[asterisk-users] call between snom 300 and aastra 6731i
hello list i need your help please regarding an issue with snom300 and aastra6731i using asterisk 11.13.0 asterisk snom 300 8.7.3.25 astra 6731i 2.6.0.2019 i have configured the trunks like below 100 in snom300 200 in snom300 300 in aastra6731i 400 in x-lite the calls between x-lite and aastra ok inbound and outbound the calls between x-lite and snom300 ok inbound and outbound the issue just between snom and aastra i can call from aastra to snom without issue but when itry to call from snom300 to aastra6731i i get bad request all the time i test with 3 snom300 i get the same result please any body have the snom and aastra can help me in order to fixe this issue thanks and regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Determining if a queue member is paused in Dialplan logic. [1.8]
I use a Perl script that monitors AMI events. It also checks the state of all queues and members and generates some basic HTML pages for monitoring the queues. It's not perfect, nor would I call it pretty, but it gets the job done. If you are interested, I can send it to you. Dale On Wed, Mar 25, 2015 at 7:18 PM, John Kiniston johnkinis...@gmail.com wrote: Thank you Kevin, I've looked at your solution and while I agree it's not ideal it does appear to be something that might work for me. I'll see if I can maybe backport the QUEUE_MEMBER stuff to 1.8 from 11. I'm also exploring an idea with a co-worker of using an AMI listener that will fire off actions in response to the member being paused and doing things that way. I looked at parsing the log but sadly the log uses the Member Name in the log instead of the actual device so I don't have a way of knowing what handset they are logged into the queue from. On Wed, Mar 25, 2015 at 12:13 PM, Kevin Larsen kevin.lar...@pioneerballoon.com wrote: First, let me say I feel dirty for even posting this. It is probably far from ideal, but it does get the job done. I had the same issue. Also, I am using Asterisk 11. I just looked and it doesn't appear that the QUEUE_MEMBER function supports the paused option in 1.8. To be honest, I am not sure if there is a good replacement for what I have done below in the 1.8 series. It isn't elegant and if you have a lot of queues/queue members to check, it will constitute a lot of looping, but it does work. Like you, I would like to have a way to check the pause status of a member easier. If the queue application could call a subroutine with it autopaused someone, that would actually make an elegant solution, but for now, this was the way I could see to do it. You could maybe call a script that would parse the queue_log file looking for an agents status and pass that back into the dialplan. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anonymous SIP calls
You have to consider whether you really want anonymous calls, or you just want to enable SIP calls from trusted companies/partners. The latter means setting up routes to these companies and (ideally) registration between peers. If you really want anonymous calls, then you will have to setup your dialplan with a guest/anonymous context for the calls to drop into. Once they arrive in that context you can route them anywhere else in your dialplan based on rules you setup. To help understand how this works, set verbose up to 10 in the Asterisk CLI and then call into your PBX using a SIP phone (without registration) . You'll quickly see how it works. The bigger concern here is security. Hackers will have a field day with an unsecured SIP connection. You will want to add some security on and around your Asterisk server. Take a look at http://www.voip-info.org/wiki/view/Asterisk+security for suggestions. To be conservative, assume someone WILL find a hole in your dialplan and attempt to commit fraud (i.e. rack up charges on your phone system). You will want to add security to your asterisk server which detects this fraud and disconnects the callers. There's a great video of an Astricon attendee explaining how callers racked up $100,000 in charges in one weekend. From: asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of James B. Byrne byrn...@harte-lyne.ca Sent: Thursday, March 26, 2015 9:24 PM To: Asterisk Users List Subject: [asterisk-users] Anonymous SIP calls We have a FreePBX-12 / Asterisk-12 setup that supports about 24 extensions, most internal Snom870s but six or so external (Jitsi-2.8). we use TLS and SRTP everywhere on our side of the fence. The server host is a dedicated atom(tm) box using the FreePBX distro (CentOS-6.x) and is up-to-date. Registrations require very long random passwords and registrable devices are further restricted by netblock filters. We have the usual firewall and fail2ban intrusion prevention and detection set-ups in place. Our connection to the rest of the world is via PSTN. We do our own DNS, both forward and reverse. We have NAPTR and SRV RRs for SIP and SIPS. That is the environment. Now for the questions. Can I safely configure FreePBX/Asterisk to allow people to call us directly via SIP? In other words, sip://someth...@harte-lyne.ca would reach us and ring internally as if someone had called our main office number via PSTN. Does it make sense to do so? I am not talking about routing our main number through a SIP trunk provider. We will remain on PSTN for the foreseeable future. But I am curious as to whether or not it it worthwhile to allow others who have the capability to simply call us via SIP rather than over PSTN. And if we do allow it what are the caveats and how does one actually configure Asterisk to do it? I have read a number of blogs, sections of the Definitive Asterisk book and mailing list archived posts respecting anonymous SIP calls. But I have to say these leave me rather more confused than informed. Virtually all sources advise against accepting any anonymous incoming SIP calls whatsoever. The few that do not absolutely advise against do not give much guidance in how to handle incoming calls. And frankly, I have only a dim idea how an incoming SIP call should be handled from a theoretical point of view. Any guidance would be welcome. -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gateway Eurotech
Hi, I know there are people with much experience in asterisk, and I want to ask if anyone had experiance with this gw http://www.eurotech-communication.com/products/voip-gateways/VoIP-32-CHANNELS-2E1-PRI-1U/ I'm having trouble getting connect with asterisk anyone has any production? regardss -- rickygm http://gnuforever.homelinux.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Auto Answer
2015-03-23 11:08 GMT-06:00 ricky gutierrez xserverli...@gmail.com: Hi , I'm having some problems with functions enable auto answer in some Grandstream GXP 1405 , I have enabled this feature in the snom 821 phone and work gr8 , in the gandstream not work, I enable the function on the phone Allow Auto Answer by Call-Info: yes Dialplan: exten = 501,1,SIPAddHeader(Call-Info: answer-after=2) exten = 501,n,Page(SIP/140SIP/110,d) exten = 501,n,Hangup() not work for me, it ring but does the function of auto answer Any idea? I found the problem, my mistake, annex the solution for someone else to help exten = 501,1,SIPAddHeader(Call-Info: answer-after=0) exten = 501,n,Dial(SIP/140SIP/137SIP/112SIP/113SIP/122SIP/120SIP/131SIP/132SIP/116SIP/136SIP/111SIP/125SIP /124) -- rickygm http://gnuforever.homelinux.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR dst value null after attended transfer
On Thu, Mar 26, 2015 at 10:24 AM, Vinicius Fontes vinic...@aittelecom.com.br wrote: I'm having an issue with CDR. Basically, I expect to have all legs of a call having the same linkedid and differing only by the sequence value. That does happen, but I'm getting null dst values after doing an attended transfer. I'm not sure if this is a bug or I'm doing something wrong. I'm running Asterisk 13.2.0. Here's the console log, step by step: First, I receive a call from 5491549116 on extension 7051 (DID 5421047051): [Mar 26 12:11:04] == Using SIP RTP TOS bits 184 [Mar 26 12:11:04] == Using SIP RTP CoS mark 5 [Mar 26 12:11:04] -- Executing [5421047051@restrito:1] Goto(SIP/pabx-e1-0252, interno,7051,1) in new stack [Mar 26 12:11:04] -- Goto (interno,7051,1) [Mar 26 12:11:04] -- Executing [7051@interno:1] Macro(SIP/pabx-e1-0252, stdexten,7051,SIP/7051) in new stack [Mar 26 12:11:04] -- Executing [s@macro-stdexten:1] NoOp(SIP/pabx-e1-0252, STDEXTEN: Arg1 = 7051 Arg2 = SIP/7051 Arg3 = ) in new stack [Mar 26 12:11:04] -- Executing [s@macro-stdexten:2] Dial(SIP/pabx-e1-0252, SIP/7051,45,tT) in new stack [Mar 26 12:11:04] == Using SIP RTP TOS bits 184 [Mar 26 12:11:04] == Using SIP RTP CoS mark 5 [Mar 26 12:11:04] -- Called SIP/7051 [Mar 26 12:11:05] -- SIP/7051-0253 is ringing [Mar 26 12:11:11] -- SIP/7051-0253 answered SIP/pabx-e1-0252 [Mar 26 12:11:11] -- Channel SIP/pabx-e1-0252 joined 'simple_bridge' basic-bridge b1c97b75-bd5f-4762-96dd-7aa68c472827 [Mar 26 12:11:11] -- Channel SIP/7051-0253 joined 'simple_bridge' basic-bridge b1c97b75-bd5f-4762-96dd-7aa68c472827 Now, extension 7051 places the call on hold and calls 7003, who answers: [Mar 26 12:11:17] -- Started music on hold, class 'default', on channel 'SIP/pabx-e1-0252' [Mar 26 12:11:20] == Using SIP RTP TOS bits 184 [Mar 26 12:11:20] == Using SIP RTP CoS mark 5 [Mar 26 12:11:20] -- Executing [7003@ddi:1] Macro(SIP/7051-0254, stdexten,7003,SIP/7003) in new stack [Mar 26 12:11:20] -- Executing [s@macro-stdexten:1] NoOp(SIP/7051-0254, STDEXTEN: Arg1 = 7003 Arg2 = SIP/7003 Arg3 = ) in new stack [Mar 26 12:11:20] -- Executing [s@macro-stdexten:2] Dial(SIP/7051-0254, SIP/7003,45,tT) in new stack [Mar 26 12:11:20] == Using SIP RTP TOS bits 184 [Mar 26 12:11:20] == Using SIP RTP CoS mark 5 [Mar 26 12:11:20] -- Called SIP/7003 [Mar 26 12:11:20] -- SIP/7003-0255 is ringing [Mar 26 12:11:25] -- SIP/7003-0255 answered SIP/7051-0254 [Mar 26 12:11:25] -- Channel SIP/7051-0254 joined 'simple_bridge' basic-bridge f4fb9d99-24b9-4d3c-9b63-41a1b84484b2 [Mar 26 12:11:25] -- Channel SIP/7003-0255 joined 'simple_bridge' basic-bridge f4fb9d99-24b9-4d3c-9b63-41a1b84484b2 Then, extension 7051 transfers the call to 7003, who hangs up after a few seconds: [Mar 26 12:11:32] -- Channel SIP/pabx-e1-0252 left 'simple_bridge' basic-bridge b1c97b75-bd5f-4762-96dd-7aa68c472827 [Mar 26 12:11:32] -- Channel SIP/7051-0254 left 'simple_bridge' basic-bridge f4fb9d99-24b9-4d3c-9b63-41a1b84484b2 [Mar 26 12:11:32] -- Channel SIP/pabx-e1-0252 swapped with SIP/7051-0254 into 'simple_bridge' basic-bridge f4fb9d99-24b9-4d3c-9b63-41a1b84484b2 [Mar 26 12:11:32] -- Stopped music on hold on SIP/pabx-e1-0252 [Mar 26 12:11:32] -- Channel SIP/7051-0253 left 'simple_bridge' basic-bridge b1c97b75-bd5f-4762-96dd-7aa68c472827 [Mar 26 12:11:32] == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'SIP/7051-0254' in macro 'stdexten' [Mar 26 12:11:32] == Spawn extension (ddi, 7003, 1) exited non-zero on 'SIP/7051-0254' [2015-03-26 12:11:32] WARNING[1561][C-015c]: channel.c:5070 ast_write: Codec mismatch on channel SIP/pabx-e1-0252 setting write format to slin from alaw native formats (alaw) [Mar 26 12:11:40] -- Channel SIP/pabx-e1-0252 left 'simple_bridge' basic-bridge f4fb9d99-24b9-4d3c-9b63-41a1b84484b2 [Mar 26 12:11:40] == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'SIP/pabx-e1-0252' in macro 'stdexten' [Mar 26 12:11:40] == Spawn extension (interno, 7051, 1) exited non-zero on 'SIP/pabx-e1-0252' [Mar 26 12:11:40] -- Channel SIP/7003-0255 left 'simple_bridge' basic-bridge f4fb9d99-24b9-4d3c-9b63-41a1b84484b2 So far so good, except that when I check the CDR lines generated, this is what I get: mysql select calldate, uniqueid, linkedid, sequence, src, dst, duration, disposition, channel, dstchannel from cdr where uniqueid = '1427382664.963'; +-+++--++--+--+-+--+---+ | calldate| uniqueid | linkedid | sequence | src | dst | duration | disposition | channel | dstchannel|
[asterisk-users] Anonymous SIP calls
We have a FreePBX-12 / Asterisk-12 setup that supports about 24 extensions, most internal Snom870s but six or so external (Jitsi-2.8). we use TLS and SRTP everywhere on our side of the fence. The server host is a dedicated atom(tm) box using the FreePBX distro (CentOS-6.x) and is up-to-date. Registrations require very long random passwords and registrable devices are further restricted by netblock filters. We have the usual firewall and fail2ban intrusion prevention and detection set-ups in place. Our connection to the rest of the world is via PSTN. We do our own DNS, both forward and reverse. We have NAPTR and SRV RRs for SIP and SIPS. That is the environment. Now for the questions. Can I safely configure FreePBX/Asterisk to allow people to call us directly via SIP? In other words, sip://someth...@harte-lyne.ca would reach us and ring internally as if someone had called our main office number via PSTN. Does it make sense to do so? I am not talking about routing our main number through a SIP trunk provider. We will remain on PSTN for the foreseeable future. But I am curious as to whether or not it it worthwhile to allow others who have the capability to simply call us via SIP rather than over PSTN. And if we do allow it what are the caveats and how does one actually configure Asterisk to do it? I have read a number of blogs, sections of the Definitive Asterisk book and mailing list archived posts respecting anonymous SIP calls. But I have to say these leave me rather more confused than informed. Virtually all sources advise against accepting any anonymous incoming SIP calls whatsoever. The few that do not absolutely advise against do not give much guidance in how to handle incoming calls. And frankly, I have only a dim idea how an incoming SIP call should be handled from a theoretical point of view. Any guidance would be welcome. -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial to PJSIP Channel with Typo PJSIP// Causes Asterisk Shutdown
I found an issue with how PJSIP handles a typo in the Dial application. If the Channel is mistakenly typed with two slashes (i.e Dial(PJSIP//...), the Dial applications fails (obviously), but it also kills the server. I put some code in my pbx_config to check for that string and not let the dialplan reload, but it seems like there should be a better way to handle in in the PJSIP stack or Dial app so that it doesn't take the server down if it gets through. I am not a developer, but I was hoping maybe someone who monitors this mailing list might feel like taking this on as a bug fix.I haven't tried with any other channel drivers, so it may cross to others. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial to PJSIP Channel with Typo PJSIP// Causes Asterisk Shutdown
On Thu, Mar 26, 2015 at 9:28 AM, Trey Hilyard kct...@gmail.com wrote: I found an issue with how PJSIP handles a typo in the Dial application. If the Channel is mistakenly typed with two slashes (i.e Dial(PJSIP//...), the Dial applications fails (obviously), but it also kills the server. I put some code in my pbx_config to check for that string and not let the dialplan reload, but it seems like there should be a better way to handle in in the PJSIP stack or Dial app so that it doesn't take the server down if it gets through. I am not a developer, but I was hoping maybe someone who monitors this mailing list might feel like taking this on as a bug fix.I haven't tried with any other channel drivers, so it may cross to others. Please open an issue on the issue tracker: https://issues.asterisk.org/jira A backtrace from the crash will be needed as well. Instructions on generating a backtrace can be found on the wiki here: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR dst value null after attended transfer
I'm having an issue with CDR. Basically, I expect to have all legs of a call having the same linkedid and differing only by the sequence value. That does happen, but I'm getting null dst values after doing an attended transfer. I'm not sure if this is a bug or I'm doing something wrong. I'm running Asterisk 13.2.0. Here's the console log, step by step: First, I receive a call from 5491549116 on extension 7051 (DID 5421047051): [Mar 26 12:11:04] == Using SIP RTP TOS bits 184 [Mar 26 12:11:04] == Using SIP RTP CoS mark 5 [Mar 26 12:11:04] -- Executing [5421047051@restrito:1] Goto(SIP/pabx-e1-0252, interno,7051,1) in new stack [Mar 26 12:11:04] -- Goto (interno,7051,1) [Mar 26 12:11:04] -- Executing [7051@interno:1] Macro(SIP/pabx-e1-0252, stdexten,7051,SIP/7051) in new stack [Mar 26 12:11:04] -- Executing [s@macro-stdexten:1] NoOp(SIP/pabx-e1-0252, STDEXTEN: Arg1 = 7051 Arg2 = SIP/7051 Arg3 = ) in new stack [Mar 26 12:11:04] -- Executing [s@macro-stdexten:2] Dial(SIP/pabx-e1-0252, SIP/7051,45,tT) in new stack [Mar 26 12:11:04] == Using SIP RTP TOS bits 184 [Mar 26 12:11:04] == Using SIP RTP CoS mark 5 [Mar 26 12:11:04] -- Called SIP/7051 [Mar 26 12:11:05] -- SIP/7051-0253 is ringing [Mar 26 12:11:11] -- SIP/7051-0253 answered SIP/pabx-e1-0252 [Mar 26 12:11:11] -- Channel SIP/pabx-e1-0252 joined 'simple_bridge' basic-bridge b1c97b75-bd5f-4762-96dd-7aa68c472827 [Mar 26 12:11:11] -- Channel SIP/7051-0253 joined 'simple_bridge' basic-bridge b1c97b75-bd5f-4762-96dd-7aa68c472827 Now, extension 7051 places the call on hold and calls 7003, who answers: [Mar 26 12:11:17] -- Started music on hold, class 'default', on channel 'SIP/pabx-e1-0252' [Mar 26 12:11:20] == Using SIP RTP TOS bits 184 [Mar 26 12:11:20] == Using SIP RTP CoS mark 5 [Mar 26 12:11:20] -- Executing [7003@ddi:1] Macro(SIP/7051-0254, stdexten,7003,SIP/7003) in new stack [Mar 26 12:11:20] -- Executing [s@macro-stdexten:1] NoOp(SIP/7051-0254, STDEXTEN: Arg1 = 7003 Arg2 = SIP/7003 Arg3 = ) in new stack [Mar 26 12:11:20] -- Executing [s@macro-stdexten:2] Dial(SIP/7051-0254, SIP/7003,45,tT) in new stack [Mar 26 12:11:20] == Using SIP RTP TOS bits 184 [Mar 26 12:11:20] == Using SIP RTP CoS mark 5 [Mar 26 12:11:20] -- Called SIP/7003 [Mar 26 12:11:20] -- SIP/7003-0255 is ringing [Mar 26 12:11:25] -- SIP/7003-0255 answered SIP/7051-0254 [Mar 26 12:11:25] -- Channel SIP/7051-0254 joined 'simple_bridge' basic-bridge f4fb9d99-24b9-4d3c-9b63-41a1b84484b2 [Mar 26 12:11:25] -- Channel SIP/7003-0255 joined 'simple_bridge' basic-bridge f4fb9d99-24b9-4d3c-9b63-41a1b84484b2 Then, extension 7051 transfers the call to 7003, who hangs up after a few seconds: [Mar 26 12:11:32] -- Channel SIP/pabx-e1-0252 left 'simple_bridge' basic-bridge b1c97b75-bd5f-4762-96dd-7aa68c472827 [Mar 26 12:11:32] -- Channel SIP/7051-0254 left 'simple_bridge' basic-bridge f4fb9d99-24b9-4d3c-9b63-41a1b84484b2 [Mar 26 12:11:32] -- Channel SIP/pabx-e1-0252 swapped with SIP/7051-0254 into 'simple_bridge' basic-bridge f4fb9d99-24b9-4d3c-9b63-41a1b84484b2 [Mar 26 12:11:32] -- Stopped music on hold on SIP/pabx-e1-0252 [Mar 26 12:11:32] -- Channel SIP/7051-0253 left 'simple_bridge' basic-bridge b1c97b75-bd5f-4762-96dd-7aa68c472827 [Mar 26 12:11:32] == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'SIP/7051-0254' in macro 'stdexten' [Mar 26 12:11:32] == Spawn extension (ddi, 7003, 1) exited non-zero on 'SIP/7051-0254' [2015-03-26 12:11:32] WARNING[1561][C-015c]: channel.c:5070 ast_write: Codec mismatch on channel SIP/pabx-e1-0252 setting write format to slin from alaw native formats (alaw) [Mar 26 12:11:40] -- Channel SIP/pabx-e1-0252 left 'simple_bridge' basic-bridge f4fb9d99-24b9-4d3c-9b63-41a1b84484b2 [Mar 26 12:11:40] == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'SIP/pabx-e1-0252' in macro 'stdexten' [Mar 26 12:11:40] == Spawn extension (interno, 7051, 1) exited non-zero on 'SIP/pabx-e1-0252' [Mar 26 12:11:40] -- Channel SIP/7003-0255 left 'simple_bridge' basic-bridge f4fb9d99-24b9-4d3c-9b63-41a1b84484b2 So far so good, except that when I check the CDR lines generated, this is what I get: mysql select calldate, uniqueid, linkedid, sequence, src, dst, duration, disposition, channel, dstchannel from cdr where uniqueid = '1427382664.963'; +-+++--++--+--+-+--+---+ | calldate| uniqueid | linkedid | sequence | src | dst | duration | disposition | channel | dstchannel | +-+++--++--+--+-+--+---+ |
