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Summary:ASTERISK-11082: Voicemail cuts off at 60 seconds regardless of config settings
Reporter:Arcadiy Ivanov (arcivanov)Labels:
Date Opened:2007-12-19 09:29:59.000-0600Date Closed:2011-06-07 14:00:34
Priority:BlockerRegression?No
Status:Closed/CompleteComponents: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