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(canreinvite) redirect the RTP media stream (audio) to go directly

venkattyso
Posts: 6
 Member Since:
2007-04-22
Submitted by venkattyso on Mon, 05/28/2007 - 9:51pm.

Hi all,
For the past few days I am trying to work on the RTP media stream to be able to transfer data directly between PHONE A AND B. That is once the session is initiated, even if the asterisk server goes down the existing session should still be able to communicate. I believe this can be done only when there is a direct link between two phones over the internet(After initiating the Session) rather than going through the asterisk server. This would also help me on the reduction of server load!
So it would be great if someone can help me configure this in asterisk/trixbox. I am breaking my head!

‹ ivr -> external Voice Tags ›

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venkattyso
Posts: 6
 Member Since:
2007-04-22
Problem Solved! Just need to
Tue, 05/29/2007 - 9:21pm

Problem Solved! Just need to remove the "tr" in the Dial Options = under [globals]>extensions_additional.conf and then add canreinvite=yes in the phone extensions and also in sip.conf, it should work! For more information refer to this webpage http://voip-info.org/wiki/view/Asterisk+sip+canreinvite


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tommyboay
Posts: 22
 Member Since:
2007-06-15
Thanks man. I was looking
Thu, 06/28/2007 - 6:54am

Thanks man. I was looking for the exact same thing. Trying your solution right now.

Works great. You can change the dial setting directly into Freepbx thru the general settings

--

ToIP & network security consultant
NetXP - France
www.netxp.fr


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eoo
Posts: 448
 Member Since:
2006-10-30
are you sure it is really
Thu, 06/28/2007 - 10:08am

are you sure it is really working? if you stop asterisk, does the call continue?
as i understand it, there are a lot of problems with this, especially if NAT traversal between the endpoints is involved. i also thought that this only even vaguely sorted out with asterisk 1.4 and doesn't work well at all with the 1.2.x versions of asterisk.


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venkattyso
Posts: 6
 Member Since:
2007-04-22
it is working, but again as
Thu, 06/28/2007 - 7:17pm

it is working, but again as you said when the NAT is involved there is some problem with just one way communication and i am not sure about asterisk 1.2 versions compatibility as i have upgraded my asterisk 1.2 V in trixbox to 1.4.
I am currently working on the NAT issue once i figure that out i can let others know, not an issue!


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chrisbware
Posts: 7
 Member Since:
2007-05-19
What you are loosing
Fri, 07/27/2007 - 12:09pm

if you don't put tT, audio is direct between UA but you cant use transfer....too bad!


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tadaxi
Posts: 9
 Member Since:
2006-09-23
i am considering
Tue, 08/21/2007 - 2:14am

i am considering it,sir

thanks first!

i will try it in trixbox pro


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