[asterisk-users] Asterisk 1.6 Questions
I have a couple of questions about asterisk 1.6: Can you change codecs mid-call upon re-invite? Can you handle the SDP offer-answer in the 200-ACK instead of the usual INVITE-200? Thanks in advance for any insight. Gary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 Questions
On 05/03/2011 12:43 PM, Gary Graves wrote: Can you change codecs mid-call upon re-invite? Do you mean: can Asterisk be configured to _initiate_ such a change at some point, mid-call? Or do you mean: Will Asterisk properly react to such a re-INVITE and change codecs if asked to do so by the dialog counterparty? Can you handle the SDP offer-answer in the 200-ACK instead of the usual INVITE-200? Doesn't seem to. Looking at chan_sip.c in 1.6.2.13, there is no call to add_sdp() that is not made either in the context of 1) an initial INVITE request or 2) a re-INVITE or 3) the construction of a response. Nothing in the case of the production of an end-to-end ACK. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 Questions
Can you answer both? Can Asterisk be configured to _initiate_ such a change at some point, mid-call? and Will Asterisk properly react to such a re-INVITE and change codecs if asked to do so by the dialog counterparty? On Tue, May 3, 2011 at 12:56 PM, Alex Balashov abalas...@evaristesys.comwrote: On 05/03/2011 12:43 PM, Gary Graves wrote: Can you change codecs mid-call upon re-invite? Do you mean: can Asterisk be configured to _initiate_ such a change at some point, mid-call? Or do you mean: Will Asterisk properly react to such a re-INVITE and change codecs if asked to do so by the dialog counterparty? Can you handle the SDP offer-answer in the 200-ACK instead of the usual INVITE-200? Doesn't seem to. Looking at chan_sip.c in 1.6.2.13, there is no call to add_sdp() that is not made either in the context of 1) an initial INVITE request or 2) a re-INVITE or 3) the construction of a response. Nothing in the case of the production of an end-to-end ACK. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 Questions
On 05/03/2011 01:16 PM, Gary Graves wrote: Can you answer both? Can Asterisk be configured to _initiate_ such a change at some point, mid-call? I don't know of a way to do that. I suppose it might be possible if a call were asynchronously transferred to a SIP peer that had different codec requirements. and Will Asterisk properly react to such a re-INVITE and change codecs if asked to do so by the dialog counterparty? It should. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
