Tom,

Looks like you understood the dialing right away. Some take a bit longer to understand it.
Just wanted to add a note to those that set this up. And to make folks think a bit more to possibilities of what can be done.

If you create many extensions, you can also create "callgroups". So by example, someone dials a "0" could be made to ring all in the callgroup for a control operator etc. You could make a macro with whatever command you wanted to ring the same callgroup. Any Multiple of extensions with "callgroup=????" in common.

These are functions of Asterisk and not app_rpt (except the macro) . So get a copy of the "Asterisk manual" as you will find many useful things you can create yourself on the fly. Extensions do not need to be a sip phone. They can be made into a function you create for your repeater. Your imagination is the limit. Asterisk is flexible and powerful.

Hope this encourages exploration,
...mike/kb8jnm




-------- Original Message --------
Subject: AW: AW: [App_rpt-users] Connect SIP Phone to allstarlink box
From: "torben@klimt-online.com" <torben@klimt-online.com>
Date: Wed, May 21, 2014 2:55 am
To: "app_rpt-users@ohnosec.org" <app_rpt-users@ohnosec.org>
Cc: "mike@midnighteng.com" <mike@midnighteng.com>

hi all,
sorry for the late answer but i sleep when you write,Hi......
@yes mike, it works fine
@robert, i try to explain with my words what i do for working - if that is the right way i don`t know ;-)
my english is not the best but i hope you can understand what i write
 
greetings from bavaria tom dh6mbt

1. Setup SIP-Phones in the sip.conf like mike write
[6010]
deny=0.0.0.0/0.0.0.0
username=6010
secret=6010
dtmfmode=rfc2833
canreinvite=no
context=radio-control
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp
encryption=no
callgroup=
pickupgroup=
dial=SIP/6010
mailbox=6010@device
permit=0.0.0.0/0.0.0.0
callerid=DH6MBT <6010>
do it also with [6020]

2. edit extension.conf for dial in the repeater my repeater has the nodenr 28175
[radio-control]
exten => 28175,1,Answer
exten => 28175,n,Wait(2)
exten => 28175,n,Playback(rpt/node)
exten => 28175,n,Playback(digits/2)
exten => 28175,n,Playback(digits/8)
exten => 28175,n,Playback(digits/1)
exten => 28175,n,Playback(digits/7)
exten => 28175,n,Playback(digits/5)
exten => 28175,n,rpt,28175|Pv
Pv stand for VOX by the phone, no other DTMF dials are needed
if you take P instead of Pv you must dial *99 for talking and # to stop PTT
3. edit the extension.conf to call from each SIP-phone to another
exten => 6010,1,Dial(SIP/6010)
exten => 6020,1,Dial(SIP/6020)
3. edit the extension.conf to call from the repeater to the SIP-phone
[autopatch]
exten => 6010,1,Dial(SIP/6010)
exten => 6020,1,Dial(SIP/6020)

now you can talk from SIp to SIP by callig 6010 or 6020
you can call from Sip the the repeater by dialing 28175
and  you can call your SIP from teh repeater by dialing *66010 or *66020 - hangup by *0