[arm-allstar] Dummies Guide for SIP phone connection to AllStar Node

Fri Apr 10 23:32:25 EDT 2020

Michael,

    You would need to turn on sip in modules.conf, setup an account in
sip.conf, extensions.conf, and an account on the phone. Then you could dial
in from the phone. If the phone only has two accounts then you would have
to give one of them up. Alternatively you could setup a PBX on a Pi running
Asterisk and let it do the connects to all your accounts, then connect that
Asterisk to your Hamvoip Asterisk as I described. This would allow multiple
phones to connect to any of the location not limited by the number of
accounts available on the phone.

*73 Doug, WA3DSP*
*http://wa3dsp.org <http://wa3dsp.org>*

On Fri, Apr 10, 2020 at 5:59 PM "Michael Carey via ARM-allstar" <
arm-allstar at hamvoip.org> wrote:

> Hi Doug,
>
> Thanks for the clarification.
>
> My SIP phone is standalone, I do not have a Asterisk server at home
> (although it is something I did have in the past when I was part of a
> city wide community WiFi network).
>
> I have Cisco SPA504G on my shack desk that has Hamshack Hotline on line
> 1 and a SIP account provided by my ISP on line 2.
>
> I'm not sure on the exact way to set up AllStar to accept a connection
> from my Cisco phone.
>
> 73
>
> Michael.
> VK5ZEA
>
>
> On 10/04/2020 5:38 am, "Doug Crompton via ARM-allstar" wrote:
> > I wrote the prior email late at night and after looking at it again and
> > while it was accurate it is probably different then your circumstance. I
> > have Asterisk running here on a separate server as my PBX. It is using an
> > old version of the hamvoip code. I did everything manually years ago
> adding
> > lots of functionality to my phone system. I can control lighting via DTMF
> > and many other things from a customized extensions.conf file. The IAX
> > protocol is specific to Asterisk and allows communications between
> Asterisk
> > servers and other devices. The radio app is just an extension of that
> over
> > radio. So I have my PBX server which has sip, iax, and analog phones
> > through an adapter connected to it, connected to several of my Allstar
> > servers via iax. The PBX also has a dedicated sip connection to a
> vacation
> > property in another state. So I can pick up any phone at either location
> > and dial a 3 digit *nu*mber and get connected to Allstar.
> >
> > I don't know what your phone system is there but In its simplest form
> this
> > could all be done in one sever if desired. You could connect a sip (or
> IAX)
> > phone on your network, and configure sip.conf and extensions.conf to
> accept
> > a call and route it to app_rpt.
> >
> > *73 Doug, WA3DSP*
> > *http://wa3dsp.org <http://wa3dsp.org>*
> >
> >
> > On Thu, Apr 9, 2020 at 2:09 AM Doug Crompton <wa3dsp at gmail.com> wrote:
> >
> >> Michael,
> >>
> >> This is actually quite easy to do. Here is an example I use for two of
> my
> >> nodes. In my case I use a Grandstream iax phone -
> >>
> >> This stanza is in extensions.conf on my PBX asterisk which happens to
> be a
> >> separate server from Allstar but it could be the same one. You would
> change
> >> the extension, node, password, IAX or SIP, connection port, IP and node
> to
> >> whatever you are using.
> >>
> >> [radio-connect]
> >> exten => 595,1,Wait(1)
> >> exten => 595,n,Dial(IAX2/telephone_dialin:password at 192.168.1.151/27225)
> >> exten => 595,n,Hangup
> >>
> >> exten => 596,1,Wait(1)
> >> exten => 596,n,Dial(IAX2/
> >> telephone_dialin:password at 192.168.1.154:4556/29014)
> >> exten => 596,n,Hangup
> >>
> >> And this stanza is in my Allstar  iax.conf on BOTH of the servers at the
> >> local IP addresses above.
> >>
> >> [telephone_dialin]
> >> type=user
> >> secret=password
> >> disallow=all
> >> allow=ulaw
> >> allow=g726aal2
> >> allow=gsm
> >> codecpriority=host
> >> context=radio-control
> >> transfer=no
> >>
> >> And in extensions.conf - this is just showing one. Another would be on
> the
> >> other node assuming you wanted to call two as shown above.
> >>
> >> [radio-control]
> >> exten=27225,1,Answer
> >> exten=27225,n,Playback(rpt/node)
> >> exten=27225,n,Playback(digits/2)
> >> exten=27225,n,Playback(digits/7)
> >> exten=27225,n,Playback(digits/2)
> >> exten=27225,n,Playback(digits/2)
> >> exten=27225,n,Playback(digits/5)
> >> exten=27225,n,Playback(rpt/connected)
> >> exten=27225,n,Rpt,27225|Pv
> >>
> >> Change password to yours and while this shows an iax connection from the
> >> phone you can certainly use SIP also. In that
> >> case you would put a [telephone-dailin] stanza in sip.conf.  Also make
> >> sure sip is running in modules.conf.
> >>
> >> This uses vox which works very well as long as you don't have exccessive
> >> background noise.
> >>
> >> *73 Doug, WA3DSP*
> >> *http://wa3dsp.org <http://wa3dsp.org>*
> >>
> >> On Thu, Apr 9, 2020 at 1:45 AM "Michael Carey via ARM-allstar" <
> >> arm-allstar at hamvoip.org> wrote:
> >>
> >>> Hi Everybody,
> >>>
> >>> Over the last few days I have resurrected my AllStar node, it's been a
> >>> few years since it has been operational.  I retired my old BBB and have
> >>> a shiny new Raspberry Pi 3B+, It's good to be back on the air again.
> >>>
> >>> One thing that I thought would be cool, would be able to pick up my
> >>> Cisco SIP phone on my shack desk and be able to talk through my AllStar
> >>> node.  I've seen YouTube videos of this being done... but I can't seem
> >>> to find a definitive guide on how to get it working.
> >>>
> >>> Can someone in here point me in the right direction?
> >>>
> >>> 73
> >>>
> >>> Michael
> >>> VK5ZEA
> >>> Node 27871
> >>>
> >>> _______________________________________________
> >>>
> >>> ARM-allstar mailing list
> >>> ARM-allstar at hamvoip.org
> >>> http://lists.hamvoip.org/cgi-bin/mailman/listinfo/arm-allstar
> >>>
> >>> Visit the BBB and RPi2/3/4 web page - http://hamvoip.org
> >>
> > _______________________________________________
> >
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> >
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