We recently came across a hair pulling situation where one of our PBXs stopped sending RTP as soon as it started recording voicemails and eventually terminated the call after 30 seconds, but it only happened when we called from a PBX in the same DataCenter.
No NAT issues, no Direct RTP, we've gone through the basic and advanced troubleshooting with no luck.
We added it under SIP Settings > Custom, restarted Asterisk and it started sending RTP.
You can see the setting if it is enabled or disabled by running 'core show settings'
So I hope someone who is in the same boat as us won't have to go through the pain we went through, and finds this post easily..
Perhaps the VitalPBX Team can make this the default, I see other major Asterisk re-compilers also have this enabled by default.
Just would like to take feedback that after making changes to transmit_silence=yes how is recording and calls working since.
Anything you would like to add?
UPDATE: After looking at this, adding these parameters in the GUI does not even work, because it adds it under the [general] settings in /etc/asterisk/ombutel/sip__10-general.conf
It needs to be added under [options].
I found under /etc/asterisk/ombutel/asterisk__20-maxfiles.conf:
maxfiles = 20000
So I manually added:
And I restarted Asterisk.
1) Is it possible to add these from the GUI?
2) Is it possible to have this be the default?
I think this parameter doesn't come by default because it is not used often. it's a very particular situation what is happening to you