IP Phone and Creating Extensions

App_rpt-users
1

Hi Everyone,

I’ve been playing with a Polycom SoundPoint IP 331 and finally got it connecting to my nodes on my server (yea!). Now I’m trying to figure out how to create extensions in the extensions.conf file to do two specific things.

  • First, from the IP phone directly dial another one of my nodes that is attached to a different server. In this case specifically, Node # 49935.

  • Second, how to generate an extension that allows the phone to be “called” directly from a node radio via DTMF. For example, to dial 1999 into my node from the DTMF pad and be connected to the IP phone.

Can anyone who is more versed in Asterisk than I am please give me some advice on this?

Thanks in advance, and 73s!

2

Hi, although I’m not an astareisk expert, i will try to point you in the right direction.

  1. In this case, I think that you will need to create an IAX trunk between the two boxes running allstarlink and then create an entry in your extensions.conf to dial the number of the second node over the trunk.
  2. If you check the stock extensions.conf included in the ASL image you will find a section that you must comment/uncomment to enable the phone patch (in my case lines 44 to 52):
; Comment-out the following clause if you want Allstar Autopatch service
[pstn-out]
exten = _NXXNXXXXXX,1,playback(ss-noservice)
exten = _NXXNXXXXXX,2,Congestion

; Un-comment out the following clause if you want Allstar Autopatch service
;[pstn-out]
;exten = _NXXNXXXXXX,1,Dial(IAX2/allstar-autopatch/\${EXTEN})
;exten = _NXXNXXXXXX,2,Busy

You will probably need to make some adjustments in your extensions.conf depending on the contexts you used for your local phone extensions.

I hope this helps.

Best.

Gustavo
LU7WA

3

Generally speaking, no custom…

Set ‘context’ to radio-contol in your sip extensions.
Now, everything you want to connect to should be inside that context in extensions.conf.

To connect to one of your nodes, create a stanza inside that ext context…

[radio-control]

exten => 29285,1,Rpt,29285|P ; is all it takes, but you might want to get fancier…

exten => 29285,1,Answer
exten => 29285,n,Wait(2)
exten => 29285,n,Playback(rpt/node)
exten => 29285,n,Playback(/var/lib/asterisk/sounds/rpt/nodenames/29285) ; plays rec file nodename
;exten => 29285,n,Playback(/etc/asterisk/msg/idmsg)
;exten => 29285,n,SayAlpha(fm)
;exten => 29285,n,Playback(remote-base)
exten => 29285,n,Rpt,29285|P ; the pipe P "|P’ says to handle this radio extension with ‘phone rules’

For internal extension patch, in rpt.conf
assuming you do not have any outbound routes and only your sip phones…
set the context the same as your sips

6=autopatchup,context=radio-control,noct=0,farenddisconnect=1,dialtime=90000,quiet=0

as long as your sips have correct routing in context [radio-control] all is fine

…and provided I did not forget anything… LOL and I did…
edit
I was assuming you made dialplan for your extensions.

roughly it needs something similar to this with your ext #'s
exten => _50X,n,Dial(sip/${EXTEN},20) ;dial ext on this box, dial 20 sec then next voicemail
exten => _50X,n,VoiceMail(5000) ; common vm
exten => _50X,n,Hangup

should be in the same context listed with your sips

4

Sorry, I did not notice the ‘different server’ in the original request.

You will need to create either a iax or sip trunk between the two servers.
Then it is a matter of dialing using the created trunk.

I have type this instruction to many times in the various forums we have had here, so I’m not spending the time to do it again. It was in our help pages at one time… take a look in the wiki.
You can get more direct help with that by google search as it is normal asterisk stuff.

5

Thanks to everyone for the input. While Ive been playing with AllStar for about a year and a half now, Iḿ still very new to the telephony side of things and am now trying to integrate that more into my existing system.

I did get the IAX trunk up and running between the two servers and Iĺl be playing with some of the autopatch stuff over this weekend. I am having a problem with one of the Polycom phones Im using but Iĺl bring that up in a separate post.

Thanks again everyone and 73s! -Tom

6

Tom, it would be important to note that some phones are a bit proprietary and may need set-up a bit different and/or have or lack of features in asterisk. Often there are workarounds.

The best advise I can give you is do a google search with the terms asterisk and your phone make and model and perhaps a one word description of the problem.

You will see you are not alone !