Corrupt MixMonitor Recordings – .gsm Format
Hi all
Asterisk 1.8.11.0 on CentOS 6.5
My VOIP phones are using G729 to a G729 trunk from a vendor (Centracom, South Africa). Unlicensed G729 codec version on server.
75% of my .gsm files from MixMonitor are coming up corrupted about 3 minutes into the recording.
The server has been up for 7 months beforehand with no problems with recordings to .gsm format files.
I noted warnings and errors in the CLI, apparently coinceding with corrupt MixMonitor recordings:
format_gsm.c:65 gsm_read: Short read (13) (Resource temporarily unavailable)!
WARNING[30727]: app_dial.c:1379 wait_for_answer: Unable to write frametype:
2
WARNING[14712]: file.c:766 ast_readaudio_callback: Failed to write frame
WARNING[2612]: file.c:766 ast_readaudio_callback: Failed to write frame
WARNING[25283]: format_gsm.c:65 gsm_read: Short read (32) (Resource temporarily unavailable)!
WARNING[28804]: file.c:766 ast_readaudio_callback: Failed to write frame
WARNING[28804]: file.c:766 ast_readaudio_callback: Failed to write frame
I’m using 1.8.11.0 on 14 sites countrywide for 3 years now, recording about
80 000 .gsm recordings per day.
The actual VOIP audio is fine, the callers don’t have any problems actually talking to people, it is just the recording that is corrupt, the conversation itself is fine.
Only one site has started producing corrupt .gsm files since last week. I’ve already replaced that server with a brand new one, reinstalled the operating system and Asterisk, problem still persists.
I’ve extensively searched online but nobody seems to have ever experienced massive .gsm files corruption from MixMonitor – any ideas where I can even start to look to solve this?
Thank you
One thought on - Corrupt MixMonitor Recordings – .gsm Format
What changed? It is unlikely that Asterisk suddenly changed without outside intervention, unless there was some slow leak and a resource limitation was reached. Your troubleshooting below would have at least temporarily resolved that issue.
If it is only the one site, and replacing Asterisk AND the entire machine with OS did not fix the issue, then I’d think something changed at the site outside your Asterisk server. I would look into what is different between VOIP traffic you are receiving on that site and other sites. Though, I really wouldn’t know what to look for in this case.
The only recent MixMonitor audio corruption issue I found reported was https://issues.asterisk.org/jira/browse/ASTERISK-24507 and it doesn’t match what you are describing here, unless maybe you only do blind transfers on this site, and no other sites and you only started having the blind transfers 7 months into usage.