[App_rpt-users] SIP JOIN app_rpt

[Michael "Lee" Lockwood]

Ahhh... some nice light reading!  LOL

I'll look through this, thank you.  I've been burning up Google, but
haven't found anything definitive that will work.

On Fri, Dec 1, 2017 at 2:24 PM, Bob Pyke <k6ecm1 at gmail.com> wrote:

> I don’t know the answer, but thought of this.. http://www.
> asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html
>
> Thanks,
> Bob
> K6ECM
> 73
>
>
> On Dec 1, 2017, at 10:02 AM, Michael Lee Lockwood <michael at lockwood.us.com>
> wrote:
>
> 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!
>
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-- 

*Michael "Lee" Lockwood*
michael at lockwood.us.com
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