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
Re: [asterisk-users] chanspy does not pull the call back to asterisk after a reinvite
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 -- Franklin Webb Asst Project Manager Inter Medi@ Marketing Solutions 610-701-9670 [EMAIL PROTECTED] ___ -- 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
Re: [asterisk-users] chanspy does not pull the call back to asterisk after a reinvite
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
Re: [asterisk-users] chanspy does not pull the call back to asterisk after a reinvite
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 ___ -- 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
[asterisk-users] chanspy does not pull the call back to asterisk after a reinvite
Hello all, I am allowing a reinvite between a snom 320 phone and a SIP gateway to take load off my Asterisk server. When I put the caller on hold, for example, Asterisk successfully reinserts itself into the rtp stream to play music on hold to the caller, but when I do a chanspy Asterisk does not seem to pull the call back. If I am spying on a channel when the call build up happens the reinvite never occurs and it works, but I cannot jump in and spy on a call in progress once the reinvite has happened. Has anyone run into this issue any maybe have a solution, or does anyone know of a good way to get that call back onto the Asterisk switch from another extension prior to calling chanspy? Thanks much, Franklin Webb -- Franklin Webb Asst Project Manager Inter Medi@ Marketing Solutions 610-701-9670 [EMAIL PROTECTED] ___ -- 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
Re: [asterisk-users] chanspy does not pull the call back to asterisk after a reinvite
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 ___ -- 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
