[asterisk-users] Answered call not bridged
Hi I've suddenly started encountering a strange issue. Sometimes, when a call is made into our system, an extension answered the phone but I can see no mention of it being bridged in the console. Also, the server does not seem to think that it is answered and then goes to voicemail. We are using asterisk 1.4.17 Here is the console output for one of these calls, it was me ringing a customer complaining about the issue [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Goto(SIP/PACK501-480b08c0, default|xxx|1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Goto (default,02034684373,1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Goto(SIP/PACK501-480b08c0, enge-xx|s|1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Goto (enge-02034684373,s,1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing NoOp(SIP/PACK501-480b08c0, ) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Wait(SIP/PACK501-480b08c0, 2) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Executing Set(SIP/PACK501-480b08c0, CALLERID(num)=PACK501) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Executing Dial(SIP/PACK501-480b08c0, SIP/ENGE103|20) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Called ENGE103 [2010-07-28 11:07:28] VERBOSE[6554] logger.c: -- SIP/ENGE103-009140e0 is ringing *** AT this point the customer had answered and I was talking to him!! [2010-07-28 11:07:28] VERBOSE[6554] logger.c: -- SIP/ENGE103-009140e0 is ringing [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- Nobody picked up in 2 ms [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- Executing Voicemail(SIP/PACK501-480b08c0, 1...@enge-local|u) [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-theperson' (language 'en') [2010-07-28 11:07:50] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/1' (language 'en') [2010-07-28 11:07:50] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/0' (language 'en') [2010-07-28 11:07:51] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/3' (language 'en') [2010-07-28 11:07:52] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-isunavail' (language 'en') [2010-07-28 11:07:53] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-intro' (language 'en') [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'beep' (language 'en') [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- Recording the message [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=0, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: wav49, 0xb75e60 [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=1, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: gsm, 0xb20720 [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=2, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: wav, 0xa1c850 [2010-07-28 11:08:00] VERBOSE[6554] logger.c: -- User hung up [2010-07-28 11:08:00] VERBOSE[6554] logger.c: == Spawn extension (enge-02034684373, s, 5) exited non-zero on 'SIP/PACK501-480b08c0' The customer is using Aastra phones but it's happened once with us when I was using a Snom phone. I'm trying to consistently replicate the issue so that I can analyse it properly but have not been able to so far. Has anyone ever experienced anything like this? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Answered call not bridged
If you run a sip debug at the same time you will get some more usefull logs. What sip client are you using? Ishfaq Malik wrote: Hi I've suddenly started encountering a strange issue. Sometimes, when a call is made into our system, an extension answered the phone but I can see no mention of it being bridged in the console. Also, the server does not seem to think that it is answered and then goes to voicemail. We are using asterisk 1.4.17 Here is the console output for one of these calls, it was me ringing a customer complaining about the issue [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Goto(SIP/PACK501-480b08c0, default|xxx|1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Goto (default,02034684373,1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Goto(SIP/PACK501-480b08c0, enge-xx|s|1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Goto (enge-02034684373,s,1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing NoOp(SIP/PACK501-480b08c0, ) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Wait(SIP/PACK501-480b08c0, 2) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Executing Set(SIP/PACK501-480b08c0, CALLERID(num)=PACK501) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Executing Dial(SIP/PACK501-480b08c0, SIP/ENGE103|20) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Called ENGE103 [2010-07-28 11:07:28] VERBOSE[6554] logger.c: -- SIP/ENGE103-009140e0 is ringing *** AT this point the customer had answered and I was talking to him!! [2010-07-28 11:07:28] VERBOSE[6554] logger.c: -- SIP/ENGE103-009140e0 is ringing [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- Nobody picked up in 2 ms [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- Executing Voicemail(SIP/PACK501-480b08c0, 1...@enge-local|u) [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-theperson' (language 'en') [2010-07-28 11:07:50] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/1' (language 'en') [2010-07-28 11:07:50] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/0' (language 'en') [2010-07-28 11:07:51] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/3' (language 'en') [2010-07-28 11:07:52] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-isunavail' (language 'en') [2010-07-28 11:07:53] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-intro' (language 'en') [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'beep' (language 'en') [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- Recording the message [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=0, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: wav49, 0xb75e60 [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=1, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: gsm, 0xb20720 [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=2, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: wav, 0xa1c850 [2010-07-28 11:08:00] VERBOSE[6554] logger.c: -- User hung up [2010-07-28 11:08:00] VERBOSE[6554] logger.c: == Spawn extension (enge-02034684373, s, 5) exited non-zero on 'SIP/PACK501-480b08c0' The customer is using Aastra phones but it's happened once with us when I was using a Snom phone. I'm trying to consistently replicate the issue so that I can analyse it properly but have not been able to so far. Has anyone ever experienced anything like this? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Answered call not bridged
On receiving a call, try using the 'Answer()' command before anything else. I once had some issues, though not similar, which were solved by this command, as it sends back a SIP acknowledgement to the calling party which is otherwise not sent. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-28 6:30 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi I've suddenly started encountering a strange issue. Sometimes, when a call is made into our system, an extension answered the phone but I can see no mention of it being bridged in the console. Also, the server does not seem to think that it is answered and then goes to voicemail. We are using asterisk 1.4.17 Here is the console output for one of these calls, it was me ringing a customer complaining about the issue [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Goto(SIP/PACK501-480b08c0, default|xxx|1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Goto (default,02034684373,1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Goto(SIP/PACK501-480b08c0, enge-xx|s|1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Goto (enge-02034684373,s,1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing NoOp(SIP/PACK501-480b08c0, ) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Wait(SIP/PACK501-480b08c0, 2) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Executing Set(SIP/PACK501-480b08c0, CALLERID(num)=PACK501) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Executing Dial(SIP/PACK501-480b08c0, SIP/ENGE103|20) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Called ENGE103 [2010-07-28 11:07:28] VERBOSE[6554] logger.c: -- SIP/ENGE103-009140e0 is ringing *** AT this point the customer had answered and I was talking to him!! [2010-07-28 11:07:28] VERBOSE[6554] logger.c: -- SIP/ENGE103-009140e0 is ringing [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- Nobody picked up in 2 ms [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- Executing Voicemail(SIP/PACK501-480b08c0, 1...@enge-local|u) [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-theperson' (language 'en') [2010-07-28 11:07:50] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/1' (language 'en') [2010-07-28 11:07:50] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/0' (language 'en') [2010-07-28 11:07:51] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/3' (language 'en') [2010-07-28 11:07:52] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-isunavail' (language 'en') [2010-07-28 11:07:53] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-intro' (language 'en') [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'beep' (language 'en') [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- Recording the message [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=0, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: wav49, 0xb75e60 [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=1, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: gsm, 0xb20720 [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=2, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: wav, 0xa1c850 [2010-07-28 11:08:00] VERBOSE[6554] logger.c: -- User hung up [2010-07-28 11:08:00] VERBOSE[6554] logger.c: == Spawn extension (enge-02034684373, s, 5) exited non-zero on 'SIP/PACK501-480b08c0' The customer is using Aastra phones but it's happened once with us when I was using a Snom phone. I'm trying to consistently replicate the issue so that I can analyse it properly but have not been able to so far. Has anyone ever experienced anything like this? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Answered call not bridged
Hi The problem is that this is a production server with usually about 10 concurrent calls going on and also that if I just run a sip debug on the customers peer, I still don't know when it's either this issue or if it genuinely went to voicemail. That's why I'm trying to consistently replicate the issue so I can do a controlled sip debug on it :( They are using Aastra 51i/2.1.0.2145 Thanks Ish On 28/07/10 11:47, Gareth Blades wrote: If you run a sip debug at the same time you will get some more usefull logs. What sip client are you using? Ishfaq Malik wrote: Hi I've suddenly started encountering a strange issue. Sometimes, when a call is made into our system, an extension answered the phone but I can see no mention of it being bridged in the console. Also, the server does not seem to think that it is answered and then goes to voicemail. We are using asterisk 1.4.17 Here is the console output for one of these calls, it was me ringing a customer complaining about the issue [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Goto(SIP/PACK501-480b08c0, default|xxx|1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Goto (default,02034684373,1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Goto(SIP/PACK501-480b08c0, enge-xx|s|1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Goto (enge-02034684373,s,1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing NoOp(SIP/PACK501-480b08c0, ) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Wait(SIP/PACK501-480b08c0, 2) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Executing Set(SIP/PACK501-480b08c0, CALLERID(num)=PACK501) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Executing Dial(SIP/PACK501-480b08c0, SIP/ENGE103|20) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Called ENGE103 [2010-07-28 11:07:28] VERBOSE[6554] logger.c: -- SIP/ENGE103-009140e0 is ringing *** AT this point the customer had answered and I was talking to him!! [2010-07-28 11:07:28] VERBOSE[6554] logger.c: -- SIP/ENGE103-009140e0 is ringing [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- Nobody picked up in 2 ms [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- Executing Voicemail(SIP/PACK501-480b08c0, 1...@enge-local|u) [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-theperson' (language 'en') [2010-07-28 11:07:50] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/1' (language 'en') [2010-07-28 11:07:50] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/0' (language 'en') [2010-07-28 11:07:51] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/3' (language 'en') [2010-07-28 11:07:52] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-isunavail' (language 'en') [2010-07-28 11:07:53] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-intro' (language 'en') [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'beep' (language 'en') [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- Recording the message [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=0, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: wav49, 0xb75e60 [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=1, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: gsm, 0xb20720 [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=2, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: wav, 0xa1c850 [2010-07-28 11:08:00] VERBOSE[6554] logger.c: -- User hung up [2010-07-28 11:08:00] VERBOSE[6554] logger.c: == Spawn extension (enge-02034684373, s, 5) exited non-zero on 'SIP/PACK501-480b08c0' The customer is using Aastra phones but it's happened once with us when I was using a Snom phone. I'm trying to consistently replicate the issue so that I can analyse it properly but have not been able to so far. Has anyone ever experienced anything like this? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Answered call not bridged
Hi Unfortunately this isn't an option as we allow customers to forward incoming calls back out to POTS or mobile. If we use an explicit Answer() all forwarded calls show as answered even if they weren't by the POTS or mobile end point. Ish On 28/07/10 11:48, Zeeshan Zakaria wrote: On receiving a call, try using the 'Answer()' command before anything else. I once had some issues, though not similar, which were solved by this command, as it sends back a SIP acknowledgement to the calling party which is otherwise not sent. Zeeshan A Zakaria -- www.ilovetovoip.com http://www.ilovetovoip.com On 2010-07-28 6:30 AM, Ishfaq Malik i...@pack-net.co.uk mailto:i...@pack-net.co.uk wrote: Hi I've suddenly started encountering a strange issue. Sometimes, when a call is made into our system, an extension answered the phone but I can see no mention of it being bridged in the console. Also, the server does not seem to think that it is answered and then goes to voicemail. We are using asterisk 1.4.17 Here is the console output for one of these calls, it was me ringing a customer complaining about the issue [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Goto(SIP/PACK501-480b08c0, default|xxx|1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Goto (default,02034684373,1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Goto(SIP/PACK501-480b08c0, enge-xx|s|1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Goto (enge-02034684373,s,1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing NoOp(SIP/PACK501-480b08c0, ) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Wait(SIP/PACK501-480b08c0, 2) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Executing Set(SIP/PACK501-480b08c0, CALLERID(num)=PACK501) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Executing Dial(SIP/PACK501-480b08c0, SIP/ENGE103|20) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Called ENGE103 [2010-07-28 11:07:28] VERBOSE[6554] logger.c: -- SIP/ENGE103-009140e0 is ringing *** AT this point the customer had answered and I was talking to him!! [2010-07-28 11:07:28] VERBOSE[6554] logger.c: -- SIP/ENGE103-009140e0 is ringing [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- Nobody picked up in 2 ms [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- Executing Voicemail(SIP/PACK501-480b08c0, 1...@enge-local|u) [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-theperson' (language 'en') [2010-07-28 11:07:50] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/1' (language 'en') [2010-07-28 11:07:50] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/0' (language 'en') [2010-07-28 11:07:51] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/3' (language 'en') [2010-07-28 11:07:52] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-isunavail' (language 'en') [2010-07-28 11:07:53] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-intro' (language 'en') [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'beep' (language 'en') [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- Recording the message [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=0, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: wav49, 0xb75e60 [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=1, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: gsm, 0xb20720 [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=2, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: wav, 0xa1c850 [2010-07-28 11:08:00] VERBOSE[6554] logger.c: -- User hung up [2010-07-28 11:08:00] VERBOSE[6554] logger.c: == Spawn extension (enge-02034684373, s, 5) exited non-zero on 'SIP/PACK501-480b08c0' The customer is using Aastra phones but it's happened once with us when I was using a Snom phone. I'm trying to consistently replicate the issue so that I can analyse it properly but have not been able to so far. Has anyone ever experienced anything like this? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
Re: [asterisk-users] Answered call not bridged
Hi! - upgrade to a current 1.4 version, 1.4.17 is very old (you probably run this because of the zaptel -- dahdi change, but still) - do you have a SIP proxy or any SIP-aware hardware in your network that might play tricks on you, e.g. a SIP ALG (application layer gateway) on your Internet router or something similar? - enable SIP debugging on your phone and check its logs; you could also do a packet capture on your router to see what exactly is happening and if Asterisk is somehow being cut out of the loop - see if canreinvite=no somehow helps; disable STUN on your phone inside the LAN, and maybe even block direct Internet traffic for your LAN phones so that they must go through Asterisk. Philipp -- _ -- 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
