[Freeswitch-users] Help with attended transfer / att_xfer

Rob Hutton justlikeef at gmail.com
Fri Feb 10 23:46:57 MSK 2012


We're using 2100s also, and the MOH plays when doing an attended transfer.
 If you look at the process, pressing the second line button actually puts
the first on hold.

On Fri, Feb 10, 2012 at 9:55 AM, Brett Wilson <brett at launch3.net> wrote:

> I have the MOH setup correctly I think. When I do a blind transfer through
> the phone, and the other extension is ringing and not picked up yet, MOH
> does work. Only the phone’s attended transfer exhibits the strange
> behavior. I’ll have to get around to doing a packet cap one day.****
>
> ** **
>
> *Brett Wilson*
>
> *Launch 3 Ventures, LLC*****
>
> 134 Myer Street****
>
> Hackensack, NJ 07601****
>
> *Phone:* 877.878.9134
> *Fax:* 646.536.3866****
>
> *Email:* Brett.Wilson at launch3.net****
>
> *AOL IM:* Brett.Wilson at launch3.net****
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> ** **
>
> *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com]
> *Sent:* Thursday, February 09, 2012 11:47 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Help with attended transfer / att_xfer**
> **
>
> ** **
>
> normal sip transfers should support MOH, you just have to set hold_music
> to the filename of your moh or install the default config and sound sets
> and swap out the sounds in the moh local_stream****
>
> 2012/2/8 João Mesquita <jmesquita at freeswitch.org>****
>
> If I am reading code correctly, you should be able to insert the required
> play code around line #2105 of mod_dptools.c****
>
> ** **
>
> I might add this patch tomorrow because I believe it is a pretty cool
> feature to have with a configurable tone to be played.****
>
> ** **
>
> Regards,****
>
> ** **
>
> -- ****
>
> João Mesquita****
>
> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>****
>
> ** **
>
> On Wednesday, February 8, 2012 at 9:39 PM, Michael Collins wrote:****
>
> I agree w/ jmesquita - this probably needs a patch, although I'd need to
> dig further to see if there's a non-patchy workaround.
>
> As far as FusionPBX goes, you should be able to do a git pull and make
> install to keep yourself updated. Just confirm with Mark Crane (IRC:
> mcrane) or one of the other fusionpbx guys about any caveats with things
> like the target install directory, etc.
>
> -MC****
>
> On Wed, Feb 8, 2012 at 4:14 PM, Brett Wilson <brett at launch3.net> wrote:
>
> ****
>
> Thanks guys I am using the FS att_xfer along with the meta bind. I tried
> attended transfer on my gxp2100 phones and it works but I am under the
> impression that the phone is actually handling the sip switching itself for
> that functionality, and our MOH was not being played while the two internal
> parties were speaking, before putting through the transfer. It was just
> dead silence which did not sit well with our customers. So I switched to
> the FS feature, and it works great. Only problem is that we don’t know when
> to start speaking after the transfer!****
>
>  ****
>
> Also I have another question. I have FS installed via the FusionPBX linux
> disk image. What is the best way to keep FS updated? Do I need to pull from
> git and build from source or what?****
>
>  ****
>
> *Brett Wilson*****
>
> *Launch 3 Ventures, LLC*****
>
> 134 Myer Street****
>
> Hackensack, NJ 07601****
>
> *Phone:* 877.878.9134
> *Fax:* 646.536.3866****
>
> *Email:* Brett.Wilson at launch3.net****
>
> *AOL IM:* Brett.Wilson at launch3.net****
>
> www.Launch3.net****
>
> *www.Launch3telecom.com <http://www.launch3telecom.com/>*****
>
> *******************************
>
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> ****
>
>  ****
>
> *From:* João Mesquita [mailto:jmesquita at freeswitch.org]
> *Sent:* Wednesday, February 08, 2012 5:08 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Help with attended transfer / att_xfer**
> **
>
>  ****
>
> I believe he is using the feature code, therefore att_xfer so this would
> require a patch of some kind. I am this on my todo list which is quite
> large nowadays and I don't even know if it is feasible yet. I can take a
> look at the code later on.****
>
>  ****
>
> Regards,****
>
>  ****
>
> -- ****
>
> João Mesquita****
>
> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>****
>
>  ****
>
> On Wednesday, February 8, 2012 at 2:40 PM, Michael Collins wrote:****
>
> Brett,
>
> No worries on the double-post - we have a first-time sender moderation
> filter so I had to allow your messages through. You should be go to go from
> here on out.
>
> As to your question, are you using the default configuration or are you
> doing something different? Also, how are you executing the transfer - using
> the transfer button on the telephone or the *1 feature code?
>
> Thanks,
> MC****
>
> On Tue, Feb 7, 2012 at 3:24 PM, Brett Wilson <brett at launch3.net> wrote:***
> *
>
> Hey guys,****
>
> I need a pointer. I’m trying to get some kind of functionality where after
> a attended transfer is completed, ie. The ouside caller is connected to the
> 2nd phone after the 1st phone has hung up or chosen to continue the
> transfer. I would like a beep to sound for the 2nd phone or for both
> legs, just to let the recipient of the transfer know that it has gone
> through and they can start talking.****
>
> Any ideas?****
>
>  ****
>
> Thanks****
>
>  ****
>
>  ****
>
> *Brett Wilson*****
>
> *Launch 3 Ventures, LLC*****
>
> 134 Myer Street****
>
> Hackensack, NJ 07601****
>
> *Phone:* 877.878.9134
> *Fax:* 646.536.3866****
>
> *Email:* Brett.Wilson at launch3.net****
>
> *AOL IM:* Brett.Wilson at launch3.net****
>
> www.Launch3.net****
>
> *www.Launch3telecom.com <http://www.launch3telecom.com/>*****
>
> *******************************
>
> [image: Description: Description: Blogger-logo]<http://launch3telecom.blogspot.com/>[image:
> Description: Description: FaceBook-Logo]<http://www.facebook.com/pages/Launch-3-Telecom/165610820178720>[image:
> Description: Description: Twitter-Logo]<https://twitter.com/launch3telecom>
> ****
>
>  ****
>
>
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