Hi everybody,
I would like to bypass Asterisk for RTP streams. I’ve tried with ‘canreinvite’ option in sip.conf but it doesn’t work.
When 2 phones are communicating, I can’t have a native bridge.
Asterisk says :
Attempting Native bridge of 'x' and 'y'
Native bridge of 'x' and 'y' was unsuccessful
Someone can help me ?
Thank you.
That’s not enough information!
You’ll have to say a bit more about your setup and what relationship - network wise - the two phones are to each other and to the Asterisk server.
Simple dialplan and phones in the same context.
Part of my sip.conf :
in extensions.conf :
My Asterisk server and phones are connected on a same switch.
I tried also with other IP Phones (pingtel) and softphones, but there isn’t difference.
Actually, i only want that my RTP Stream bypass Asterisk.
Canreinvite does work, as long as the incoming and outgoing streams are passthrough (ie - the same codec). Please advise as to what your codecs in and out are and post a verbose CLI when the error occurs.
Maybe it depends from the Transfer option in the Dial() command;
C.
[quote][internal]
200,1,Dial(SIP/sjphone,20,rtT)
300,1,Dial(SIP/cisco,20,rtT) [/quote]
I would remove the “t” and “T” from the Dial() command. This requires asterisk to ‘listen’ to the RTP stream for the ‘#’ key.
[quote][internal]
200,1,Dial(SIP/sjphone,20,r)
300,1,Dial(SIP/cisco,20,r) [/quote]
might work.
Lonnie
Indeed, that is the problem. Remove the transfer capabilities in the dialplan (tT).
What about music on hold? If RTP stream doesnt pass thru * can I lunch MOH?
What about music on hold? If RTP stream doesnt pass thru * can I lunch MOH?[/quote]
No. You could have Music on Hold before the call is answered by the endpoint, but once answered and the REINVITE occurs Asterisk is out of the picture.
thank you.
it works 
Music on hold can be available when you do native bridge RTP.
I’ve got different multiline phones on my desk.
When I place a call to any extension, I have my system configured to use native bridged RTP. (To cut down on the traffic to the Asterisk system.)
If you press the hold button on any station, it directs the RTP stream to the Asterisk system which begins to play the MOH music.
I’m not sure how well that would work for analog stations, but it seems to work for multiline SIP stations.
[quote=“dufus”]
I’m not sure how well that would work for analog stations, but it seems to work for multiline SIP stations.[/quote]
It MUST work with analog stations because they are pluged into PC box with Linux and Asterix.
I was thinking of analog stations connected with an ATA. (Analog Telephone Adapter.)
In those cases, I’m not sure what sort of signalling a hookswitch flash will do.