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|xxxxxxxxxxx|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-xxxxxxxxxx|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 20000 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