Voicemails sometimes cut off at 30 seconds

First time poster here. I am currently deal with a rather annoying issue that has taken up much of my time. For some reason, one of my deployments of FreePBX started to intermittently cut off random voicemails @ about the 30 second mark. It would cut people off mid sentence, and they would need to call back just to finish leaving the voicemail.

This had all started when I had upgraded their hardware based FreePBX 15 device up to FreePBX 17. I had virtualized the new device, and had done a config export/import to bring over most of the needed configuration.

So far, for troubleshooting, I have:

  • Modified the max voicemail time in the FreePBX config.
  • Swapping them back to a hardware-based FreePBX device.
  • Set up very verbose logging to monitor packets at the times of the issues. Which had resulted in me finding this:

Part of what makes this such a headache is the unpredictability as to when this would happen. I had tested calling an extension upwards of 20 times throughout a day to see if I could get it to act up, and every single one of my calls worked perfectly. I used 2 different phones, one AT&T and one Verizon.It just so happens that one random call had come in during the aforementioned troubleshooting session, and said individual did end up getting cut off mid-voicemail (I.E. the screenshot above).

I am currently investigating phone configs, and/or the possibility of building them an entire new PBX device built from the ground up, since up until this point I had only done backup restores.

If anyone has any experience with a similar issue, or has some advice for me, I would greatly appreciate it.

Edit /etc/asterisk/asterisk.conf
Uncommon this Value (delete the leading ; ) : transmit_silence_during_record=yes

fwconsole restart

Validate that your settings are good by reviewing
asterisk -x “core show settings”

Make sure your RTP Keep Alive settings in your Settings > Asterisk SIP Settings are configured as well for cases of one way RTP.

What’s happening is this

Caller is sending RTP
PBX is NOT sending RTP because it’s recording the call

The Calling Party is expecting RTP and not receiving RTP. A session timer expires on the Calling side and a hang up occurs.

I appreciate the assistance. Upon inspection, under /etc/asterisk/asterisk.conf, there is not field specifically named transmit_silence_during_record=yes. There IS just a basic transmit_silence=yes field though. Would this be the field you are currently recommending I modify?

Yep! Set it to yes.

Another solution, which I used before is adding “RTP-Comfort-Noise”.

Add a line with “rtp_keepalive=1” in the audio settings under the section where you also define outgoing calls. (in pjsip.conf)

Not sure which solution is the best :)?