Hi, Response below.
In the meantime: I would REALLY appreciate comments from an ATA186 SIP user who can tell me: - How to transfer a call without using #-transfer - Preferably more or less like how we are used to transferring in a classic pbx system Noteworthy: - Which Asterisk version (CVS/CVS-HEAD/...) - Which ATA186 firmware Thanks, Florian > -----Original Message----- > 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 I have not found any setting on the ATA that can make such a difference in approach. > > -----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/0 18388.html > > > > Is this: > > > http://lists.digium.com/pipermail/asterisk-users/2003-August/0 18414.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
