search for: record_status

...ress or say 1 to 5". If > no DMTF is pressed, I want to send the recording to Google Speech to > get the number back (got that part working already). > > If any dtmf key is pressed while Application_Record is running with > option y, then the recording terminates and sends > RECORD_STATUS of "DTMF" (A terminating DTMF was received). > > But I need to know **what** number that DTMF was, and I can't see a > way of grabbing it after the fact. > > I can see in the code where the right variables are.. > > https://github.com/asterisk/asterisk/blob/master...

...ressed, I want to send the recording to Google Speech to > >> get the number back (got that part working already). > >> > >> If any dtmf key is pressed while Application_Record is running with > >> option y, then the recording terminates and sends > >> RECORD_STATUS of "DTMF" (A terminating DTMF was received). > >> > >> But I need to know **what** number that DTMF was, and I can't see a > >> way of grabbing it after the fact. > >> > >> I can see in the code where the right variables are.. > >&g...

...d that it doesn't do DTMF: https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Application_SpeechBackground So then there's https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Application_Record, which can terminate on any DTMF key with "y", but according to the docs, "RECORD_STATUS" only sets a flag of "DTMF" (A terminating DTMF was received ('#' or '*', depending upon option 't')). So, I don't get to know which key was pressed via that method, either. There's very little information I can find about the built-in functions for s...

Thanks for your responses - it looks like I have the following options, in order of ease: 1: Modify and recompile app_record.c Change line 471 https://github.com/asterisk/asterisk/blob/master/apps/app_record.c#L471 from status_response = "DTMF"; to status_response = dtmf_integer; Pro: Free, easy Con: Have to remember to edit module each time a new Asterisk update comes out 2:

I don't see a method to detect the success or failure for the Record CMD. I'd like to know the reason why the recording ended Am I wrong? exten => recordmsg,1,Noop() exten => recordmsg,n,Record(${NEWPHRASEID}:ulaw|4|180) Bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

...dro Pimenta) * ASTERISK-27238 - Bridging: Crash freeing a frame that's already been freed (Reported by Richard Kenner) * ASTERISK-27412 - core: Audiohook freeing interpolated frame when it shouldn't. (Reported by Mikhail) * ASTERISK-27423 - app_record: We set the RECORD_STATUS channel variable before closing the file (Reported by George Joseph) * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk insert same ip address in "source ip address" and "destination ip address" fields in HEP packets (Reported by M...

...dro Pimenta) * ASTERISK-27238 - Bridging: Crash freeing a frame that's already been freed (Reported by Richard Kenner) * ASTERISK-27412 - core: Audiohook freeing interpolated frame when it shouldn't. (Reported by Mikhail) * ASTERISK-27423 - app_record: We set the RECORD_STATUS channel variable before closing the file (Reported by George Joseph) * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk insert same ip address in "source ip address" and "destination ip address" fields in HEP packets (Reported by M...

...dro Pimenta) * ASTERISK-27238 - Bridging: Crash freeing a frame that's already been freed (Reported by Richard Kenner) * ASTERISK-27412 - core: Audiohook freeing interpolated frame when it shouldn't. (Reported by Mikhail) * ASTERISK-27423 - app_record: We set the RECORD_STATUS channel variable before closing the file (Reported by George Joseph) * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk insert same ip address in "source ip address" and "destination ip address" fields in HEP packets (Reported by M...

...dro Pimenta) * ASTERISK-27238 - Bridging: Crash freeing a frame that's already been freed (Reported by Richard Kenner) * ASTERISK-27412 - core: Audiohook freeing interpolated frame when it shouldn't. (Reported by Mikhail) * ASTERISK-27423 - app_record: We set the RECORD_STATUS channel variable before closing the file (Reported by George Joseph) * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk insert same ip address in "source ip address" and "destination ip address" fields in HEP packets (Reported by M...

...dro Pimenta) * ASTERISK-27238 - Bridging: Crash freeing a frame that's already been freed (Reported by Richard Kenner) * ASTERISK-27412 - core: Audiohook freeing interpolated frame when it shouldn't. (Reported by Mikhail) * ASTERISK-27423 - app_record: We set the RECORD_STATUS channel variable before closing the file (Reported by George Joseph) * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk insert same ip address in "source ip address" and "destination ip address" fields in HEP packets (Reported by M...

...dro Pimenta) * ASTERISK-27238 - Bridging: Crash freeing a frame that's already been freed (Reported by Richard Kenner) * ASTERISK-27412 - core: Audiohook freeing interpolated frame when it shouldn't. (Reported by Mikhail) * ASTERISK-27423 - app_record: We set the RECORD_STATUS channel variable before closing the file (Reported by George Joseph) * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk insert same ip address in "source ip address" and "destination ip address" fields in HEP packets (Reported by M...

...dro Pimenta) * ASTERISK-27238 - Bridging: Crash freeing a frame that's already been freed (Reported by Richard Kenner) * ASTERISK-27412 - core: Audiohook freeing interpolated frame when it shouldn't. (Reported by Mikhail) * ASTERISK-27423 - app_record: We set the RECORD_STATUS channel variable before closing the file (Reported by George Joseph) * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk insert same ip address in "source ip address" and "destination ip address" fields in HEP packets (Reported by M...

...ner) - [ASTERISK-27412 <https://issues.asterisk.org/jira/browse/ASTERISK-27412>] - core: Audiohook freeing interpolated frame when it shouldn't. (Reported by Mikhail) - [ASTERISK-27423 <https://issues.asterisk.org/jira/browse/ASTERISK-27423>] - app_record: We set the RECORD_STATUS channel variable before closing the file (Reported by George Joseph) - [ASTERISK-26758 <https://issues.asterisk.org/jira/browse/ASTERISK-26758>] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip address in "source ip address" and "destination ip address"...