[Freeswitch-users] Consultation Call via event_socket interface

Michael Collins msc at freeswitch.org
Tue Sep 18 21:37:53 MSD 2012


Seven,

I think you're right about att_xfer being a tool for a different job. I'm
also interested in knowing if using a conference with the mute/unmute and
relations is a feasible approach. However, I think the OP may be able to do
this all with a little bit of scripting and maybe uuid_dual_transfer +
uuid_bridge.

I'll ruminate on this one a bit and maybe we can kick it around after the
featured presentation on tomorrow's conference call.

-MC

On Mon, Sep 17, 2012 at 4:49 PM, Seven Du <dujinfang at gmail.com> wrote:

> I had the similar issue, when using mod_fifo with mixed inbound/outbound
> agents, I have to disable the multiline feature on hard phones (any one
> knows how to disable on eye beam or xlite?)
>
> att_xfer could do this job except it has problems when use with
> event_socket http://jira.freeswitch.org/browse/FS-4419
>  , maybe it's not designed to this kind of usage.
>
> Yes, I would use the similar approach with Alexander if I'm doing this.
> Also I wonder if it's possible to just use a conference with mute/unmute
> hear/nohear and relate/unrelate combinations.
>
>
> --
> Seven Du
> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>
> On Tuesday, September 18, 2012 at 7:19 AM, Michael Collins wrote:
>
> JM,
>
> That actually explains a lot. I appreciate the information. I can
> definitely see where the relative expense of a hard phone would make this
> kind of application quite valuable. I hope you guys are able to iron out
> the details. Also, maybe you could see if Mitch Capper could add line
> appearances to FSClient. :)
>
> -MC
>
> On Mon, Sep 17, 2012 at 3:42 PM, Jmesquita at freeswitch.org <
> jmesquita at freeswitch.org> wrote:
>
> 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****
>
>
> _________________________________________________________________________
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> 
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>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
> ****
>
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>
> 
> 
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>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
> 
> 
>
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> 
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>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
> 
> 
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>
> 
> 
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-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
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