I have a large number of Asterisk systems in production and know that direct media has worked previously for me.
With recent development of my new release along with an upgrade to Asterisk 13, I cannot for the life of me get direct media to work. I am testing by just making a simple extension to extension call and doing an RTP debug from the Asterisk CLI.
The following is confirmed:
There are no Dial options in use
The codecs are the same as they use the same sip template
sip.conf has directmedia=yes which is confirmed with sip show peer in the Asterisk CLI
As you re using the same codec and no options on the dial command, I readed a thread about the same issue and it was solved by setting directmedia option directly on peer section instead of the [general] section, Also I dont think it is related with the RTP but it also mentioned changing type to peer instead of friend
when using directmedia=yes , Asterisk will attempt to send SIP reinvites in order to allow the endpoints to communicate directly. You can also try directrtpsetup=yes This sets up the call directly with media peer-2-peer without re-invites. This last option wont work for video and others conditions, so read the asterisk documentation http://svn.digium.com/svn/asterisk/trunk/configs/samples/sip.conf.sample
Enabling debug to the console in logger.conf and doing “core set debug 2” will cause a message to get output amongst everything which tells you why it didn’t do it.
Yes awesome thanks Josh.
Im learning about the new bridging architecture and local and remote Native RTP bridge.
Im not getting either as it comes up with:
‘can not use native RTP bridge as two channels are required’
Channels are added individually to the bridge, so the real reason is further down:
[Jul 21 12:14:08] DEBUG[11955][C-00000009]: bridge_native_rtp.c:334 native_rtp_bridge_compatible_check: Bridge ‘b25a204f-aea7-4cd1-abed-d7b38693970d’ can not use native RTP bridge as channel ‘SIP/1402-00000013’ has features which prevent it
Which will occur if there is something on one of the channels which needs to get access to the audio stream. WIth chan_sip I’m not sure what that would be.
Im a little unsure of your first question. The call is always successful, but never natively bridged which will be an issue for some of my systems that are remotely hosted.
I have attached a file containing debugs and SDP of a call from 1403 to 1402.
I have directmedia=yes and directrtpsetup=no in the [general] section of sip.conf
I think the first question was the result of confusing two threads.
Your log is saying that the channel has incompatible features, which suggests you have something requiring DTMF to be detected, (e.g. TtXx, etc. on DIal) or media to be intercepted or recorded (Monitor, ChanSpy, etc.).
Thanks David but I dont have these things you mentioned configured.
I am not passing ANY Dial options for this call and I do not have any Monitoring or ChanSpy activated.
Can you think of anything else that could be doing it? I think from the logs, it appears it does not like something configured on the destination peer?
Why do you feel surprised, you are also capable to produce effective knowledge and find answers to your own problem and share the answer with others as a good member