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" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@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 <[EMAIL PROTECTED]> wrote: > >> On Jan 29, 2008 5:55 PM, Alex Balashov <[EMAIL PROTECTED]> 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 _______________________________________________ -- 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