Benjamin Jacob
2008-Apr-21 08:34 UTC
[asterisk-users] re-invite (bypass asterisk) post call establishment
Hello ppl, Any way to do a re-invite and make RTP bypass Asterisk, after call establishment. In other words, I would like to control when to do the bypass work for peer-peer RTP flow. The issue is that I need to send DTMFs after dialing the user because most of the users are behind PBXes (having individual extensions) themselves and almost all of the PBXes send a 200 OK and then play out the PBX messages. So I need to send the extension DTMFs first, bridge the calls and then re-invite users for them to do a peer-peer rtp conversation. TiA, - Ben. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080421/de44dfdd/attachment-0001.htm
Steve Davies
2008-Apr-21 09:16 UTC
[asterisk-users] re-invite (bypass asterisk) post call establishment
On 21/04/2008, Benjamin Jacob <ben4asterisk at yahoo.com> wrote:> > > Hello ppl, > Any way to do a re-invite and make RTP bypass Asterisk, after call > establishment. > In other words, I would like to control when to do the bypass work for > peer-peer RTP flow. > The issue is that I need to send DTMFs after dialing the user because most > of the users are behind PBXes (having individual extensions) themselves and > almost all of the PBXes send a 200 OK and then play out the PBX messages. > So I need to send the extension DTMFs first, bridge the calls and then > re-invite users for them to do a peer-peer rtp conversation. > > TiA, > - Ben.You don't say what you've tried already, but as long as canreinvite=yes is set against the SIP peer, the RTP stream should be redirected once the connection is open. As far as DTMF to dial an extension at the remote end, have you looked at the D() parameter to the Dial command? Regards, Steve
Benjamin Jacob
2008-Apr-22 11:54 UTC
[asterisk-users] re-invite (bypass asterisk) post call establishment
Hi again, I tried this again, but the reInvite happens immediately after the 200 OK/ACK. And then the D() specified DTMF is sent. Attached is the SIP trace for the calls. I call (from Asterisk) - 0119198807xxxxx After connect, I dial - 31927xxxxx. This number 31927xxxxx is the conference bridge and I need to send DTMF (the bridge PIN) to it after connection. But alas, the reinvite happens before the D() is executed. The SIP gateway is MySIPGateway at 204.aaa.bbb.ccc. cheers - Ben. Steve Davies <davies147 at gmail.com> wrote: 2008/4/22 Benjamin Jacob : [snip]> > So, my question : once the SDPs are exchanged, what will happen to the DTMFs > sent by Asterisk using sendDTMF or the D option in dial. >[snip] As far as I can tell, the D() option will be executed before the re-invite takes place, so Asterisk will still be in-line. I believe that the dial is not considered "complete/connected" until the D() is finished. Cheers, Steve _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --------------------------------- Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080422/a2753a4d/attachment-0001.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: reInvite Type: application/octet-stream Size: 24529 bytes Desc: 1957794313-reInvite Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20080422/a2753a4d/attachment-0001.obj