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

vip killa vipkilla at gmail.com
Wed Feb 23 12:27:41 CST 2011 

It's simple, if a product is broken shouldn't it be fixed? In this case the
answer is "for a price" which is absurd because it is an open source
product. If there was a decent community of developers surrounding this
"open source project", it would be fixed simply because it's broken, no
questions asked.

On Wed, Feb 23, 2011 at 1:19 PM, Watkins, Bradley <
Bradley.Watkins at compuware.com> wrote:

> Implying that the Asterisk developers (which is itself a fairly nebulous
> statement since those who contribute to Asterisk are many and come from
> different companies/countries/etc.) are “not in it to make a good product”
> but to make a “profit” is not only highly insulting but a complete
> mischaracterization of what you were told on IRC.
>
>
>
> What you were told was that there are essentially three choices (and this
> goes for pretty much any open source software, as already stated).
>
>
>
> You may either fix it yourself (if you have the skills), pay someone to fix
> it for you (if you can or must trade money for expediency), or wait for
> someone else with the skills and/or money necessary to fix it.
>
>
>
> Regards,
>
> - Brad
>
>
>
> *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 1:05 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)
>
>
>
> Yes, they want money, they've told me that several times...it's unfortunate
> that asterisk's dev community is not in it to make a good product but a
> profit
>
> On Wed, Feb 23, 2011 at 1:01 PM, Danny Nicholas <danny at debsinc.com> wrote:
>
> 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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