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


 
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jonas.kellens at telen...
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PostPosted: Sat Sep 17, 2016 4:47 am    Post subject: [asterisk-users] Ast 13.11.2 : bridgepeer variable empty ? Reply with quote

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@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@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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gmludo at gmail.com
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PostPosted: Sun Sep 18, 2016 12:59 pm    Post subject: [asterisk-users] Ast 13.11.2 : bridgepeer variable empty ? Reply with quote

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@telenet.be (jonas.kellens@telenet.be)>:
Quote:
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@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@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

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jonas.kellens at telen...
Guest





PostPosted: Mon Sep 19, 2016 8:43 am    Post subject: [asterisk-users] Ast 13.11.2 : bridgepeer variable empty ? Reply with quote

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:

Quote:
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@telenet.be (jonas.kellens@telenet.be)>:
Quote:
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@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@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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mjordan at digium.com
Guest





PostPosted: Mon Sep 19, 2016 3:57 pm    Post subject: [asterisk-users] Ast 13.11.2 : bridgepeer variable empty ? Reply with quote

On Mon, Sep 19, 2016 at 8:34 AM, Jonas Kellens <jonas.kellens@telenet.be> wrote:
Quote:
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.


It's intentional.

The BRIDGEPEER variable is set to the parties that you are bridged
with at that moment in time. As participants enter/leave a bridge, the
BRIDGEPEER variable gets set (up to some somewhat reasonable number).
When a channel leaves a bridge, it is removed from the BRIDGEPEER
list.

You can imagine then why the BRIDGEPEER variable isn't typically set
any longer when you are in the 'h' extension - the participants all
left.

Why did this change occur?

In Asterisk 12+, all bridging in Asterisk happens using a flexible
bridging framework. That framework accommodates not just two-party
bridges, but multi-party bridges as well. In fact, all bridges can be
turned into a multi-party bridge simply by adding additional channels.
That flexibility is pretty nice, and enables some pretty interesting
features. Unfortunately, it also makes the value of BRIDGEPEER
somewhat hard to predict. It's not hard to create a scenario where the
value of BRIDGEPEER - if we didn't remove parties that left a bridge -
becomes completely arbitrary.

So what is BRIDGEPEER good for?

It's pretty useful if you're building applications on top of Asterisk
outside of the dialplan. For example, using AMI, you can query that
channel variable to get a snapshot of who all you are in a bridge with
at that point in time.

Why wasn't DIALEDPEERNAME not affected in a similar fashion?

Mostly because dialling is still 'atomic' from the perspective of the
dialplan. When Dial ends, you presumably didn't perform 10 other dials
while that application was executing. Bridging isn't that way; phones
have the ability to manipulate the bridge themselves outside of
Asterisk's control (via attended transfers).

Matt

--
Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

--
_____________________________________________________________________
-- 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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gmludo at gmail.com
Guest





PostPosted: Mon Sep 19, 2016 6:29 pm    Post subject: [asterisk-users] Ast 13.11.2 : bridgepeer variable empty ? Reply with quote

Hi Matt,

Thank you for your explanation, it's clear to me.


Nevertheless, it doesn't help me in my use case: I'm trying to collect all SIP callid and store that with cdr_adaptive_odbc.
The business use case is to link Asterisk CDRs with logs from our operators via their API and our capture tool to give an access in our GUI.


I retrieve easily SIP callid of caller, but not easily for the callee.
For now, the best I have is to use the b option in Dial application and take that via a subroutine with:
same  =   n,ExecIf($[${LEN(${SIPCALLID})} > 0]?Set(CDR(callee_callid)=${SIPCALLID}))
same  =   n,ExecIf($[${LEN(${SIPCALLID})} = 0]?Set(CDR(callee_callid)=${PJSIP_HEADER(read,call-id)}))



It works, only if the remote party answers. If not, I don't retrieve callid.

Somebody has an idea with dialplan ? My B and C plan is to poll current channels via asterisk or to intercept callids with a kamailio, but it seems a little bit overkill, at least to me.


Thanks a lot for your answer.


Have a nice week.

--
Ludovic Gasc (GMLudo)

http://www.gmludo.eu/





2016-09-19 22:57 GMT+02:00 Matthew Jordan <mjordan@digium.com (mjordan@digium.com)>:
Quote:
On Mon, Sep 19, 2016 at 8:34 AM, Jonas Kellens <jonas.kellens@telenet.be (jonas.kellens@telenet.be)> wrote:
Quote:
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.


It's intentional.

The BRIDGEPEER variable is set to the parties that you are bridged
with at that moment in time. As participants enter/leave a bridge, the
BRIDGEPEER variable gets set (up to some somewhat reasonable number).
When a channel leaves a bridge, it is removed from the BRIDGEPEER
list.

You can imagine then why the BRIDGEPEER variable isn't typically set
any longer when you are in the 'h' extension - the participants all
left.

Why did this change occur?

In Asterisk 12+, all bridging in Asterisk happens using a flexible
bridging framework. That framework accommodates not just two-party
bridges, but multi-party bridges as well. In fact, all bridges can be
turned into a multi-party bridge simply by adding additional channels.
That flexibility is pretty nice, and enables some pretty interesting
features. Unfortunately, it also makes the value of BRIDGEPEER
somewhat hard to predict. It's not hard to create a scenario where the
value of BRIDGEPEER - if we didn't remove parties that left a bridge -
becomes completely arbitrary.

So what is BRIDGEPEER good for?

It's pretty useful if you're building applications on top of Asterisk
outside of the dialplan. For example, using AMI, you can query that
channel variable to get a snapshot of who all you are in a bridge with
at that point in time.

Why wasn't DIALEDPEERNAME not affected in a similar fashion?

Mostly because dialling is still 'atomic' from the perspective of the
dialplan. When Dial ends, you presumably didn't perform 10 other dials
while that application was executing. Bridging isn't that way; phones
have the ability to manipulate the bridge themselves outside of
Asterisk's control (via attended transfers).

Matt

--
Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

--
_____________________________________________________________________
-- 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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