vip killa
2011-Feb-23 16:10 UTC
[asterisk-users] REFER and dialplan broken (as documented in chan_sip.c on line 11951)
There is a problem when transferring calls using REFER, asterisk does not notify dialplan. I've been told to use AMI as a workaround to notify my dialplan/routing program but that would require a huge change to our software. I was wondering if there is any intention of fixing this problem. Here is issue as stated in chan_sip.c "this is currently broken as we have no way of telling the dialplan engine whether a transfer succeeds or fails." Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110223/dc58b8d8/attachment.htm>
Danny Nicholas
2011-Feb-23 16:28 UTC
[asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
_____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 10:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) There is a problem when transferring calls using REFER, asterisk does not notify dialplan. I've been told to use AMI as a workaround to notify my dialplan/routing program but that would require a huge change to our software. I was wondering if there is any intention of fixing this problem. Here is issue as stated in chan_sip.c "this is currently broken as we have no way of telling the dialplan engine whether a transfer succeeds or fails." Thanks. I'm quite certain that this problem is being considered (for reference, this is a 1.8.X issue). If you aren't satisfied with the progress being made, you should research your own solution and/or offer a bounty. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110223/df598029/attachment.htm>
Jeff LaCoursiere
2011-Feb-23 19:38 UTC
[asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
On Wed, 23 Feb 2011, Danny Nicholas wrote:> > ____________________________________________________________________________________________________________________________________________________________ > > > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of vip killa > Sent: Wednesday, February 23, 2011 12:44 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) > > ? > > I recognize all the options given yet as I explained before they are not viable. I do not have the resources to pay someone, I do not have the expertise to > fix this issue because according to an asterisk developer "any fix in that area would be deeply architectural in nature"... what other options are there? > > ? > > <snip> > > From what I see, the ?source fix? on the Asterisk level would indeed be a major undertaking.? But since you are using an AGI to control the Queue command > instead of using it from the dialplan, you have more control over this problem than you realize.? For simplicity of illustration, let?s say your AGI simply > wants to take a call and send it to the next agent in the queue. Your Agents are Agent007, AgentQ and AgentM.? Because you did the Polycom transfer from > Agent007 to pussygalore, Agent007 is marked as busy in the queue although the call is no longer active for 007. One possible workaround would be to have a > duplicate ?bail queue? set up the same way.? If my AGI does a ?core show channels? and sees that 007 is not on the phone, I can do queue(bail) instead of > queue(normal). > > ? >Watch out for race conditions doing things like this... j