Dinesh Nair
2005-Oct-05 23:06 UTC
[Asterisk-Users] SIP Attended Transfer using REFER and Replaces: headers
hey all, am wondering if anyone has successfuly done a SIP attended transfer using the REFER method (after an INVITE obviously) and the Replaces: header. we're writing our own SIP UAC and the asterisk code seems to support it, but we're not really sure if this is so. we plan on the following call flows: 1. incoming call from exten 1111 is sent to our UAC with Dial() 2. our UAC makes another call (via asterisk) to exten 2222 3. 2222 answers the call 4. our UAC sends REFER with Replaces: and Call-Id: of call from 1111 to the SIP session with asterisk for 2222. 5. asterisk bridges 1111 and 2222. is this the way it's supposed to work ? (am not sure if this is a -users or -dev question, so pardon the x-posting) -- Regards, /\_/\ "All dogs go to heaven." dinesh@alphaque.com (0 0) http://www.alphaque.com/ +==========================----oOO--(_)--OOo----==========================+ | for a in past present future; do | | for b in clients employers associates relatives neighbours pets; do | | echo "The opinions here in no way reflect the opinions of my $a $b." | | done; done | +=========================================================================+
Olle E. Johansson
2005-Oct-06 03:29 UTC
[Asterisk-Users] SIP Attended Transfer using REFER and Replaces: headers
Dinesh Nair wrote:> > hey all, > > am wondering if anyone has successfuly done a SIP attended transfer > using the REFER method (after an INVITE obviously) and the Replaces: > header.That is not supported today. However, I have working code that will be submitted to the bug tracker after Astricon.> > we're writing our own SIP UAC and the asterisk code seems to support it, > but we're not really sure if this is so. > > is this the way it's supposed to work ? >Yes. /O
Kib Eki
2005-Oct-06 05:19 UTC
[Asterisk-Users] SIP Attended Transfer using REFER and Replaces: headers
Is there a schedules for this? Olle E. Johansson wrote:> This work will belong to a future version of Asterisk, not 1.2 release. > > /Olle > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >
Dinesh Nair
2005-Oct-06 09:42 UTC
[Asterisk-Users] SIP Attended Transfer using REFER and Replaces: headers
On 10/06/05 18:29 Olle E. Johansson said the following:> That is not supported today. However, I have working code that will be > submitted to the bug tracker after Astricon.there seems to be some support for REFER with the Replaces: header in CVS HEAD, with the Call-ID being used to match an existing SIP session and ending with a call to ast_channel_masquerade(). what does ast_channel_masquerade do in this case ? (i assumed, from looking at the code, that it tries to copy one channels params into another, and from then on mark the dead leg as a zombie) -- Regards, /\_/\ "All dogs go to heaven." dinesh@alphaque.com (0 0) http://www.alphaque.com/ +==========================----oOO--(_)--OOo----==========================+ | for a in past present future; do | | for b in clients employers associates relatives neighbours pets; do | | echo "The opinions here in no way reflect the opinions of my $a $b." | | done; done | +=========================================================================+