[App_rpt-users] SIP JOIN app_rpt

Sun Dec 3 02:59:30 UTC 2017

I think that will work once the call is connected, or on an incoming call.
But, I need to have the app_rpt Asterisk machine "dial" out and initiate
the call.

On Fri, Dec 1, 2017 at 6:35 PM, Tim Sawyer <tisawyer at gmail.com> wrote:

> 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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>>>>
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>>>
>>>
>>>
>>> --
>>>
>>> *Michael "Lee" Lockwood*
>>> michael at lockwood.us.com
>>>
>>> _______________________________________________
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>>>
>>>
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>>>
>>
>>
>>
>> --
>> --
>> Tim
>>
>
>
>
> --
> --
> Tim
>
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>
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-- 

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