On Mon, Sep 19, 2016 at 8:34 AM, Jonas Kellens <[email protected]> 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
