Showing posts with label freepbx. Show all posts
Showing posts with label freepbx. Show all posts

Thursday, May 02, 2013

FreePBX Voicemail Drops Call With Error: lack of rtp activity in 31 seconds

I have got FreePBX setup @home and for some reasons my voicemail is not working properly. It basically drops the connection after 30 seconds while the person on the phone is waiting on Music on Hold.

It turns out the FreePBX detects there was no audio/RTP activity within 30 seconds (configurable) and drops the connection.

To change this: on your FreePBX, navigate to: Settings > Asterisk SIP Settings > Media and RTP Settings
Change the rtptimeout from 30 to 300, change rtpkeepalive from 0 to 30

Sunday, March 31, 2013

FreePBX SIP Debugging

To debug FreePBX SIP, just get into the asterisk context by typing:

> asterisk -vvvvvr

localhost*CLI> sip show peers

it shows all your peers, then:

localhost*CLI> sip set debug peer (peer_name)

To stop debug, type:

localhost*CLI> sip set debug off