[asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
Warren Selby
wcselby at selbytech.com
Wed Feb 23 12:47:45 CST 2011
Sorry for the top post - this is from my phone.
Sounds like the issue may actually be with the AGI that is handling your ACD queue. I've used the built-in Queue() command to handle situations like you describe without running into the issues you detailed. And that's with Polycom phones, too.
Without more details, I'm not sure how much help you're going to get. Show us some console output of the issue, capture the proper debug logs, etc, and perhaps you'll find more help.
Thanks,
--Warren Selby, dCAP
On Feb 23, 2011, at 11:57 AM, vip killa <vipkilla at gmail.com> wrote:
> I'm sorry i don't know what you mean by natively. I'm almost certain the queue is handled via AGI and not using asterisk's queue.
>
> On Wed, Feb 23, 2011 at 12:52 PM, Danny Nicholas <danny at debsinc.com> wrote:
> Do you use the Queue command “natively” or from the AGI? In the example you gave, if you did a “core show channels”, I assume that Agent007 would be idle, but ineligible for Queue activity.
>
>
>
> 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 11:37 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)
>
>
> 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 <danny at 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-bounces at lists.digium.com [mailto:asterisk-users-bounces at 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 <danny at debsinc.com> wrote:
>
> Have you read this thread?
>
> http://forums.digium.com/viewtopic.php?t=74418
>
>
>
>
>
> 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 10:36 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 <danny at debsinc.com> 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 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.
>
>
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