Calling into the node operator's local SIP phone

App_rpt-users
1

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

2

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

Cheers,

Keith

···

On Sun, Feb 7, 2010 at 1:53 PM, Stephen - K1LNX k1lnx@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@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

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–

Stephen Brown - ARS K1LNX
Johnson City, TN EM86
http://www.k1lnx.net

google voice: 423-665-9367

3

I was thinking more like erector-set. Solid metal, instead of plastic!

There is no other way of doing some of the stuff I have my box doing.

Asterisk and Linux with that magic app_rpt and the chan_usb just making

life possible for my repeater. It wouldn’t exist without those tools.

I don’t think it would be possible with IRLP, especially because of

their “rules”. My operation was called “illegal” by the IRLP guys

and I got kick/banned a couple times because I came in via my

sip phone instead of rf.

I can’t even begin to describe the handiness of being able to use a wifi sip

phone and get into my repeater, connect it via echolink or allstar and

have a converstation with someone on the other end in RF.

Linking repeaters, sip/iax phones in/out, reverse patch, autopatch, cepstral

voice for an unlimited number of things, remote base, shell scripts…

leaves us only to the limit of our imagination.

Thanks again Jim, the Steves’, Mark, Scott, and anyone else who has had their hands

in the code.

···

–

Don Russell, CBRE
Director of IT/Chief Operator - Maverick Media
W9DRR - ARRL OES, Technical Specialist
Winnebago County AEC
http://www.socialengineer.us