[App_rpt-users] SIP JOIN app_rpt

[Tim Sawyer]

I just add this to extensions.conf in my [sip-phones] context. Of course
substitute your own node number in place of my 2522 and 2533


; Allow calls to local nodes

exten => 2533,1,rpt(2533|P)

exten => 2522,1,rpt(2522|P)



On Fri, Dec 1, 2017 at 4:33 PM, Tim Sawyer <tisawyer at gmail.com> wrote:

> 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 at gmail.com> wrote:
>
>> Seems to me there are some names of SMEs that may be useful for queries.
>>
>> Thanks,
>> Bob
>> K6ECM
>> 73
>>
>>
>> On Dec 1, 2017, at 12:51 PM, Michael Lee Lockwood <
>> michael at lockwood.us.com> wrote:
>>
>> 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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>>
>>
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>>
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>>
>
>
>
> --
> --
> Tim
>



-- 
--
Tim
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