Re: [asterisk-users] Call recording and transfer issue (asterisk 1.8)
On Mon, 2012-07-30 at 08:39 -0500, Matthew Jordan wrote: - Original Message - From: Ishfaq Malik i...@pack-net.co.uk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 18, 2012 9:58:47 AM Subject: Re: [asterisk-users] Call recording and transfer issue (asterisk 1.8) On Thu, 2012-04-19 at 12:20 +0100, Ishfaq Malik wrote: Hi I'm having a problem with the entirety of a call being recorded in the following scenario I'm using asterisk 1.8.7.0 Person A (asterisk peer) calls Person B (not on asterisk, real world number via a SIP trunk) Mixmonitor is invoked by Person A in the outbound context and AUDIOHOOK_INHERIT(MixMonitor)=yes is also set Person a transfers Person B to Person C (another asterisk peer) Person A is no longer involved in the call and the call is bridged between Person B and Person C The call recording stops as soon as Person A hangs up, even though AUDIOHOOK_INHERIT is set Is there any way we can get the entire call recorded in one file? Thanks in advance Ish Ish: Leif had a pretty good explanation of why AUDIOHOOK_INHERIT behaves this way in the comments of ASTERISK-16013. I'll quote it here: Well I just tested this scenario. ... after a bit of testing I determined the scenarios. Working: * Party A places a call to Party B * Party B places an attended transfer to Party C * Party A and C are not talking * Call recording works as expected Not working: * Party A places a call to Party B * Party A places an attended transfer to Party C Call recording works up to this point – the recording of the conversation between Party A and Party B, and the portion of the conversation between Party A and Party C is recorded * Party A now hangs up * Call recording is now stopped * Party B and Party C are now speaking (unrecorded) To me, this is actually the intended and expected behavior. The AUDIOHOOK_INHERIT() function is executed on the channel created by Party A, and thus the call recording is going to follow Party A around when it is transferred around the system. However, once Party A is kicked out of the conversation (i.e. they hangup) then the call recording stops because that is the channel the recording is associated with. Note that if you read the scenario description of AUDIOHOOK_INHERIT at https://wiki.asterisk.org/wiki/display/AST/Function_AUDIOHOOK_INHERIT, the function works in the transfer scenarios where the called party initiates the transfer, not the callee. For your scenario, you could try setting the MixMonitor on the called party channel as opposed to the callee channel, using one of the Dial GoSub/Macro options (U,M,b). https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial Note that Macro is deprecated in more recent versions of Asterisk, and the 'b' option will only be available in Asterisk 11. Thank you for the hints at the end, using the M option has sorted my issue out Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording and transfer issue (asterisk 1.8)
- Original Message - From: Ishfaq Malik i...@pack-net.co.uk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 18, 2012 9:58:47 AM Subject: Re: [asterisk-users] Call recording and transfer issue (asterisk 1.8) On Thu, 2012-04-19 at 12:20 +0100, Ishfaq Malik wrote: Hi I'm having a problem with the entirety of a call being recorded in the following scenario I'm using asterisk 1.8.7.0 Person A (asterisk peer) calls Person B (not on asterisk, real world number via a SIP trunk) Mixmonitor is invoked by Person A in the outbound context and AUDIOHOOK_INHERIT(MixMonitor)=yes is also set Person a transfers Person B to Person C (another asterisk peer) Person A is no longer involved in the call and the call is bridged between Person B and Person C The call recording stops as soon as Person A hangs up, even though AUDIOHOOK_INHERIT is set Is there any way we can get the entire call recorded in one file? Thanks in advance Ish Ish: Leif had a pretty good explanation of why AUDIOHOOK_INHERIT behaves this way in the comments of ASTERISK-16013. I'll quote it here: Well I just tested this scenario. ... after a bit of testing I determined the scenarios. Working: * Party A places a call to Party B * Party B places an attended transfer to Party C * Party A and C are not talking * Call recording works as expected Not working: * Party A places a call to Party B * Party A places an attended transfer to Party C Call recording works up to this point – the recording of the conversation between Party A and Party B, and the portion of the conversation between Party A and Party C is recorded * Party A now hangs up * Call recording is now stopped * Party B and Party C are now speaking (unrecorded) To me, this is actually the intended and expected behavior. The AUDIOHOOK_INHERIT() function is executed on the channel created by Party A, and thus the call recording is going to follow Party A around when it is transferred around the system. However, once Party A is kicked out of the conversation (i.e. they hangup) then the call recording stops because that is the channel the recording is associated with. Note that if you read the scenario description of AUDIOHOOK_INHERIT at https://wiki.asterisk.org/wiki/display/AST/Function_AUDIOHOOK_INHERIT, the function works in the transfer scenarios where the called party initiates the transfer, not the callee. For your scenario, you could try setting the MixMonitor on the called party channel as opposed to the callee channel, using one of the Dial GoSub/Macro options (U,M,b). https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial Note that Macro is deprecated in more recent versions of Asterisk, and the 'b' option will only be available in Asterisk 11. -- Matthew Jordan Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording and transfer issue (asterisk 1.8)
On Thu, 2012-04-19 at 12:20 +0100, Ishfaq Malik wrote: Hi I'm having a problem with the entirety of a call being recorded in the following scenario I'm using asterisk 1.8.7.0 Person A (asterisk peer) calls Person B (not on asterisk, real world number via a SIP trunk) Mixmonitor is invoked by Person A in the outbound context and AUDIOHOOK_INHERIT(MixMonitor)=yes is also set Person a transfers Person B to Person C (another asterisk peer) Person A is no longer involved in the call and the call is bridged between Person B and Person C The call recording stops as soon as Person A hangs up, even though AUDIOHOOK_INHERIT is set Is there any way we can get the entire call recorded in one file? Thanks in advance Ish Has anyone else encountered this as it's becoming a real problem. Does anyone know a way of getting continuity of call recording in this scenario? -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
