reply to Stewart
Re: RTP/NAT router, how to configure PAP2T/FreePBX?
said by Stewart:This is a subject that interests me quite a bit. I have been using G722 codec with Asterisk for a couple years now. It sounds great, even if it is only available between my extensions and the one other person I call with a SIP phone.
If no luck, look on the CLI (with verbosity 3) for "Remotely bridging ..." If you don't see that, track down why it's not attempting to re-invite. If you do see it, use SIP debug to see why the re-invite is being rejected by the extension or trunk.
Using Asterisk CLI debug on a call between extensions "1" and "2" I see:
-- Called 1
-- SIP/1-0000005e is ringing
-- SIP/1-0000005e answered SIP/2-0000005d
-- Remotely bridging SIP/2-0000005d and SIP/1-0000005e
So this means Asterisk is not proxying the media? I would like to try SILK-WB codec, which Asterisk does not support. But my phones, Bria for Android, do support this codec. I had thought Asterisk support for the codec would not be needed. Am I wrong?
I can not get SILK codec to work between these extensions.
Edit: I see that Asterisk will not even attempt a connection between the two extensions when SILK is set as the only codec. I read somewhere there was a fix to Asterisk for this issue . . . in Asterisk 10 and 11. I guess it will be time to upgrade soon.