[App_rpt-users] Example of how to setup a sip phone for reverse autopatch

Keith Williamson hkwilliamson at gmail.com
Wed Feb 3 02:52:12 UTC 2010 

Thought I'd share this thread with the list since it might be helpful to
someone trying to setup a local sip phone for reverse autopatch.

Bob and I went off the list when we first got started on this because I was
afraid the thread could get quite lengthy if we ran into the (almost)
inevitable problems. However, everything worked the first time through a
combination of luck on my part explaining things (blue moon) and Bob's
excellent ability to read, interpret, and follow my drivel.

Bob, if there were things I explained incorrectly below that you figured
your way around, please chime in so that the next unwary ham who tries this
will be forewarned. Also note that some of this is specific to Bob's SIP
phone but in looking through the manual for the phone, I found it to be
pretty typical of other SIP phones I've worked with. Finally, apologies for
the reverse ordering of the thread versus the way we normally (try) to order
thread (newest reply at bottom).

73's

Keith

KF7DRV
Allstar Node 2541

---------- Forwarded message ----------
From: Bob Brown - WØNQX <bbrown at byrg.net>
Date: Sun, Jan 31, 2010 at 12:03 PM
Subject: SIP IP pHONE config - was: Connect notifications via AGI or System
scripts
To: Keith Williamson <hkwilliamson at gmail.com>


hi keith

got the first part done
~~~~~~~~~~~~~~~~~~~~~~
sip show peers

Name/username              Host            Dyn Nat ACL Port
Status
9009/9009                  10.0.0.40        D          5060
Unmonitored
1 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 0
offline]
 ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

whats next?

--
Thanks in Advance

  Bob Brown, WØNQX

  Kansas City Metro Area

  http://sm0kenet.net

  http://byrg.net

  http://kcdstar.byrg.net

Quis custodiet ipsos custodes?



On Sun, Jan 31, 2010 at 12:09, Keith Williamson <hkwilliamson at gmail.com>wrote:

> OK great. Next step is to get the phone registered with the Allstar node.
> I'm going to assume the phone is connected to the internal network the node
> is on and for the purposes of this explanation I'll say the internal IP of
> your Allstar node is 192.168.0.1 (and that that IP is either static or gets
> a static lease via DHCP).  Obviously, substitute your node's actual internal
> IP.
>
> In the web interface for the phone, enter the Allstar node internal IP in
> ALL of the following fields of Line 1 settings for Profile 1:
>
> SIP Proxy Server
> Outbound Proxy Server
> Registrar Server
> Outbound Registrar Server
>
> Leave the SIP Proxy Server Port and Registrar Proxy Server Port set at the
> default of 5060.
>
> Also in the same Line 1 Profile 1 menu, you need to set the phone's
> extension info. You can make this whatever you want but for the purposes of
> this explanation we'll call it Bobs208. Set Bobs208 in the following fields:
>
> Phone Number
> User Name
> Authorized ID
>
> Finally, you need to set a password for authentication. We'll use
> "secret123". Set the following field to secret123:
>
> Authorized Password
>
> This should be enough on the phone side of things so now down at the bottom
> of the menu, do Save Settings followed by Logout.
>
> Now we move over to the Allstar node. Open /etc/asterisk/sip.conf with your
> favorite editor (I'm a vi guy myself) and add a stanza for your phone (below
> the end of the [general] stanza) that would be something like this:
>
> [Bobs208]
> type=friend
> secret=secret123
> context=radio-control
> host=dynamic
> disallow=all
> allow=ulaw
> transfer=no
>
> Now save the changes to sip.conf and exit the editor. Next go into the CLI
> (e.g. asterisk -vr) and type "sip reload" to force asterisk to reread the
> sip.conf file. At this point, asterisk should be ready register your phone.
>
> Probably the most dependable way to get the phone to activate the changes
> you made to the phone's SIP parameters and to force the phone to attempt to
> register with Allstar/Asterisk is to just power-cycle the phone.
>
> I haven't read enough about this phone to know how the phone indicates it's
> registration state after it finishes booting but you can watch the Allstar
> CLI and you should be able to see the registration occuring. You can also
> type the following CLI command to check if you phone is registered:
>
> CLI> sip show peers
>
> Let me know how all this goes and then we can proceed to the final step
> once the phone is registered.
>
> 73's
>
> Keith
>
> KF7DRV
>
>
>
>
>
> On Sat, Jan 30, 2010 at 11:57 PM, Bob Brown - WØNQX <bbrown at byrg.net>wrote:
>
>> ya mine is a virgin fone and the default is 1234
>>
>> i got this from a ip phone vendor/wholsaler it was a freebie he says it
>> was sip compatable
>>
>> i have managed to log into it from the web interface and from the admin
>> menu in the front panle.
>>
>>
>>
>> --
>> Thanks in Advance
>>
>>   Bob Brown, WØNQX
>>
>>   Kansas City Metro Area
>>
>>   http://sm0kenet.net
>>
>>   http://byrg.net
>>
>>   http://kcdstar.byrg.net
>>
>> Quis custodiet ipsos custodes?
>>
>>
>>
>>   On Sat, Jan 30, 2010 at 22:45, Keith Williamson <hkwilliamson at gmail.com
>> > wrote:
>>
>>> Hi Bob,
>>>
>>> I found the manual for your T207/T208 phone..hopefully you have that too.
>>> Getting the phone registered with Asterisk requires that the phone be an "S"
>>> model (T207S or T208S) which means it has SIP firmware. The T207M and T208M
>>> models run MGCP which isn't compatible with Asterisk. The other requirement
>>> is that you need to know the admin password. If it's never been changed it
>>> should be the default "1234". You can configure the phone settings either
>>> directly on the phone or via your web browser (although you need to at least
>>> configure the TCP/IP parameters using the phone menus before you can switch
>>> to using the more convenient web interface).
>>>
>>> Perhaps you have already done this part and have the phone connected via
>>> Ethernet to the network the node is on?
>>>
>>> Let me know and we can go from there.
>>>
>>> 73s,
>>>
>>> Keith
>>>
>>>
>>> On Sat, Jan 30, 2010 at 10:02 AM, Bob Brown - WØNQX <bbrown at byrg.net>wrote:
>>>
>>>> my ipphone: TSIPP2008BG-R1
>>>>
>>>> sure any help would be great!
>>>>
>>>>
>>>>
>>>> --
>>>> Thanks in Advance
>>>>
>>>>   Bob Brown, WØNQX
>>>>
>>>>   Kansas City Metro Area
>>>>
>>>>   http://sm0kenet.net
>>>>
>>>>   http://byrg.net
>>>>
>>>>   http://kcdstar.byrg.net
>>>>
>>>> Quis custodiet ipsos custodes?
>>>>
>>>>
>>>>
>>>>   On Fri, Jan 29, 2010 at 23:01, Keith Williamson <
>>>> hkwilliamson at gmail.com> wrote:
>>>>
>>>>> Hi Bob,
>>>>>
>>>>> No problem. What make/model of ipphone is it? The first thing required
>>>>> is a stanza in sip.conf defining the extension, user, password, etc
>>>>> for that particular phone. The required elements vary from phone to
>>>>> phone so let me know what kind you have. After creating the entry in
>>>>> sip.conf, you just do a "sip reload" at the CLI and check to see if
>>>>> the phone gets registered OK. Once you have that, you just need to add
>>>>> a stanza to extensions.conf to enable the reverse autopatch.
>>>>>
>>>>> There's a pretty complete description of what's required in the ACID
>>>>> Admin Manual.
>>>>>
>>>>> Cheers,
>>>>>
>>>>> Keith
>>>>>
>>>>>
>>>>> On 1/29/10, Bob Brown - WØNQX <bbrown at byrg.net> wrote:
>>>>> > hi keith
>>>>> >
>>>>> > i would be interrested in your config files to set up the ipphone
>>>>> > on your node
>>>>> >
>>>>> > i have a 4 line ip desk phone i would like to set up to do what you
>>>>> describe
>>>>> >
>>>>> > i am not very fluent in asterisk set up for ip fones.
>>>>> >
>>>>> > would you be willing to share?
>>>>> >
>>>>> >
>>>>> > --
>>>>> > Thanks in Advance
>>>>> >
>>>>> >   Bob Brown, WØNQX
>>>>> >
>>>>> >   Kansas City Metro Area
>>>>> >
>>>>> >   http://sm0kenet.net
>>>>> >
>>>>> >   http://byrg.net
>>>>> >
>>>>> >   http://kcdstar.byrg.net
>>>>> >
>>>>> > Quis custodiet ipsos custodes?
>>>>> >
>>>>> >
>>>>> >
>>>>> > On Fri, Jan 29, 2010 at 22:04, Keith Williamson
>>>>> > <hkwilliamson at gmail.com>wrote:
>>>>> >
>>>>> >> Hi,
>>>>> >>
>>>>> >> Recently, I got reverse autopatch configured on my simplex node
>>>>> (2541) and
>>>>> >> everything works great. In the shack, I have a Polycom IP501 speaker
>>>>> phone
>>>>> >> that I use to monitor or connect into QSO's on the node. Out of the
>>>>> shack,
>>>>> >> I
>>>>> >> can call into the node using my Blackberry and do the same. So the
>>>>> next
>>>>> >> challenge is to get quick notifications of users connecting into the
>>>>> node
>>>>> >> either via radio or Internet (Echolink and Allstar..no IRLP yet). I
>>>>> >> created
>>>>> >> a pair of Twitter accounts; one for the node and another for me
>>>>> >> personally.
>>>>> >> I added my node's Twitter account to the ones I follow with my
>>>>> personal
>>>>> >> account and created an AGI scripts that formats a curl command which
>>>>> posts
>>>>> >> some of the context variables passed in when Asterisk invokes the
>>>>> script
>>>>> >> (context name, extension, callerid, etc). I then added the AGI call
>>>>> into
>>>>> >> various extension stanzas in extensions.conf to test. If, for
>>>>> instance, I
>>>>> >> do
>>>>> >> a reverse autopatch and connect to the reverse autopatch
>>>>> "extension", I
>>>>> >> almost immediately get an SMS tweet on the Blackberry.  Great.
>>>>> However, it
>>>>> >> seems that connections via the radio are not processed in any way in
>>>>> >> extensions.conf (thought [default] stanza would apply but apparently
>>>>> only
>>>>> >> applies to autopatch), and inbound connections from either Allstar
>>>>> or
>>>>> >> Echolink, while processed through [radio-secure], hang on the AGI
>>>>> call and
>>>>> >> don't proceed to the following call to rpt. I'm assuming this is
>>>>> because
>>>>> >> AGI
>>>>> >> calls generally follow an "Answer" and are only valid in the
>>>>> connected
>>>>> >> state. I tried "deadAGI" since it doesn't seems to be dependent on
>>>>> being
>>>>> >> in
>>>>> >> the answered state but it hung the same way. It's possible I've got
>>>>> a
>>>>> >> problem in the AGI script which is causing the hang but it certainly
>>>>> >> doesn't
>>>>> >> occur when the AGI call follows "Answer".
>>>>> >>
>>>>> >> So I'm looking for help to understand other possible ways to invoke
>>>>> a
>>>>> >> script that calls curl to post the tweet when a radio user makes an
>>>>> >> outbound
>>>>> >> connection to Allstar or Echolink and when an Echolink or Allstar
>>>>> user
>>>>> >> connects in to the radio.
>>>>> >>
>>>>> >> Ideas?
>>>>> >>
>>>>> >> Thanks and 73's
>>>>> >>
>>>>> >> Keith
>>>>> >>
>>>>> >> KF7DRV
>>>>> >>
>>>>> >> _______________________________________________
>>>>> >> App_rpt-users mailing list
>>>>> >> App_rpt-users at qrvc.com
>>>>> >> http://qrvc.com/mailman/listinfo/app_rpt-users
>>>>> >>
>>>>> >>
>>>>> >
>>>>>
>>>>> --
>>>>> Sent from my mobile device
>>>>>
>>>>
>>>>
>>>
>>
>
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