Voicemail cutting off at 10 seconds

[cylon21]

Hello! I’m running FreePBX 15.0.37.4. I figured out how to set up voicemail but it cuts off the recording at 10 seconds. In other words, as the user is leaving a voicemail, it hangs up at 10 seconds. I looked through this: Comedian Mail
per another posting, I adjusted:

voicemail.conf

• maxsecs = 300
• maxsilence = 120

None of this has worked. Is there anything that can be done to prevent it from hanging up after 10 seconds of recording a voicemail?

[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.

[cylon21]

I added the [options] section (it wasn’t already in the config file) then added the transmit string you mentioned. I then restarted it. Unfortuntely it still hangs up at 10 seconds.

I look at the thread you referenced… they mentioned:

" Try to set RTP keepalive on the sip channels to one second. it sends a CNG (comfort noise generation) frame to keep the session alive especially in NAT environments.

The name of the setting for the chan_sip driver is:
rtpkeepalive=1

and for chan_pjsip is:
rtp_keepalive=1

Where would rtp_keepalive=1 go?

[Stewart1 (Stewart)]

In Asterisk SIP Settings, General tab, you should find an RTP Keep Alive setting.