[Freeswitch-users] att_xfer limitations

[Anthony Minessale]

Emrah,

I suggest you further study the application as a whole before making
determinations on where improvements need to be made.
You may need to think a bit more "out of the box" if you want to
be successful.



On Thu, Sep 6, 2012 at 2:03 PM, João Mesquita <jmesquita at freeswitch.org>wrote:

> I do believe that there are some legit cases where att_xfer is an
> application that needs to be fully functional. The most obvious case is
> where analog phone are used with boards such as Sangoma, Khomp or Digium
> boards. It is in this instance arguable if that needs to be implemented as
> an application on the "core" so to speak on or the endpoint module.
>
> It is not a trivial task (at least beyond my capabilities) to make some
> extra things work on att_xfer such as call return.
>
> Nonetheless, Emrah, what you are asking is doable with the current
> implementation. Just use the loopback channel and make
> loopback_bowout=false. The downside in this case is that the loopback
> channel will be up during the entire duration of the call, but everything
> else will work.
>
> Also, this adds some extra complexity to CDR processing, but I haven't
> worked on that to really know how much complexity.
>
> I hope this works for you.
>
> Regards,
> João Mesquita
>
>
>
>
> On Thu, Sep 6, 2012 at 12:46 PM, Emrah <lists at kavun.ch> wrote:
>
>> Hi Michael,
>>
>> It was great to hear you as well, will try to be a regular. :)
>>
>> I use a Polycom and it does have all the capabilities I need, but I
>> wanted to have the PBX side working for a couple of scenarios where I
>> thought I might need it.
>> E.g.: pick up a call on your cellphone, walk into the office, att_xfer
>> the call from your cell to your desk and continue the conversation. The
>> reason of using att_xfer is to make sure to release the call when the line
>> is properly established and prevent calls from landing in voicemail
>> inadvertently.
>>
>> And on a final note, I never had any issue transferring calls from my SIP
>> phones, but for some reason I find the Polycom way somewhat cumbersome and
>> repelling.
>>
>> Best,
>> Emrah
>> On Sep 6, 2012, at 11:23 AM, Michael Collins <msc at freeswitch.org> wrote:
>>
>> > What kind of phone are you using? The reason I ask is that most desk
>> phones have real transfer keys that do all of this. If you're stuck using
>> something like x-lite then yeah, I can see the dilemma. In all honesty, the
>> att_xfer app wasn't designed as the be all end all of call transfers. If
>> you are in a scenario where your only option is to use att_xfer then I
>> recommend solving that problem. In the long run it will be much better for
>> you.
>> >
>> > -MC
>> >
>> > P.S. - It was nice having you on the conf call yesterday! Hope you can
>> join more calls. :)
>> >
>> > On Thu, Sep 6, 2012 at 7:38 AM, Emrah <lists at kavun.ch> wrote:
>> > Hi guys,
>> >
>> > I just briefly tried the att_xfer app and it seems very limited to me.
>> It seems to be either using a gateway or the db, but it never hits the
>> dialplan at any point.
>> > I love the extra add-ons you guys built, especially the option to
>> conference before finalizing the transfer, but I think the app should be
>> revised at some point to support a more dialplan oriented approach.
>> > Now, when a call comes in, I can att_xfer to a SIP user, but can't call
>> out a cellphone using my dialplan logic with my multiple outbound providers.
>> >
>> > I encourage you to give it a look and see if it's worth an upgrade.
>> >
>> > Best as always,
>> > Emrah
>> >
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>> > --
>> > Michael S Collins
>> > Twitter: @mercutioviz
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>
>
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-- 
Anthony Minessale II

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