[asterisk-users] AsyncAGI - How to jump in dial plan when no action initiated on channel or AMI user is disconnected

[Amit Patkar]

Hi

Is there any way to detect inactivity on channel when AsyncAGI is used?
I want to detect whether application handling calls using AMI & AGI has 
stopped responding.


Alternatively, how can dialplan check if there is any AMI user connected 
and decide dial plan execution?


Thanks & Regards,
Amit Patkar

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[asterisk-users] Ast 13.11.2 : bridgepeer variable empty ?

[Jonas Kellens]

Hello

a call goes out and is answered :

[Sep 17 11:29:52] VERBOSE[23420][C-0051] app_dial.c: 
SIP/myprovider-010b is making progress passing it to 
SIP/mysippeer-0108
[Sep 17 11:30:05] VERBOSE[23420][C-0051] app_dial.c: 
SIP/myprovider-010b answered SIP/mysippeer-0108
[Sep 17 11:30:05] VERBOSE[23522][C-0051] bridge_channel.c: Channel 
SIP/myprovider-010b joined 'simple_bridge' basic-bridge 

[Sep 17 11:30:05] VERBOSE[23420][C-0051] bridge_channel.c: Channel 
SIP/mysippeer-0108 joined 'simple_bridge' basic-bridge 



Call ends :
[Sep 17 11:34:36] VERBOSE[23420][C-0051] bridge_channel.c: Channel 
SIP/mysippeer-0108 left 'simple_bridge' basic-bridge 

[Sep 17 11:34:36] VERBOSE[23522][C-0051] bridge_channel.c: Channel 
SIP/myprovider-010b left 'simple_bridge' basic-bridge 





When the call ends in Asterisk version 1.8.32.3 I can read the variable 
in h-context.

In Asterisk 13.11.2 this variable is always empty. How come ??


Dialplan logic :
...
exten => h,n,NoOp(bridgepeer = ${BRIDGEPEER})
...


CLI on Asterisk 13.11.2 :
 -- Executing [h@calling:15] NoOp("SIP/mysippeer-4c80", "bridgepeer 
= SIP/myprovider-4c83") in new stack



CLI on Asterisk 13.11.2 :
VERBOSE[23420][C-0051] pbx.c: Executing [h@calling:15] 
NoOp("SIP/mysippeer-0108", "bridgepeer = ") in new stack



What has changed and how can I get the 13.11 version of ${BRIDGEPEER} ??





Thanks in advance !


Kind regards.

Jonas.
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[asterisk-users] Asterisk 13 + PJSIP softphone issues

[Madushan Geethanga]

Hi,

Do all sip phones supports Asterisk 13 + PJSIP?. I'm having issues of
connecting ekiga softphone  and Snom 710 to Asterisk. zoiper phones works
fine. Asterisk is behind NAT. The phone sends register with expires 0.

Best Regards,
Madushan
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