SIP Registering. Rings External Phone. No Audio/Sound

SIP Registering. Rings External Phone. No Audio/Sound

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
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

Peer details for the non working trunks are:


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

[2015-10-06 10:37:36] ERROR[9304] res_config_ldap.c: Cannot load configuration file: res_ldap.conf
[2015-10-06 10:37:36] NOTICE[9304] res_config_ldap.c: Cannot reload LDAP RealTime driver.

[2015-10-06 10:37:36] ERROR[9304] res_config_sqlite3.c: Missing config file ‘res_config_sqlite3.conf’

[2015-10-06 10:37:36] ERROR[9304] phone_message.c: Unable to build dialplan routing - invalid license
[2015-10-06 10:37:36] WARNING[9304] res_digium_phone.c: No Valid DPMA License found. Module is loaded but disabled. Please reload module once valid license is installed.

Thanks of anyone can help with this issue. I have been unable to find many threads related to the issue.

I am suspecting that the RTP port being used may be blocked by Comcast or interference of some sort

Calling a verizon cell phone I hear no return ringing and the handset is dead for audio

Calling my comcast home phone I do hear the phone ring on the other end but there is no audio once the call is connected

I called a local company and got the same result as the verison cell phone

It seems signals are getting through on port 5060, but voice is not getting though on an RTP between 10000 and 20000

Does anyone know of any conflicts that SIPSTATION has with Comcast who is our ISP Here??

Thanks if you can help

I don’t follow all of this but this seems to be related to the issue

Please help

Why does Flowroute work and SIP Station doesn’t?


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…

(I didn’t put any ACL on my PBX’s RTP ports…)

Have a nice day!


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.