Re: [asterisk-users] re-invite (bypass asterisk) post call establishment
2008/4/22 Benjamin Jacob [EMAIL PROTECTED]: [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
Re: [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) - 0119198807x After connect, I dial - 31927x. This number 31927x 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 [EMAIL PROTECTED] 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. reInvite Description: 1957794313-reInvite ___ -- 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
Re: [asterisk-users] re-invite (bypass asterisk) post call establishment
On 21/04/2008, Benjamin Jacob [EMAIL PROTECTED] 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 ___ -- 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
[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___ -- 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
Re: [asterisk-users] re-invite (bypass asterisk) post call establishment
Apologies for not explaining the set up . Using AstMan API, I Originate a call to user A. User A is a conference bridge which needs pin authentication. So post 200 OK, I need to send DTMFs for that pin. After sending the pin, I Dial (using the Originate context) user B. Now user B is behind a PBX, so I need to dial the extension for user B. I send the extension digits using DTMFs again. So, if I set canreinvite=yes, as soon as I get a 183/200 OK from user B, re-Invites are sent to both participants with the other's SDP. 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. Another scenario would be to call user B first and then user A first. The same case applies over there as well. Is there any other way to tell asterisk when to do a re-Invite/control the timing of the re-Invite? Hope I am clear this time. cheerz - Ben. Steve Davies [EMAIL PROTECTED] wrote: On 21/04/2008, Benjamin Jacob 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 ___ -- 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.___ -- 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
