SIP Registering. Rings External Phone. No Audio/Sound
FreePBX 12.0.76.2
Asterisk 11.19.0
Linux 2.6.32-431.el6.i686
I am using SIP Station trunks. The sips are registering.
I am able to ring my cell phone from the handset But there is no audio from the handset
The SIP trunks are in use on a second server and working fine. This PBX Server is running version FreePBX 2.9.0.14
I copied the trunk data and have changed a few things in my updated server
I have not proceeded to incoming routes yet.
I have configured a Flowroute trunk and have it working on the updated server fine
some errors and warnings are in the log file
[2015-10-06 10:37:36] WARNING[9304] pbx.c: Context ‘from-internal-xfer’ tries to include nonexistent context ‘from-internal-custom’
[2015-10-06 10:37:36] WARNING[9304] pbx.c: Context ‘from-internal-noxfer’ tries to include nonexistent context ‘from-internal-noxfer-custom’
[2015-10-06 10:37:36] WARNING[9304] pbx.c: Context ‘from-pstn’ tries to include nonexistent context ‘from-pstn-custom’
[2015-10-06 10:37:36] WARNING[9304] pbx.c: Context ‘from-internal-additional’ tries to include nonexistent context ‘ext-meetme’
[2015-10-06 10:37:36] ERROR[9304] res_clialiases.c: res_clialiases configuration file ‘cli_aliases.conf’ not found
It’s actually quite peculiar that Flowroute work and another provider fails since with Flowroute you are never sure where the RTP packets will come from…
The communication with port 5060 is established with known servers but the RTP packets comes from other servers which are not documented.
I actually had to change my firewall rules to deal with Flowroute…
Well, I finally beat the issue… Got some tech support for the sip trunks from schmooze. Have to use the SIPSTATION module (it is commercial I assume it comes with our monthly fee) Once you have a key code from them you enter the key and whalla! Trunks and Routes are set up automatically. I am not sure why my manually entered trunks and routes didn’t work.
SIP provides 2 trunk with their service primary and secondary. They can only work together on one server. I was trying to disable Trunk 2 on in use Server and trying to get outbounds going with the second trunk on a second server. Schmooze says it is not allowed.
I did have my external IP set to something other than the external IP address my second server was on. Once we pushed the detect external IP in Asterisk SIP settings the functionality was there.
Moral of the story… if you are paying someone for a service get in touch with them ASAP. I spent many hours trying to figure it out chasing dead ends.