search for: master_channel

...n the g option where the caller hangs up the call. Looks like I can either use h or a hangup handler along with the shared function. On Tue, Dec 26, 2017 at 4:40 PM, Eric Wieling <ewieling at nyigc.com> wrote: > Don't use an 'h' extension, use a hangup handler. > Use the MASTER_CHANNEL() function to set variables to ensure they are > always set in the "top most" channel. Below is an untested example, but is > inspired by dialplan code I use in production. Maybe it will help. > > [outbound] ; this is called on the incoming (caller) channel > exten =>...

...-record:5] GotoIf("SIP/1010-00000162", "0?stopped") in new stack -- Executing [s at macro-one-touch-record:6] GotoIf("SIP/1010-00000162", "0?recording") in new stack -- Executing [s at macro-one-touch-record:7] Set("SIP/1010-00000162", "MASTER_CHANNEL(ONETOUCH_REC)=RECORDING") in new stack -- Executing [s at macro-one-touch-record:8] Set("SIP/1010-00000162", "MASTER_CHANNEL(REC_STATUS)=RECORDING") in new stack -- Executing [s at macro-one-touch-record:9] Set("SIP/1010-00000162", "AUDIOHOOK_INHERIT(...

Hi, I have a dial plan where I need to notify an external system when a call was answered and when the call hung up. In both requests the start time needs to be the same. My Dialplan looks something like this: [outbound] Exten => _X.,1,Dial(SIP/${EXTEN}@1.1.1.1,,U(call-answer-from-carrier)) Exten => h,1,NoOp(ANSWERED_TIME: ${ANSWEREDTIME} >>> DIAL_TIME: ${DIALEDTIME}

I think HANGUPCAUSE is channel agnostic. See: core show function HANGUPCAUSE Some thing like this IIRC: Set(my_cause=${HANGUPCAUSE(${CHANNEL(name)},tech)}) Remember the incoming leg of the call and the outgoing leg of the call are different channels. Make sure you are giving HANGUPCAUSE the correct channel. On 06/09/2018 02:01 PM, Khalil Khamlichi wrote: > It seems very weird to me

Greetings, Recently a performance regression in chan_sip was discovered in Asterisk 1.8. The regression is caused by chan_sip setting MASTER_CHANNEL(HASH(SIP_CAUSE,<chan name>)) after each response received on a channel. That feature has been made optional in the latest 1.8 SVN code, but is currently still enabled by default. After some internal discussion, we decided to consider disabling this feature by default in future 1.8 versions. T...

...e issues resolved in this release: * Added the 'storesipcause' option to sip.conf to allow the user to disable the setting of HASH(SIP_CAUSE,) on the channel. Having chan_sip set HASH(SIP_CAUSE,) on the channel carries a significant performance penalty because of the usage of the MASTER_CHANNEL() dialplan function. We've decided to disable this feature by default in future 1.8 versions. This would be an unexpected behavior change for anyone depending on that SIP_CAUSE update in their dialplan. Please refer to the asterisk-dev mailing list more information: http://l...

...e issues resolved in this release: * Added the 'storesipcause' option to sip.conf to allow the user to disable the setting of HASH(SIP_CAUSE,) on the channel. Having chan_sip set HASH(SIP_CAUSE,) on the channel carries a significant performance penalty because of the usage of the MASTER_CHANNEL() dialplan function. We've decided to disable this feature by default in future 1.8 versions. This would be an unexpected behavior change for anyone depending on that SIP_CAUSE update in their dialplan. Please refer to the asterisk-dev mailing list more information: http://l...

Hey all, I've been fighting with this all morning, and I feel like this should be a relatively simple task, but I just can't get it to work. I currently have a very basic asterisk v11.6 setup with a single extension (a Bria softphone) and a single sip trunk to my carrier. What I'm trying to accomplish is simply adding the asterisk generated SIPCALLID of the leg between asterisk and

B.H. Hello, i'm using AMI Originate action (with async=true) to send outgoing calls to a SIP trunk (using asterisk-java library to connect to AMI). The problem is that in case of failed originate, OriginateResponse event is returning only the reason code which is sometimes not sufficient to determine the real cause of failure. Also, there's no way to link between the channel that was

...LT=") in new stack -- Executing [s at macro-auto-blkvm:2] Macro("SIP/41712-00000166", "blkvm-clr,") in new stack At this point the original incoming call is answered. . . . -- Executing [s at macro-auto-blkvm:3] ExecIf("SIP/41712-00000166", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=41712)") in new stack And then transferred to 41720 [2013-07-03 13:43:03] WARNING[7954][C-00004685]: res_srtp.c:406 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 110 -- Stopped music on hold on DAHDI/1-1 == Extension Changed 41712[ext-local]...

...ed. (Reported by Tzafrir Cohen) * ASTERISK-27498 - ICE candidate parser - ICE foundation parsing too short (Reported by Michele Pr??) * ASTERISK-27366 - Asterisk Turkish Language Set Problem (Reported by Halil ??brahim YILDIZ) * ASTERISK-23133 - Documentation fix - MASTER_CHANNEL Unexpected Behaviour (Reported by Shane Mitchell) * ASTERISK-27531 - Compiler optimizations can break module load sequence. (Reported by abelbeck) * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without Contact crashes asterisk (Reported by Ross Beer) * A...

...ed. (Reported by Tzafrir Cohen) * ASTERISK-27498 - ICE candidate parser - ICE foundation parsing too short (Reported by Michele Pr??) * ASTERISK-27366 - Asterisk Turkish Language Set Problem (Reported by Halil ??brahim YILDIZ) * ASTERISK-23133 - Documentation fix - MASTER_CHANNEL Unexpected Behaviour (Reported by Shane Mitchell) * ASTERISK-27531 - Compiler optimizations can break module load sequence. (Reported by abelbeck) * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without Contact crashes asterisk (Reported by Ross Beer) * A...

...ed. (Reported by Tzafrir Cohen) * ASTERISK-27498 - ICE candidate parser - ICE foundation parsing too short (Reported by Michele Pr��) * ASTERISK-27366 - Asterisk Turkish Language Set Problem (Reported by Halil ��brahim YILDIZ) * ASTERISK-23133 - Documentation fix - MASTER_CHANNEL Unexpected Behaviour (Reported by Shane Mitchell) * ASTERISK-27531 - Compiler optimizations can break module load sequence. (Reported by abelbeck) * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without Contact crashes asterisk (Reported by Ross Beer) * A...

...too short (Reported by Michele Pr��) * ASTERISK-25128 - Datastore: Implement automatic module references. (Reported by Corey Farrell) * ASTERISK-27366 - Asterisk Turkish Language Set Problem (Reported by Halil ��brahim YILDIZ) * ASTERISK-23133 - Documentation fix - MASTER_CHANNEL Unexpected Behaviour (Reported by Shane Mitchell) * ASTERISK-27531 - Compiler optimizations can break module load sequence. (Reported by abelbeck) * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without Contact crashes asterisk (Reported by Ross Beer) * A...

...too short (Reported by Michele Pr��) * ASTERISK-25128 - Datastore: Implement automatic module references. (Reported by Corey Farrell) * ASTERISK-27366 - Asterisk Turkish Language Set Problem (Reported by Halil ��brahim YILDIZ) * ASTERISK-23133 - Documentation fix - MASTER_CHANNEL Unexpected Behaviour (Reported by Shane Mitchell) * ASTERISK-27531 - Compiler optimizations can break module load sequence. (Reported by abelbeck) * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without Contact crashes asterisk (Reported by Ross Beer) * A...

...rted by Corey Farrell) - [ASTERISK-27366 <https://issues.asterisk.org/jira/browse/ASTERISK-27366>] - Asterisk Turkish Language Set Problem (Reported by Halil İbrahim YILDIZ) - [ASTERISK-23133 <https://issues.asterisk.org/jira/browse/ASTERISK-23133>] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour (Reported by Shane Mitchell) - [ASTERISK-27531 <https://issues.asterisk.org/jira/browse/ASTERISK-27531>] - Compiler optimizations can break module load sequence. (Reported by abelbeck) - [ASTERISK-27480 <https://issues.asterisk.org/jira/browse/ASTERISK-2...