[arm-allstar] SIP phone

[iabarry Buel]

I've tried this before with no luck. Hoping someone can get me over the 
rock in the road.

Here are the references I'm looking at:
    https://w2ymm.home.blog/allstar-howto/#BM4      The SIP phone 
section is about 30% down the page
http://lists.hamvoip.org/pipermail/arm-allstar/2020-April/014644.html

A couple of questions:

 1. Is the SIP phone it's own unique NODE?
    If yes, then it appears to need a stanza in sip.conf
    Do I need to tell allstarlink that it is a sip phone?
 2. OR Should the SIP be the same node # as the already existing node?

I have:

 1. A new Grandstream SIP phone configured to point to the IP of my pi.
    It gives me a dial tone and I can see its IP address on the LCD.
 2. A pi 3 running a single hamvoip node. It works fine.  It has static
    IP on my LAN.

On the pi:

 1. Commented out noload the SIP module.
 2.   Edits to:
    modules.conf  (commented out noload=chan_sip.so   therefore it
    should load)
    sip.conf   (as shown below)
    extensions.conf (as shown below)
 3. Performed asterisk Restart after edits (but not a reboot)

Apologies for needing to be spoon fed on this but it is frustrating to 
have 90% of the needed answers. It would be nice to have all of the 
recipe in one place.

Are there any asterisk CLI commands I can run to debug?

Thanks

Barry w0iy


----------  sip.conf excerpt

[42147] ; my sip phone node number
deny=0.0.0.0/0.0.0.0
username=42147            ;USERNAME        <----------------- is node # 
the correct answer???????????
secret=xxxxxx                   ; SECRET PASSWORD <-----------  is this 
the node pw from allstarlink webpage????????????
dtmfmode=rfc2833
canreinvite=no
context=radio-control
host=dynamic ; unless you set it to a static ip it is dynamic / dhcp
trustrpid=yes
sendrpid=no
type=friend
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp
encryption=no
callgroup=
pickupgroup=
dial=SIP/42147                                        ; NODE NUMBER
mailbox=42147 at device                        ;N NODE NUMBER
permit=0.0.0.0/0.0.0.0
callerid=w0iy ; pick your own cid name CALLSIGN-NODE#


-------------- extensions.conf excerpt

[radio-control] ;ADDED FOR SIP PHONE
exten => 42147,1,Answer                ; <-------- using my node #
exten => 42147,n,Wait(2)
exten => 42147,n,Playback(rpt/node)
exten => 42147,n,Playback(digits/4) ;say node digits one at a time
exten => 42147,n,Playback(digits/2)
exten => 42147,n,Playback(digits/1)
exten => 42147,n,Playback(digits/4)
exten => 42147,n,Playback(digits/7)
exten => 42147,n,Rpt,42147|P|${CALLERID(name)}   ; make connection – the 
P is for phone control mode  -for callerid can be name, num, all, etc
;