This is a problem in chan_sip.c After REFER asterisk does not notify dialplan or AGI of REFER. I've tried to convince asterisk developers this is a problem but they only offered me 3 solutions: 1. Fix it yourself 2. Pay someone to fix it 3. Try to convince enough people that this is a problem and it may get fixed.
BTW this is not a simple fix, it would require architectural changes in asterisk. On Sun, Mar 6, 2011 at 9:32 PM, Louis Carreiro <[email protected]> wrote: > So does anyone have any other thoughts about this? I've done some searching > through the bug tracker for Asterisk but haven't seen anything related to > refer's failing. Does anyone know of a specific issue number for this? If > not, is this a valid bug to submit? Also, does anyone remember an Asterisk > version that this worked on? > > Thanks all! > > > On Fri, Mar 4, 2011 at 1:35 PM, Louis Carreiro <[email protected]>wrote: > >> Ha! Thanks Vip! >> >> Sorry about not including my version numbers too. On my production box I'm >> using 1.8.3 (that's the debug from the original email). On my demo box I >> just build I'm using 1.8 SVN-trunk-r309404 and that's what generated these >> logs. I'm not sure if this is a chan_sip.c problem or if this is a dial >> plan problem. >> >> So digging in a bit deeper, Asterisk is receving the real REFER message. >> The "REFER-TO: >> >> <sip:Lyncserver.internal.domain:5067;transport=Tls;ms-opaque=3bb3da5834ad2 >> 787?REPLACES=aa6f8871-4151-4149-ad5a-29ab941bf4d0%3Bfrom-tag%3D9227b8a39d% >> 3Bto-tag%3D8be38bb187>" is accurate and in chan_sip.c it knows how to >> manipulate it. It does grab the "from-tag" and "to-tag" and parses the >> data. On one of the lines below you can see it says "Looking for Call >> ID: [email protected]:5060 (Checking From) >> --From tag 15826bef52 --To-tag as41bacc0b". Then it moves on to bridging >> the peers/channels together. It's not until later that I get the final " >> SIP/2.0 481 Call leg/transaction does not exist" which doesn't make sense >> to me. Also, the Lync client says "Call was not transferred because >> [Original Extension] cannot be reached and may be offline." >> <-------- SNIP ---------> >> > > -- > _____________________________________________________________________ > -- 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 >
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