Summary: | ASTERISK-11082: Voicemail cuts off at 60 seconds regardless of config settings | ||
Reporter: | Arcadiy Ivanov (arcivanov) | Labels: | |
Date Opened: | 2007-12-19 09:29:59.000-0600 | Date Closed: | 2011-06-07 14:00:34 |
Priority: | Blocker | Regression? | No |
Status: | Closed/Complete | Components: | Applications/app_voicemail |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ( 0) full_voice.txt ( 1) full_voicemail.txt ( 2) msg0005.WAV | |
Description: | Voicemail cuts off at 60 seconds regardless of voicemail.conf setting. Attached is the file exactly 1:00 min long cutting off. Source of the call doesn't matter. ****** ADDITIONAL INFORMATION ****** Asterisk verbose 9999: ---------------------------------------------------------- -- Executing [s@incoming-voicemail:12] VoiceMail("SIP/sTp57GEb98-08218510", "0001|s") in new stack -- <SIP/sTp57GEb98-08218510> Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/0001/tmp/5ZGWsZ format: wav49, 0x8217e10 == Refreshing DNS lookups. -- User hung up == Spawn extension (incoming-voicemail, s, 12) exited non-zero on 'SIP/sTp57GEb98-08218510' ---------------------------------------------------------- Voicemail.conf maxmessage=600 | ||
Comments: | By: Arcadiy Ivanov (arcivanov) 2007-12-19 09:34:09.000-0600 Changing maxmessage=600 to maxmessage=300 still results in 60 sec cutoff. Not a second to millisecond conversion issue. By: Jason Parker (jparker) 2007-12-19 10:50:30.000-0600 I assume this is being tested from a SIP phone? Can you get a SIP debug near the end, when the "User hung up" message would occur? Have you tried recording from IAX2 or zap channels? By: Joshua C. Colp (jcolp) 2007-12-19 10:52:43.000-0600 Complete console log with debug enabled in logger.conf would also be useful. By: Arcadiy Ivanov (arcivanov) 2007-12-22 18:07:11.000-0600 Attached are requested log files containing the results of three tests. Test 1 (full_voicemail.txt): Call Voicepulse Connect number A from mobile B, do not pick up, get connected to voicemail. Test 1 result: Reached voicemail, cutoff at exactly 60 seconds. Test 2 (full_voice.txt): Call Voicepulse Connect number A from mobile B, pick up, talk. Test 2 result: Talked for 120 seconds with no problems. Hangup. Test 3 (logs not included): Call Telphin number C from SkypeOut, do not pick up, get connected to voicemail. Test 3 result: Reached voicemail, left voicemail over 3 minutes long. Hangup. Conclusion: Looks like Voicepulse Connect-specific problem while recording voicemail: no keep-alives/pings/pongs are sent back? Voicepulse thinks * dropped the connection? Logs: Files have been cleaned up: nonces, usernames, phone numbers removed. By: Joshua C. Colp (jcolp) 2007-12-26 09:18:12.000-0600 Voicepulse Connect seems to have the rtptimeout option set to 60 seconds, and by default nothing is sent back when Asterisk is doing a recording. This can be changed so it sends back silence by setting transmit_silence_during_record to yes in the options context of asterisk.conf |