Recording channel does not hang up

[alerios]

Hi.

I'm using vicidial 2.0.5, with asterisk 1.4.24.1.

I've been using app_conference, because I use only SIP trunks and have no PCI timing cards. The system has worked fine for several weeks until this one.

At random times, some of the recording channels get stuck and won't hangup when the outgoing call hangs up. The stuck channel looks like this in show channels concise:

Code: Select all

Local/58600060@default-442b,2!default!58600060!1!Up!Conference!8600060!!!3!1091!(None)
Local/58600060@default-442b,1!default!8309!3!Up!Wait!3600!!!3!1091!(None)

How can I force the hanging of this Monitor calls when the outgoing call hangs-up? does AST_conf_update.pl has something to do with this?

here is the moment when the outgoing call hangsup:

Code: Select all

<------------>
^[[1;30;40m  == ^[[0;37;40mSpawn extension (macro-mexico2, s, 14) exited non-zero on 'Local/8600060@default-299e,1' in macro 'mexico2'
^[[1;30;40m  == ^[[0;37;40mSpawn extension (real-out-marcador, 524777621625, 5) exited non-zero on 'Local/8600060@default-299e,1'
^[[1;30;40m    -- ^[[0;37;40mExecuting [h@real-out-marcador:1] ^[[1;36;40mDeadAGI^[[0;37;40m("^[[1;35;40mLocal/8600060@default-299e,1^[[0;37;40m", "^[[1;35;40magi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----51-----19)^[[0;37;40m") in new stack
^[[1;30;40m    -- ^[[0;37;40mAGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----51-----19) completed, returning 0

[/code]

[mflorell]

We do not recommend using app_conference, it is less stable with a much higher possibility of audio issues than meetme. Under load it will loose audio unlike meetme.

[alerios]

I know you don't recommend app_conference, but sometimes app_meetme just can't be used. I've had several problems with some machines that get blocked due to zt/dadhi dummy incompatibilities with the rtc module. Also, sometimes you just can't use a timing card.

So, while someone can help with the record hang up problem, I'm using this awful hack to live with it:

Code: Select all

exten=h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME}))
exten=h,n,Set(conf=${CUT(CHANNEL,@,1)})
exten=h,n,Set(conf=${CUT(conf,/,2)})
exten=h,n,System(asterisk -rx "conference kick ${conf} 1")
exten=h,n,System(asterisk -rx "conference kick ${conf} 2")
exten=h,n,System(asterisk -rx "conference kick ${conf} 3")


[mflorell]

We have never run into a case where we cannot get zaptel/dahdi running. under what circumstances are you having this problem?

[alerios]

Hi.

The issue was recently solved, and now I can use dadhi_dummy with the last dahdi version: 2.2.0

More info: https://issues.asterisk.org/view.php?id=13930