[Freeswitch-users] att_xfer limitations

Fri Sep 7 00:09:46 MSD 2012

Dear Anthony,

You're message did not come across very pleasantly, but as it is often the case in writing, sometimes the receiver does not perceive things as they were meant originally. I'll disregard the judgmental portion of your email.

When you refer to the application as a whole, should I assume FreeSWITCH or ATT_XFER? 
When you mean thinking outside the box, am I to assume creating work arounds (e.g.: by using loopback calls?) I don't find that quite sexy to be honest.

My previous message never demanded or even asked for an upgrade of a functionality. I stated my thoughts and encouraged the developers to consider my approach and see whether it was worth a revision. I obviously did not mean to attack your work in any way.

So here are my 2 cents:
The att_xfer feature has been implemented for a certain reason. I've seen it working differently on Asterisk where the dialplan is used all the time. Based on that, and as a personal thought worth mentioning, I suggested a similar approach. Leaving the feature as it is, and based on the limited knowledge I currently have on FreeSWITCH, it makes it unusable in any of my use cases.

I divorced my Asterisk boxes and made a commitment to FS. You've built something amazing and I will always try to be constructive where I can.

Best to you Anthony, and again congrats for the super initiative you took by creating FS.

Emrah

On Sep 6, 2012, at 3:23 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> 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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> -- 
> Anthony Minessale II
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