[Freeswitch-users] Consultation Call via event_socket interface

Tue Sep 18 02:42:41 MSD 2012

MC, I believe I have to make a statement here in favor of some realities that are different from the ones most people on this list live in. I live in South America and an IP Phone here does not get lower than 120usd per unit. And I am talking about the cheapest yea link phone model. In brazil, where I come from, this is even higher.

Taking that into consideration, several lower end ip phones started to appear as well as hybrid systems where the majority of the extensions are still analog. In a system like that, CTI applications like the one our friend is describing is really the way out to really add value to a solution.

Even on the asterisk world we see applications like these. For example the flash operator panel and the HUD for asterisk. I believe most of us have heard of it before.

Anyhow, I didn't mean to write an essay on the subject but I see this kind of feature being constantly rejected by the community and I really can't understand why that is.

João Mesquita

On 17/09/2012, at 01:39 p.m., Michael Collins <msc at freeswitch.org> wrote:

> Or get a hard phone that has a hold button and at least two line keys.
> -MC
> 
> On Mon, Sep 17, 2012 at 5:19 AM, João Mesquita <jmesquita at freeswitch.org> wrote:
> From personal experience, I believe that how you described is the right way to do it. The only caveat is that you will have to add variables do the channels so you can properly track what is going on in the cdrs . If you don't process cdrs, then it is all good. Careful with pickup and such too...
> 
> On Sep 17, 2012 3:53 AM, "Alexander Haugg" <Alexander.Haugg at c4b.de> wrote:
> Hi MC,
> 
>  
> 
> thank you for the answer.
> 
> To your question, all call legs in this scenario (outgoing or incoming direction) are connected over a sip trunk of a pbx.
> 
>  
> 
> Scenario:
> 
> My CTI
> 
>   |  A
> 
>   |  |       event socket
> 
>   V  |
> 
> Freeswitch         Sip Trunk
> 
> <-                           PBX
> 
> ->                          
> 
>  
> 
> My Client control (CTI) the call legs over the event socket interface and the call legs are only legs over the sip trunk to or from the PBX.
> 
>  
> 
> After some tests i have found a possible solution:
> 
> -          Leg A and Leg B are bridged (all legs get the flag park_after_bridge = true)
> 
> -          For Consultation i park Leg B and transfer this Leg to Moh in my default context of my dialplan
> 
> -          i originate Leg C and bridge this Leg with Leg A (Leg C have the flag park_after_bridge = true too) Consultation is comlete now.
> 
> -          with the principle same think i can toggle Leg A <-> Leg B and Leg A <-> Leg C.
> 
>  
> 
> What is your think for this solution? I have tested this on the FS CLI and it works.
> 
>  
> 
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Collins
> Gesendet: Freitag, 14. September 2012 19:13
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] Consultation Call via event_socket interface
> 
>  
> 
> Hi Alex,
> 
> Welcome to the FreeSWITCH mail list! 
> 
> First question for you: what kind of telephone are you using? The reason I ask is that this kind of function is trivially achieved with a good hard phone, like a Polycom, Yealink, Snom, or Cisco with two or more line keys. If you can use a hard phone w/ multiple line keys then you don't even need to mess with the dialplan, uuid_bridge, etc.
> 
> -MC
> 
> On Fri, Sep 14, 2012 at 12:24 AM, Alexander Haugg <Alexander.Haugg at c4b.de> wrote:
> 
> Hi All,
> 
>  
> 
> I’m new on the mailing list.
> 
> I have a problem with a call scenario.
> 
> -          Channel A and channel B are bridged (A is my own channel and B is my calling partner)
> 
> -          Now i set channel B on hold with the command „uuid_hold xxx“ and create a new channel to C with the command:
> 
> bgapi originate {channel_csid=num,accountcode=num,origination_caller_id_name='num at ip',origination_caller_id_number=num at ip}sofia/external/num at ip  &park
> 
> this works correctly, the partner C answer and the channel is established.
> 
> -          Now the Problem:
> 
> I try to bridge the channel a to channel c with the uuid_bridge command, now the channel b will hangup, why? The variable hangup_after_bridge is by default false.
> 
> Other problem: channel A can hear the voice of channel C but not speak with him, channel C can hear and speak. But this problem is not the important think at the moment.
> 
> Is there a general problem in my plan to do that?
> 
> Is it a better plan to do this over the dialplan?
> 
> The next step in this scenario is to toggle the connection A -> B and A -> C.
> 
>  
> 
> Thanks for your help!
> 
> Nice regards,
> 
> Alex
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120917/f062ca64/attachment-0001.html 