Sure, it really manifests itself whenever using AGI for call flow, but this is how it affects us... incoming call -> queue -> agent007 -> xfer -> pussygalore now the AGI/dialplan thinks agent007 is on phone with pussygalore until that xfered call terminates so if another call comes into queue while pussygalore is on the phone w/ that xfered call, agent007 will not even be attempted by queue
I'm sure there could be other scenarios in which this REFER issue could pose a problem but this is the most consequential scenario which we have to deal with everyday. On Wed, Feb 23, 2011 at 12:23 PM, Danny Nicholas <da...@debsinc.com> wrote: > 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 >
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