Voicemail cutting off at 10 seconds

[Stewart1 (Stewart)]

In /etc/asterisk/asterisk.conf, under [options]
transmit_silence_during_record=yes
If you change this, restart Asterisk.

If you already had this setting, or it doesn’t help, see

RTP Issues with ChanSIP

Hey everyone, I know the correct way of dealing with this issue is to migrate to PJSIP, but at the time of writing this, we cannot fully migrate them just yet. ChanSIP=5060 PJSIP=5062, most phones are already on PJSIP, however, the Trunk is still ChanSIP (provider limitation to 5060 only) Client has one phone number they started using recently to record certain messages. They are complaining that after being on the phone for a while during recording a message, the call drops. Looking at the l…

If you still have trouble, report whether this problem also occurs on internal calls.