I have a similar issue with Sipura using compact headers, but not with regular headers. I am working on reproducing with the latest CVS. Maybe you are using compact SIP headers on your ATA186?
http://bugs.digium.com/bug_view_page.php?bug_id=0001843 Stephen > -----Original Message----- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Florian Overkamp > Sent: Wednesday, June 16, 2004 12:20 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] ATA186 v3.1 SIP - Attended transfer: NO JOY > > Hi, > > I'm still hassling with the consultative/attended transfer stuff. Someone > please help me identify this > > A lot has already been said about the ATA186. Some report it works fine, > others say it doesn't. Lets get clarity on this. > > My scenario is reasonably simple (I think) > Phone A: SIP/video1 > Phone B: SIP/werkkamer > Phone C: IAX2/provider > > Phone A calls phone B, they chat: > *CLI> show channels > Channel (Context Extension Pri ) State Appl. Data > SIP/werkkamer-91f5 (from-werkkamer 1 ) Up Bridged > Call > SIP/video1-e2a0 > SIP/video1-e2a0 (pbx 1202 1 ) Up Dial > SIP/swiss&SIP/snom&SIP/werkkamer&IAX2/florian&SIP/video1 > 2 active channel(s) > > Phone B hits flash and gets a dialtone. Dials a number and connects to > phone > C: > *CLI> show channels > Channel (Context Extension Pri ) State Appl. Data > IAX2[172.28.8.8:4569]/7 ( s 1 ) Up Bridged > Call > SIP/werkkamer-2507 > SIP/werkkamer-2507 (pbx 4307076 2 ) Up Dial > IAX2/provider/4307076 > SIP/werkkamer-91f5 (from-werkkamer 1 ) Up Bridged > Call > SIP/video1-e2a0 > SIP/video1-e2a0 (pbx 1202 1 ) Up Dial > SIP/swiss&SIP/snom&SIP/werkkamer&IAX2/florian&SIP/video1 > 4 active channel(s) > > Phone A now hears music on hold. Phone B and C can chat. > > Phone B now hits flash again. All phones end in a three-way conversation: > *CLI> show channels > Channel (Context Extension Pri ) State Appl. Data > IAX2[172.28.8.8:4569]/7 ( s 1 ) Up Bridged > Call > SIP/werkkamer-2507 > SIP/werkkamer-2507 (pbx 4307076 2 ) Up Dial > IAX2/provider/4307076 > SIP/werkkamer-91f5 (from-werkkamer 1 ) Up Bridged > Call > SIP/video1-e2a0 > SIP/video1-e2a0 (pbx 1202 1 ) Up Dial > SIP/swiss&SIP/snom&SIP/werkkamer&IAX2/florian&SIP/video1 > 4 active channel(s) > > Now the misery starts: If Phone B wants to back out of the conversation, > it > seems phones C and A are also disconnected. > > I've tried doing this with SIP firmwares, 2.15, 2.16, 3.0 and 3.1 and CVS > HEAD as of today. > > Other people have claimed success: > http://lists.digium.com/pipermail/asterisk-users/2003-August/018388.html > > Is this: > http://lists.digium.com/pipermail/asterisk-users/2003-August/018414.html > also related ? > > By the way, canreinvite=no as suggested by Mark in one of the slightly > related conversations on bugs.digium.com does not help... > > I would really _love_ to know why this is and to see it fixed somehow. A > bounty would be in order. Can anyone comment on this ?? > > On a related note: If the consultation ends in a failure (user unavailable > or unable to talk) the way to back out is hitting flash once if the remote > hung up (ata doesn't give any tone at that time??) or twice if you got > voicemail. The remote (phone A) briefly hears this, as the first flash > opens > a three-way conversation with phones A, B and the voicemail. The second > one > then disconnects the voicemail again. Not really elegant (albeit useable). > Is there a better way ? > > Best regards, > Florian > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
