Example of how to setup a sip phone for reverse autopatch

Hkwilliamson · February 3, 2010, 2:52am

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@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@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@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:

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@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@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@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@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@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@gmail.comwrote:

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@qrvc.com
http://qrvc.com/mailman/listinfo/app_rpt-users

–
Sent from my mobile device

K1lnx · February 3, 2010, 4:27am

Thanks Keith for sharing that with us, it will certainly help

I’ll also add my own experiences to this for others who may be trying to accomplish what I have as well. I run a seperate Asterisk based PBX for my home phone system which I have 2 Cisco IP phone’s peered off of.

My ultimate goal was to be able to use my existing phone system to get to the radio and out to the network. I use FreePBX on the PBX box, and just set up a SIP trunk to point to the app_rpt box. In the outgoing settings area, you can name the trunk anything you’d like (I named mine app_rpt, original huh?) and in the peer details box I have this:

host=192.168.1.1
username=k1lnx
secret=password
type=peer

Where host is the IP of the app_rpt machine, username is the username you will define in sip.conf stanza, secret is the password, and type=peer defines that we are only sending calls to this machine.

You’ll also need to put registration syntax in the Register String box in the form of username:password@192.168.1.1/context which was defined in the previous section. The “context” portion of the string is what your context name will be on the app_rpt box. For example, mine is k1lnx. You could also do IP based authentication, which would eliminate the need to register.

Next, add an outbound route with whatever dial pattern you want to match (I just used 2335, that’s my node number, to keep it simple) and use the newly created app_rpt SIP trunk. I named my route RADIO. This will now send any calls dialed with “2335” over the app_rpt trunk!

So on the app_rpt box in /etc/asterisk/sip.conf put an entry something similar to this:

[k1lnx]
type=friend
username=k1lnx
password=password
context=radio-control
host=dynamic
disallow=all

allow=ulaw

And in extensions.conf edit (or add if not existing) the [radio-control] stanza:
[radio-control]
;exten => 2335,1,Answer
exten => 2335,1,Playback(rpt/node)
exten => 2335,n,Playback(digits/2)

exten => 2335,n,Playback(digits/3)
exten => 2335,n,Playback(digits/3)
exten => 2335,n,Playback(digits/5)
exten => 2335,n,Rpt,2335|P

Notice I have the first priority commented out? This was due to an issue I was having, the console was spewing error messages about not being able to use the Answer application. No idea why, could be due to the fact that I compiled from scratch or screwed something else up somewhere that I haven’t found yet. YMMV

I also had a hard time getting calls from my PBX to hit this context properly, it took a lot of wrangling and reloading for it to finally work properly. Whenever I would dial “2335” from my desk phone it would get rejected on the basis of not being able to find the extension, I am still largely confused as to why it didn’t simply work however. On the stock extensions.conf that comes with ACID as well as what is in the svn version at the top of the file there are two parameters in the general section:

[general]

static=yes ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.

I had to play with these via commenting/uncommenting them to get things to work, although I don’t believe this was the source of my problem originally. Perhaps Steve or Jim can elaborate on this.

After you issue a dialplan reload you can check for the presence of your newly created extension by issuing “dialplan show extension@context” from the CLI, where extension is the extension number, and context is the context you created (radio-control in my example), and it will return the routing for that extension:

bubastis*CLI> dialplan show 2335@radio-control
[ Context ‘radio-control’ created by ‘pbx_config’ ]
‘2335’ => 1. Playback(rpt/node) [pbx_config]

                2. Playback(digits/2)                         [pbx_config]
                3. Playback(digits/3)                         [pbx_config]
                4. Playback(digits/3)                         [pbx_config]

                5. Playback(digits/5)                         [pbx_config]
                6. Rpt(2335|P )                              [pbx_config]

-= 1 extension (6 priorities) in 1 context. =-

There are other methods to accomplish what I have above, and of course security concerns. This works extremely well and now that I have it working I am going to continue to tweak things and make it as robust as I possibly can. My next challenge is to get inbound and outbound autopatch calls from the app_rpt box to my provider.

Love this project, I’ve been playing with it for just over a year now and I am still learning stuff. So much fun…

tnx and 73
Stephen
K1LNX

···

On Tue, Feb 2, 2010 at 9:52 PM, Keith Williamson hkwilliamson@gmail.com wrote:

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@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@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@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:

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@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@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@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@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@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@gmail.comwrote:

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@qrvc.com
http://qrvc.com/mailman/listinfo/app_rpt-users

–
Sent from my mobile device

App_rpt-users mailing list

App_rpt-users@qrvc.com

http://qrvc.com/mailman/listinfo/app_rpt-users

–

Stephen Brown - ARS K1LNX
Johnson City, TN EM86
http://www.k1lnx.net
google voice: 423-665-9367

william-t2 · August 28, 2023, 3:08pm

Did you ever get this to work properly? Have you a newer version that works with the latest ASL (not the beta version)?

I followed your example and all I get is dead silence for about 3 three minutes then “the number is not answering”.

KI5PHN

Mike · August 28, 2023, 5:28pm

Nothing has changed in the dialplan since that version that would affect it in app_rpt versions.
Normally the largest issue might be context issues depending on the specifics of your plan.

But this thread is over 13 years old.

I suggest you start a new one titled ‘reverse phone patch help’ or similar and state the source of the inbound/inward dialing and give samples of your dialplan for it to allow someone to help you see and understand what you are not.