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Jonas Kellens

2016-Sep-17 09:47 UTC

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

Hello

a call goes out and is answered :

[Sep 17 11:29:52] VERBOSE[23420][C-00000051] app_dial.c: 
SIP/myprovider-0000010b is making progress passing it to 
SIP/mysippeer-00000108
[Sep 17 11:30:05] VERBOSE[23420][C-00000051] app_dial.c: 
SIP/myprovider-0000010b answered SIP/mysippeer-00000108
[Sep 17 11:30:05] VERBOSE[23522][C-00000051] bridge_channel.c: Channel 
SIP/myprovider-0000010b joined 'simple_bridge' basic-bridge 
<ab233c52-249f-4370-bcdd-3eb9af7c6cfd>
[Sep 17 11:30:05] VERBOSE[23420][C-00000051] bridge_channel.c: Channel 
SIP/mysippeer-00000108 joined 'simple_bridge' basic-bridge 
<ab233c52-249f-4370-bcdd-3eb9af7c6cfd>

Call ends :
[Sep 17 11:34:36] VERBOSE[23420][C-00000051] bridge_channel.c: Channel 
SIP/mysippeer-00000108 left 'simple_bridge' basic-bridge 
<ab233c52-249f-4370-bcdd-3eb9af7c6cfd>
[Sep 17 11:34:36] VERBOSE[23522][C-00000051] bridge_channel.c: Channel 
SIP/myprovider-0000010b left 'simple_bridge' basic-bridge 
<ab233c52-249f-4370-bcdd-3eb9af7c6cfd>



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 at calling:15] NoOp("SIP/mysippeer-00004c80",
"bridgepeer
= SIP/myprovider-00004c83") in new stack


CLI on Asterisk 13.11.2 :
VERBOSE[23420][C-00000051] pbx.c: Executing [h at calling:15] 
NoOp("SIP/mysippeer-00000108", "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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Ludovic Gasc

2016-Sep-18 17:58 UTC

head link

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

Hi,

You might use DIALEDPEERNAME instead of BRIDGEPEER.

Nevertheless, I've the same issue with another BRIDGE prefix variable:

I never retrieve at one moment BRIDGEPVTCALLID variable, even if it's
documented in Asterisk wiki:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables

Nevertheless, the variable seems to be set in the Asterisk source code:
https://github.com/asterisk/asterisk/blob/13.10/main/bridge.c#L1222
I see no issues open about that, do I need to open an issue ?

Have a nice week.
--
Ludovic Gasc (GMLudo)
http://www.gmludo.eu/

2016-09-17 11:47 GMT+02:00 Jonas Kellens <jonas.kellens at telenet.be>:
> Hello
>
> a call goes out and is answered :
>
> [Sep 17 11:29:52] VERBOSE[23420][C-00000051] app_dial.c:
> SIP/myprovider-0000010b is making progress passing it to
> SIP/mysippeer-00000108
> [Sep 17 11:30:05] VERBOSE[23420][C-00000051] app_dial.c:
> SIP/myprovider-0000010b answered SIP/mysippeer-00000108
> [Sep 17 11:30:05] VERBOSE[23522][C-00000051] bridge_channel.c: Channel
> SIP/myprovider-0000010b joined 'simple_bridge' basic-bridge
> <ab233c52-249f-4370-bcdd-3eb9af7c6cfd>
> [Sep 17 11:30:05] VERBOSE[23420][C-00000051] bridge_channel.c: Channel
> SIP/mysippeer-00000108 joined 'simple_bridge' basic-bridge
> <ab233c52-249f-4370-bcdd-3eb9af7c6cfd>
>
> Call ends :
> [Sep 17 11:34:36] VERBOSE[23420][C-00000051] bridge_channel.c: Channel
> SIP/mysippeer-00000108 left 'simple_bridge' basic-bridge
> <ab233c52-249f-4370-bcdd-3eb9af7c6cfd>
> [Sep 17 11:34:36] VERBOSE[23522][C-00000051] bridge_channel.c: Channel
> SIP/myprovider-0000010b left 'simple_bridge' basic-bridge
> <ab233c52-249f-4370-bcdd-3eb9af7c6cfd>
>
>
>
> 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 at calling:15] NoOp("SIP/mysippeer-00004c80",
"bridgepeer > SIP/myprovider-00004c83") in new stack
>
>
> CLI on Asterisk 13.11.2 :
> VERBOSE[23420][C-00000051] pbx.c: Executing [h at calling:15]
> NoOp("SIP/mysippeer-00000108", "bridgepeer = ") in new
stack
>
>
> What has changed and how can I get the 13.11 version of ${BRIDGEPEER} ??
>
>
>
>
>
> Thanks in advance !
>
>
> Kind regards.
>
> Jonas.
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>       http://www.asterisk.org/community/astricon-user-conference
>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
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Jonas Kellens

2016-Sep-19 13:34 UTC

head link

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

Hello

I can confirm that the variable DIALEDPEERNAME contains the information 
that I would expect in the variable BRIDGEPEER.

But I read nowhere that DIALEDPEERNAME has replaced BRIDGEPEER as of 
Asterisk version 13 ?!

So if this is not the intention, then yes this is probably a bug and 
should be reported.




Kind regards.

Jonas.



On 18-09-16 19:58, Ludovic Gasc wrote:> Hi,
>
> You might use DIALEDPEERNAME instead of BRIDGEPEER.
>
> Nevertheless, I've the same issue with another BRIDGE prefix variable:
>
> I never retrieve at one moment BRIDGEPVTCALLID variable, even if it's 
> documented in Asterisk wiki: 
>
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables
>
> Nevertheless, the variable seems to be set in the Asterisk source code:
> https://github.com/asterisk/asterisk/blob/13.10/main/bridge.c#L1222
> I see no issues open about that, do I need to open an issue ?
>
> Have a nice week.
> --
> Ludovic Gasc (GMLudo)
> http://www.gmludo.eu/
>
> 2016-09-17 11:47 GMT+02:00 Jonas Kellens <jonas.kellens at telenet.be 
> <mailto:jonas.kellens at telenet.be>>:
>
>     Hello
>
>     a call goes out and is answered :
>
>     [Sep 17 11:29:52] VERBOSE[23420][C-00000051] app_dial.c:
>     SIP/myprovider-0000010b is making progress passing it to
>     SIP/mysippeer-00000108
>     [Sep 17 11:30:05] VERBOSE[23420][C-00000051] app_dial.c:
>     SIP/myprovider-0000010b answered SIP/mysippeer-00000108
>     [Sep 17 11:30:05] VERBOSE[23522][C-00000051] bridge_channel.c:
>     Channel SIP/myprovider-0000010b joined 'simple_bridge'
>     basic-bridge <ab233c52-249f-4370-bcdd-3eb9af7c6cfd>
>     [Sep 17 11:30:05] VERBOSE[23420][C-00000051] bridge_channel.c:
>     Channel SIP/mysippeer-00000108 joined 'simple_bridge'
basic-bridge
>     <ab233c52-249f-4370-bcdd-3eb9af7c6cfd>
>
>     Call ends :
>     [Sep 17 11:34:36] VERBOSE[23420][C-00000051] bridge_channel.c:
>     Channel SIP/mysippeer-00000108 left 'simple_bridge'
basic-bridge
>     <ab233c52-249f-4370-bcdd-3eb9af7c6cfd>
>     [Sep 17 11:34:36] VERBOSE[23522][C-00000051] bridge_channel.c:
>     Channel SIP/myprovider-0000010b left 'simple_bridge'
basic-bridge
>     <ab233c52-249f-4370-bcdd-3eb9af7c6cfd>
>
>
>
>     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 at calling:15]
NoOp("SIP/mysippeer-00004c80",
>     "bridgepeer = SIP/myprovider-00004c83") in new stack
>
>
>     CLI on Asterisk 13.11.2 :
>     VERBOSE[23420][C-00000051] pbx.c: Executing [h at calling:15]
>     NoOp("SIP/mysippeer-00000108", "bridgepeer = ") in
new stack
>
>
>     What has changed and how can I get the 13.11 version of
>     ${BRIDGEPEER} ??
>
>
>
>
>
>     Thanks in advance !
>
>
>     Kind regards.
>
>     Jonas.
>
>     --
>     _____________________________________________________________________
>     -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
>     Join the Asterisk Community at the 13th AstriCon, September 27-29,
>     2016
>     http://www.asterisk.org/community/astricon-user-conference
>     <http://www.asterisk.org/community/astricon-user-conference>
>
>     New to Asterisk? Start here:
>     https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>     <https://wiki.asterisk.org/wiki/display/AST/Getting+Started>
>
>     asterisk-users mailing list
>     To UNSUBSCRIBE or update options visit:
>     http://lists.digium.com/mailman/listinfo/asterisk-users
>     <http://lists.digium.com/mailman/listinfo/asterisk-users>
>
>
>
>
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