[App_rpt-users] SIP JOIN app_rpt

[Michael "Lee" Lockwood]

I'm hoping one of the Asterisk gurus out there can help me with this.

I have Asterisk running on a private node with app_rpt (DIAL).  That node
acts as a hub for other private nodes to connect to via IAX.  Everything
works as it should, with regards to audio being routed to all linked nodes.

Here's what I'm trying to accomplish:

1.  Hub node registers to a remote SIP server (no problem, I have that
covered).
2.  Once SIP registration occurs, hub node needs to automatically "dial" a
4 digit extension.
3.  The remote SIP server automatically answers and the audio is routed to
a SIP end appliance (no problem, that part is done by remote SIP server)
4.  Once the SIP call on the Hub node is "connected" to the remote SIP
server and the call is active, join that SIP call in to the existing linked
nodes audio bridge (DAHDI) so that node audio is passed to/from the SIP
call.

So, first off is this possible?  Secondly, who among this group can point
me in the direction to accomplish this?

Thanks in advance!
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.keekles.org/pipermail/app_rpt-users/attachments/20171201/71bd6054/attachment.html>