[App_rpt-users] Calling into the node operator's local SIP phone

Keith Williamson hkwilliamson at gmail.com
Sun Feb 7 21:28:26 UTC 2010


On Sun, Feb 7, 2010 at 1:53 PM, Stephen - K1LNX <k1lnx at k1lnx.net> wrote:

> Keith,
>     This works like a champ! I just so happened to re-flash one of my Cisco
> phones back to SIP this afternoon for playing with and the timing could not
> have been more perfect lol. A very very useful feature!
>
> 73
> Stephen
> K1LNX
>
>
> On Sun, Feb 7, 2010 at 2:29 PM, Keith Williamson <hkwilliamson at gmail.com>wrote:
>
>> Hi,
>>
>> Several people have expressed an interest in how to configure Allstar to
>> allow radio users to connect to the node-operator's local SIP phone. It
>> turns out it's pretty easy (once you have a local SIP phone configured of
>> course). To do this without configuring more general autopatch access with
>> it's potential risks, you can create a specific context for just allowing a
>> radio user to access one "outbound" SIP connection, the local SIP phone. In
>> rpt.conf, I uncommented the "autopatchdn" function and duplicated and
>> uncommented the "autopatchup" function. These two functions are in the
>> [functions] stanza available to radio users. In the new "autopatchup"
>> function, I changed the default DTMF string from 6 to 61 and added the
>> "context=" option. I set the option to "context=node-op". So the function
>> now looks like this:
>>
>> 61=autopatchup,context=node-op,noct=1,farenddisconnect=1,dialtime=20000
>>
>> Then in extensions.conf, I added a stanza for [node-op]:
>>
>> [node-op]
>> exten => 1,1,Answer
>> exten => 1,n,Dial(SIP/200,10)
>> exten => 1,n,Playback(vm-nobodyavail)
>> exten => 1,n,Hangup
>>
>> Change the SIP/200 above to SIP/whatever-your-local-extension-is. Since we
>> modified rpt.conf, you need to restart asterisk.
>>
>> Now, if the radio user keys in *611, the autopatch will be invoked and
>> extension "1" in context [node-op] will be called where it will be answered
>> and then will dial the local SIP phone extension. If you don't pickup, it
>> will timeout, play the "nobody available to take your call" message, and
>> hangup.
>>
>> Of course you can integrate this into a full autopatch configuration by
>> modifying the default dialplan in context "radio" but I'm not willing to
>> open my node up to dialing out to the world (yet).
>>
>> 73's,
>>
>> Keith
>> KF7DRV
>>
>>
>> _______________________________________________
>> App_rpt-users mailing list
>> App_rpt-users at qrvc.com
>> http://qrvc.com/mailman/listinfo/app_rpt-users
>>
>>
>
>
> --
> **********************************
> Stephen Brown - ARS K1LNX
> Johnson City, TN EM86
> http://www.k1lnx.net
> google voice: 423-665-9367
> **********************************
>

Yeah, isn't allstar/asterisk the greatest? It's like Lego's for computers,
radios, and telephones.

Cheers,

Keith
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.keekles.org/pipermail/app_rpt-users/attachments/20100207/ec20f1af/attachment.html>


More information about the App_rpt-users mailing list