[arm-allstar] SIP phone

[Robert Prybyzerski]

Barry, I have a separate page where I tried to clarify some of the information.  

https://w2ymm.home.blog/sip_phone_extension/

I added additional notes.  
Bob W2YMM 

Sent from my iPhone
Bob P W2YMM 
(631)960-1051

> On Jan 27, 2021, at 4:33 PM, iabarry Buel via ARM-allstar <arm-allstar at hamvoip.org> wrote:
> 
> 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
> ;
> 
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