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

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