How do I call SIP phones on Allstar system from a radio?

AllStarLink
1

I know this is a simple process to set up, and has probably been asked and answered many times, but it is eluding me. I want to be able to place calls to SIP phones on my Allstar/Asterisk system from a radio. I can access other features of my node from the radio, like *711 or *712, or *3node to connect to other Allstar nodes without issue.

For example, I have a SIP phone on my system of extension 602, and it can call other SIP phones on the system just fine, and can be called by other SIP phones. I've tried to use the autopatch feature from a radio but it doesn't seem to work; my system's rpt.conf shows an autopatchup code of 61, but when I dial *61602 nothing happens, no response from my Allstar node back to the radio.

I am sure I am missing some other configuration settings in other CONF files, just can't figure out what those are. Any help appreciated. Thanks.

2

This is all down to your dial plan as defined in extensions.conf.

Basically, you will have to explain to asterisk how to get to whatever "602" is.

In my case, I have some SIP extensions (101 and 102) accessable like this:

exten => 101,1,Dial(SIP/101)
exten => 102,1,Dial(SIP/102)

You can also do two at once, as a sort of "ring all these phones":

exten => _X.,n,Dial(SIP/101&SIP/102,60,r(ring))

These need to go in extensions.conf within the appropriate context for outgoing autopatch calls. I've copied and pasted for you small bits of my extensions file. You will need to work out exactly where to place these within yours.

--E
de W6EL

3

Bill, To expand on what @w6el mentioned,
Provided you have a valid phone patch string, (rpt.conf)
61=autopatchup,context=radio,noct=1,farenddisconnect=1,dialtime=20000,quiet=1

You need to have a route for that in extensions.conf
Think of that file as a router. Making the connections to a endpoint.
That endpoint must be a valid extension with authentication in sip,iax,rpt config files. Meaning it must be present and logged in to the system.

And many folks can get that far with some head scratching.

But often, they do not use the correct context in extensions.conf as described in that autopatch string. When you make a call with it, it goes to that context in extensions.conf and looks for a matching string dialed.
It may be necessary to change that context for specific purposes.
You might have the entry in there somewhere, but not in the proper context.

So, somewhere in [radio} in extensions.conf, you would then describe the route to that sip phone, for example...

[radio] ; example context
exten => _50X,1,SetMusicOnHold(default)    
exten => _50X,n,Dial(sip/${EXTEN},20)        ;dial ext/functions on this box
exten => _50X,n,VoiceMail(5000)              ; common vm
exten => _50X,n,Hangup

This would dial any exact sip phone whos extension is 500 - 509
When you have more extensions, you will need to expand your learning on dialplan.
Yes, a few other things injected in there for you to think about.
Hope you get the gist of them without confusion.

exten => _50X,1,Dial(sip/${EXTEN},20)        ;dial ext/functions on this box
exten => _50X,n,Hangup

The old default was [radio] and I'm not sure if that is the same with asl3 as I use my old config files and even those are not defaulted with recent versions.

So, just some more things to pay attention to.

Hope that helps some.

1 Like
4

Thanks for the info guys, I did have to make some mods to my extensions.conf and rpt.conf to match what you sent, but still no go. I am not entirely sure I have the correct config to make the autopatch function work. Here is what I have in rpt.conf for the autopatch code; is there anything else that needs to be done to allow autopatch to work?

61 = autopatchup,context=radio,noct = 1,farenddisconnect = 1,dialtime = 20000 ; Autopatch up
62 = autopatchdn ; Autopatch down

And this is what I have in extensions.conf for my current extensions;

[radio]

exten=602,1,Dial (SIP/602,30)
exten=602,2,Hangup

exten=610,1,Dial (SIP/610,30)
exten=610,2,Hangup

I get on my HT, on the freq of my node radio, and key up and input the following;

*61602 or *61610

I get only the low tone, standard courtesy beep back, and nothing after that. I am thinking that I don't have the correct config to properly activate the autopatch feature. Is there any other config I need in rpt.conf to fully activate the autopatch feature? I don't intend to do any other autopatch functions other than calling SIP phones on my system, no other outdialing, as to the PSTN or the like. Thanks.

5

You are missing the
,quiet=1
at the end

61 = autopatchup,context=radio,noct=1,farenddisconnect=1,dialtime=20000,quiet=1

There has been a issue with dahdi for several versions playing the dialtone, so it is a workaround.

You should be watching asterisk in the foreground to see the actions taken.
and know errors, if any

asterisk -rvvv

6

Thanks Mike, yes, I just read some older posts about the quiet=1 setting, and when I added that it worked just as it should have...well, that is after I found one fat finger on my part. In the [radio] context, for each extension, the "Dial" option had one, apparently un-needed space after it, so I removed the space after "Dial", and added the quiet=1 line and it works as it should. It's a simplex node but the VOX on the SIP phone side works just fine. There is a bit of timing you need to be aware of from the radio side, but that's a small inconvenience to deal with.

Thanks to you and W6EL for the info...I was so close to getting it to work, but definitely needed that additional info from you guys!