from the top.
15:28.59SamotI call your DID
15:29.02SamotIt goes to your PBX
15:29.03Maver|cKokay
15:29.05Maver|cKyes
15:29.08SamotYou make me press a digit
15:29.11Maver|cKyes
15:29.14SamotI press 1 and that's OK
15:29.19Maver|cKokay
15:29.24SamotThen it dials the other PBX?
15:29.29WIMPycloses this window for a while
15:29.39Maver|cKit will dial to my pstn line
15:29.49SamotSo it's sending the call back out to the PSTN?
15:29.54Maver|cKyes
15:30.06SamotOK, so then you still use Dial()
15:30.13SamotAnd it routes back out your PSTN lines.
15:30.17Maver|cKbut when i use background when it say press 1 .. dialed number changes to 1
15:30.23SamotRight.
15:30.31Maver|cKi dont want tht number to change
15:30.31SamotBut that doesn't matter.
15:30.44SamotThen you need to store the DID in  variable.
15:31.04Maver|cKbut call coming from iax2 trunk
15:31.14Maver|cKin sip trunk i can make wht i need
15:31.19SamotWhat IAX trunk?
15:31.32SamotBetween you and your provider?
15:32.24Maver|cKlike my client using sip he send call to my asterisk sever A .. i PAssed This Call tow my asterisk server b using iax2 trunk
15:32.45Maver|cKit is okay fine till now
15:33.00Maver|cKnow my asterisk server b will asterisk please press 1
15:33.20Maver|cKclient press 1 but .. my extension change to 1 i want to dial the dnid number he send
15:33.43SamotThen you need to store it.
15:33.52Maver|cKhow can i store tht number
15:33.59Samot~book
15:33.59infobotAsterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
15:34.12Samot^ You're going to need to read up on dialplan and functions.
15:34.57*** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com)
16:04.30*** join/#asterisk [TK]D-Fender (~joe@64.235.216.2)
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16:19.48mcflopyhello
16:22.00mcflopyi am new to asterisk and i use it with the freepbx frontend. i have configured some extensions and a sipgate trunk - everything works fine. now i want to include a second trunk which must be connected over another route. so sipgate trunk over 192.168.0.1 and the german telecom over 192.168.0.2 - is this possible?
16:22.46Samot#freepbx is the best place.
16:22.49Samot~freepbx
16:22.49infobot[~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
16:23.47mcflopyoh okay - thank you :)
16:23.48WIMPyI'm not sure that's an Asterisk related question at all. But it's too vague to tell.
16:24.19SamotWell unless there is a VPN or some sort of direct connection between him and the provider, private IPs aren't going to route publicly.
16:24.24mcflopyWIMPy: i think this "only" needs some pf config magic... but.... i am not deep into it :)
16:28.52igcewielingmcflopy: If the 2nd IP is on a different subnet from the main IP of the box, then the OS routing rules will apply.   If they are on the same subnet, well good luck with that.
16:29.57WIMPyWe don't even know what these IPs belong to.
16:31.23igcewielingGenerally the OS routing doesn't care who the IPs belong to.
16:33.19*** join/#asterisk iheartlinux (~jwpierce3@mail.johnthecomputerguy.com)
16:34.36WIMPyOk, let me rephrase: We don't even know if they are local IPs, IP of gateways or IPs of a proxy.
16:35.10iheartlinuxblind transfer on polycom 550's: what is needed to retain/forward caller id? the originating sip/username is what is being displayed on destination ext.
16:35.15[TK]D-FenderWIMPy, Answered in #freepbx
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16:48.52mcflopyso i solved it with a simple route command :)
16:48.58mcflopyroute add -net 217.0.0.0/13 gw 192.168.0.5 dev eth0
16:50.12[TK]D-Fenderiheartlinux, Then you aren't doing a blind transfer
16:51.19iheartlinux[TK]D-Fender: then what am I doing?
16:52.05[TK]D-FenderIffffSo far not confirming the exact steps and methodology
16:52.48[TK]D-Fenderbecause a proper one jsut works
16:53.21WIMPyUsually even a blone transfer just works.
16:53.26WIMPy+d
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17:04.49*** mode/#asterisk [+o putnopvut] by ChanServ
17:09.04iheartlinuxwell for some reason the receiving ext is showing the sip id
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17:23.18igcewielinglike this?  74afe02a609ec5e3512e35ec6a89d009@208.22.22.150:5060
17:23.39igcewielingah, sup username, not sip ID
17:24.16*** join/#asterisk chris1980 (~chris@host86-177-212-20.range86-177.btcentralplus.com)
17:26.23igcewielingPolycom phones can be told how/where to get the incoming callerid info.
17:26.54iheartlinuxsome freeking module I wasn't loading. went back to autoload.
17:29.15igcewielingvoIpProt.SIP.CID.sourcePreference: Specify the priority order for the sources of caller ID information. The headers can be in any order. If Null, caller ID information comes from P-Asserted-Identity, Remote-Party-ID, and From in that order. The values From,P-Asserted-Identity, Remote-Party-ID and P-Asserted-Identity,From, Remote-Party-ID are also valid.
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17:41.16chris1980hi, ive been following the tutorials and managed to get a basic call working between two softphones on mobile handsets working, however ive tried to introduce secure calling but I get unknown ca and failiure to register 503 error, im stumped as how to resolve this as others seem to had had success with the same
17:52.20*** join/#asterisk tomcruise (tom@eyeswideshut.xs4all.nl)
18:04.20igcewielingTry: https://www.google.com/webhp?q=asterisk+tls+503
18:04.56*** join/#asterisk BlackMaria (~BlackMari@dsl-104.163.140-101.ebox.ca)
18:09.44SamotIf it's a self-signed cert, you need to put the cert and ca on the device.
18:17.35chris1980ive tried multiple methods, resetting the keys and retransferring them to the devices, tried just using the same host/server, im going to try using openssl instead of the ast script
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18:23.00Samot503 is a service unavailable error.
18:23.25SamotIs this  Chan_SIP or Chan_PJSIP?
18:26.36chris1980chan_sip
18:28.34chris1980the asterisk console gives error tlsv1 alert unknown ca and warning file * open failed
18:29.30chris1980but when sip reload is run it finds both the key and the crt with no warnings
18:30.01SamotDo: asterisk -rx "sip show settings"
18:30.03Samot~pb
18:30.03infoboti guess 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
18:30.10Samot^ give the pb link.
18:35.35chris1980http://pastebin.com/h8SSdrF1 <- link from the search bar after clicking create paste
18:37.53igcewielingThe worst thing about Asterisk's TLS support is the total lack of useful error messages for some common problems.
18:38.15igcewielingfor example, the actual file which failed to open.
18:40.21*** join/#asterisk miralin (~Thunderbi@194.8.128.47)
18:40.58chris1980it took me a while to figure out where to put the actual tls config aswell, the tutorial wasnt to clear about where in sip.conf to actually put it
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19:01.09SamotAllowed transports:     UDP
19:01.45SamotNm...that's not for this.
19:03.46SamotShow your sip.conf and the TLS settings.
19:08.43chris1980http://wwww.pastebin.com/Apzs00VN
19:17.19SamotAnd those are owned by the user that Asterisk runs as?
19:17.25SamotSo it can open and read those files?
19:20.56igcewielingcan you use s_client (or openssl client) to connect to tcp/5061 check the certificates and keys?
19:22.21chris1980im running asterisk with sudo, which i had to use for the ast script also, im pretty new to linux so im not sure if the permissions are correct
19:23.05Samotls -l /etc/asterisk/keys
19:23.07Samot~pb
19:23.07infoboti heard 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
19:24.18igcewielinghttps://www.feistyduck.com/library/openssl-cookbook/online/ch-testing-with-openssl.html
19:25.16chris1980http://www.pastebin.com/g77egsZ4
19:27.45SamotWhat user does Asterisk run as?
19:28.22chris1980i run it from the chris but I have to use sudo otherwise it does not work
19:28.32Samotsudo to what?
19:28.37SamotTo root?
19:28.39igcewielingyou sudo to 'chris'
19:28.45chris1980sudo asterisk -rvvvvv
19:29.01SamotThat's the CLI/console
19:29.10igcewielingso you are *not* running it as user chris.
19:29.17chris1980i run service asterisk start to run it
19:29.18SamotSo basically it's running as root.
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19:29.33SamotYou do "sudo service asterisk start"?
19:29.42SamotOr just "service asterisk start"?
19:30.12igcewielingyou tell what user for asterisk to run as in asterisk.conf or -u and -g.  don't use sudo
19:30.14chris1980when i run service asterisk start it asks for my passwrd
19:30.51chris1980but to get to the cli i have to use sudo otherwise it wont run
19:31.08igcewielingchris1980: getting to the CLI and starting Asterisk are two totally different things.
19:31.22chris1980i know, I just wanted to let you know
19:31.41igcewielingsudo to root.
19:32.05SamotWell outside of ownership permissions...
19:33.12Samotrw only by the owner isn't going to work.
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19:35.27chris1980so i should make the asterisk keys owned by root perhaps ? apologies
19:35.50SamotWhatever user that Asterisk is truly running as.
19:36.04igcewielingchris1980: If you start Asterisk as root, then you won't have this issue.
19:36.11SamotThere's that.
19:36.48igcewielingthen once Asterisk does all the "rooty" stuff, then Asterisk will change its userid to whatever is configured in asterisk.conf and/or the -u and -g options.
19:36.57SamotBut the permissions should be at least 644, probably 664.
19:37.22igcewielingSamot: Nothing sane will start with a world readable key.
19:38.01SamotOK. So at least 640
19:38.05lvlinuxHi everyone! Can i stop asterisk from regenerating DTMF when it detects a feature code? In other words if a feature uses something like the star key, can I prevent asterisk from sending that star to the other end of the call and just process the feature?
19:38.23igcewielingin order to run asterisk -rvvv the file /var/run/asterisk/asterisk.ctl needs to be read/writable by the user running asterisk -rvvv
19:38.47igcewielinglvlinux: as far as I know, that is the default.
19:39.12lvlinuxigcewieling: hmmm, well it doesn't seem to be that way...
19:39.29igcewielinglvlinux: maybe asterisk isn't detecting the feature code?
19:39.51lvlinuxI have a feature code set for * and the other end is catching the * and doing it's own thing. (the feature plays back a file, which asterisk does as it should)
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19:40.08igcewielinglvlinux: does * in fact playback a file?
19:40.16lvlinuxyes
19:40.55igcewielingWeird.  Maybe nobody tried using * as a code instead of *X or *XX before.
19:41.09igcewielingdoes the same happen with # or even "5"?
19:41.13lvlinux(it's a voicemail system on the other end, with dtmf codes for functions, so I need it to not hear any dtmf from my side when the feature code is pressed)
19:41.31lvlinuxi didn't use a # or single digit, but I did try * and then a digit.
19:41.45lvlinuxsame result (asterisk played the file as it should but the other end reacted to the code too
19:41.53lvlinuxlet me try just with # or a single digit
19:42.37igcewielingin any case, if Asterisk sees the digits as a feature code, then it should handle it locally and not pass the digits on.  Otherwise nothing would work right in asterisk.
19:42.53lvlinuxthat's what I thought shoud happen.
19:43.07igcewielingpaste your features.conf, maybe someone will want to dig into it.
19:44.40lvlinuxsingle digit does same thing.
19:44.55SamotShow the call.
19:45.00lvlinuxlet me check over my features.conf first and make sure I didn't mess with something years ago...
19:45.04SamotShow what the dialplan is doing with the call..
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19:55.36lvlinuxhere you go: http://pastebin.com/wF3yWZ3R
19:57.24igcewielingI don't see a sequence to match a single *
19:57.50lvlinuxi changed it to 9
19:57.53igcewielingIt is very possible Asterisk requires a min of 2-digit feature codes.
19:58.20lvlinuxbut it worked fine---Asterisk played the file as it should with both double and single digits
19:59.08igcewielingif it passed the DTMF, then it is NOT working fine.
19:59.28igcewielingyou could try asking on the asterisk-users mailing list or maybe file a bug report.
19:59.32lvlinuxwell i mean that it did recognize a single digit at a feature code and processed it.
19:59.45lvlinuxs/at/as/g
20:03.18SamotWait, this is because when you're on the call the other party hears you pressing DTMF?
20:07.02lvlinuxyes
20:10.20lvlinuxi want the other side to hear nothing but the file being played back
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20:21.00SamotHrm..
20:21.15SamotI think the issue is playbackfile => 9,caller,Macro,playbackfile
20:21.24SamotI think caller should be "self" or "peer"
20:22.49lvlinuxk i'll try that
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20:27.59SamotWho's supposed to hear the playback?
20:28.34lvlinuxboth sides preferrably
20:28.47lvlinuxpeer seemed to work, only i couldn't hear it, but that's ok
20:29.22lvlinuxthanks for the tip!
20:29.36SamotYou can try self/peer
20:29.45lvlinuxyou mean both with the slash?
20:30.00SamotYes.
20:30.05lvlinuxoh yes i see it in the help
20:30.18Samotself = channel that initiates the feature
20:30.22lvlinuxi forgot about that---i was thinking there was only activated by
20:30.22Samotpeer = other channel
20:30.39SamotActivatedBy is no longer honored.
20:31.17lvlinuxhuh? Then wouldn't it ignore the "peer" in "self/peer"??
20:31.27lvlinux;<FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>[,<AppArguments>[,MOH_
20:31.45SamotActivatedBy is no longer honored. The feature is activated by which channel DYNAMIC_FEATURES includes the feature is on.
20:31.48igcewielingthe most you should expect to hear on the other end is a short chirp when Asterisk intercepts DTMF.
20:32.01SamotUse predial to set different values of DYNAMIC_FEATURES on the channels. Historic values are: "caller", "callee", and "both".
20:32.53lvlinuxyeah that's right (it's been a while since I messed with features)
20:33.16SamotI think you need self/peer because there's no "both" option.
20:33.34igcewielingLooks like someone didn't read features.conf.sample
20:33.49lvlinuxSamot: so it would use both as options to ActivateOn?
20:33.56SamotI think so.
20:34.14SamotHaven't done it, I'm just basing this on what I see in the Wiki.
20:34.17SamotAnd what I would do.
20:34.22SamotWhich is set it and test it.
20:34.56lvlinuxigcewieling: I read it, that's why I'm confused! lol
20:35.19lvlinuxThe line I pasted above is from the sample file
20:35.36lvlinuxand still lists the ActivatedBy.
20:35.42SamotSo does the Wiki
20:35.51SamotBut the Wiki also has the lines I pasted.
20:37.42igcewielingwhich wiki?   https://wiki.asterisk.org/wiki/display/AST/Home or voipinfo,.org 8-)
20:38.38igcewielingthe first wiki should be the one of the first places you look.  the second one should be the last place you check.  voip-info.org is outdated and disorganized.
20:39.28SamotThe Asterisk Wiki is always the first place I look
20:41.51[TK]D-Fenderthe sample configs should be the first place....
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21:17.38Samotlvlinux: Did self/peer work?
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23:55.36funxiondoes anyone why a call would not hunt through the codecs between asterisk and freeswitch?

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