I use Polycom 501's and use the Transfer Key to send inbound calls to other extensions. Can you give me an A-B-C example of how this problem manifests itself?
_____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Interesting but the issue I'm having relates to Inbound and Outbound REFERs since I'm using Polycom's Transfer softkey (which allows for both Inbound and Outbound Transfers). I know this is not an issue when using Asterisk's built-in transfer (only allows Inbound transfers). On Wed, Feb 23, 2011 at 12:01 PM, Danny Nicholas <da...@debsinc.com> wrote: Have you read this thread? http://forums.digium.com/viewtopic.php?t=74418 _____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 10:36 AM 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 did not see this issue anywhere on issues.asterisk.org Can you give me a reference number to the issue? Also, it is a problem with all releases of asterisk. On Wed, Feb 23, 2011 at 11:28 AM, Danny Nicholas <da...@debsinc.com> wrote: _____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@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. -- _____________________________________________________________________ -- 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 -- _____________________________________________________________________ -- 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
-- _____________________________________________________________________ -- 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
