On Tue, Aug 24, 2021 at 1:27 PM Nikša Baldun <[email protected]> wrote:
> The thing is, when attended transfer happens, Asterisk connects previously > unrelated channels without going to the dialplan. I keep a number of > variables which help me track call state on both channels, and their values > become obsolete on attended transfer, as channels are now in a completely > new call. My idea was to gosub both sides to a dialplan context so I can > refresh call state variables. But that has proven difficult if one of the > calls is not answered. I am certain this can't be solved by straight > dialplan, not without modifying Asterisk code. I haven't delved into ARI > yet. I'd rather avoid it, unless it is the only option. > I'm not sure this would work for you but ... Have app_dial create a channel variable on the incoming channel that has the list of channels being dialed. You could also use the DB functions to store key value pairs and have pre-dial-handlers on the outgoing channels store their IDs there keyed by the incoming channel. > On 24. 08. 2021. 20:51, Kevin Harwell wrote: > > What's the overall scenario you are trying to solve? Perhaps there is > another way to do what you want to do without even modifying Asterisk code? > For example, maybe this is something an ARI application could handle, or > even straight dialplan using a combination of app_dial, pre-dial handlers, > and such. > > On Mon, Aug 23, 2021 at 5:29 AM Nikša Baldun <[email protected]> wrote: > >> Hello, >> >> I am trying to modify bridge.c (function ast_bridge_transfer_attended) >> in order to send channels involved in SIP attended transfer to the >> dialplan. If both transferee and transfer target are bridged, that is >> relatively easy. However, if transfer target is ringing, I don't know >> how to find B-leg channels (there could be multiple, I suppose). So the >> question is, having a reference to A-leg channel, how to obtain a list >> of B-leg channels? >> >> Best regards, >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-dev mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-dev > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev
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