[asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

Wed Feb 23 12:01:32 CST 2011

My bad - "natively" means using the Queue command from the dialplan.  Since
the "powers that be" are aware of this problem,  I suppose it will get fixed
when somebody either has some spare time or a sufficient bounty is offered.

  _____  

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 11:57 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'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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