; Allow calls to local nodes
exten => 2533,1,rpt(2533|P)
exten => 2522,1,rpt(2522|P)
I just add this to extensions.conf in my [sip-phones] context.; Allow calls to local nodes
exten => 2533,1,rpt(2533|P)
exten => 2522,1,rpt(2522|P)
On Fri, Dec 1, 2017 at 12:53 PM, Bob Pyke <k6ecm1@gmail.com> wrote:Seems to me there are some names of SMEs that may be useful for queries.Thanks,BobK6ECM73Ahhh... some nice light reading! LOLI'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@gmail.com> wrote:I don’t know the answer, but thought of this.. http://www.asteriskdocs.org/en/3rd_Edition/asterisk-b ook-html/asterisk-book.html Thanks,BobK6ECM73I'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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