[asterisk-users] chanspy does not pull the call back to asterisk after a reinvite

[Anthony Francis]

Franklin Webb wrote:
> Thanks to both of you for your input.  I'll be in touch off list Steve.
>
> -Franklin
> ----- Original Message -----
> From: "Steve Totaro" <stotaro at totarotechnologies.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com>
> Sent: Wednesday, January 30, 2008 8:00:48 AM (GMT-0500) America/New_York
> Subject: Re: [asterisk-users] chanspy does not pull the call back to asterisk after a reinvite
>
> On Jan 29, 2008 8:36 PM, Steve Totaro <stotaro at totarotechnologies.com> wrote:
>   
>> On Jan 29, 2008 5:55 PM, Alex Balashov <abalashov at evaristesys.com> wrote:
>>     
>>> Franklin,
>>>
>>> Because ChanSpy() is a "passive" monitor, there is nothing about the
>>> implementation that would cause Asterisk to shunt the speech back to
>>> itself.  Asterisk only does this in situations where it is out of the
>>> media path and needs to insinuate itself back into it for the purpose
>>> of generating media, such as on-hold music, IVR, etc.
>>>
>>> What you're wanting should, in my opinion, basically be submitted as a
>>> feature request.  Perhaps the developers can add a flag to the ChanSpy()
>>> invocation repertoire to make this work.
>>>
>>> Cheers,
>>>
>>> -- Alex
>>>
>>> --
>>> Alex Balashov
>>> Evariste Systems
>>> Web    : http://www.evaristesys.com/
>>> Tel    : +1-678-954-0670
>>> Direct : +1-678-954-0671
>>>       
>> Alex, he was not asking why, it is obvious he knows why.
>>
>> He was asking for a solution or idea on how to work around this issue.
>>
>> Are you using Sangoma cards?  If so, I might have a very good answer
>> for you, as well as another very possible different solution.  Both
>> would be outside of Asterisk so some kind of magic would have to
>> happen to associate the call being spied on to the channel but that
>> should not be that difficult if you even need it.
>>
>> Another solution is to track down the code referenced here
>> http://bugs.digium.com/view.php?id=9888 and modify chanspy to do a
>> reinvite back to asterisk before starting the spy.
>>
>> Anyways, I am sure it can be done.  The question is how much time is
>> it worth to make it happen.
>>
>> Maybe we should meet for lunch this week.  I can meet you in cow
>> country or Philly if you want, your choice.  I have to go to both this
>> week anyways and would like to catch up with things since Astricon.
>>
>> Thanks,
>> Steve Totaro
>>
>>     
>
> I just confirmed that there is a solution that is perfect for this
> that has been developed with a web interface to select the call to
> monitor.  A little added code and you can pretty easily look up who
> the agent handling the call is.
>
> Let's test it out on your call center.  Again, it is not an Asterisk
> app and would have no impact on your operations if it does not work.
>
> Thanks,
> Steve Totaro
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>   
in sip.conf do canreinvite=no, and suddenly the audio is always 
available to asterisk.

Anthony