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Asterisk phone sjphone

Created by: oej,Last modification on Thu 10 of May, 2007 [20:21 UTC] by Darthapache

How to configure Asterisk for the SJphone


Asterisk configuration


In this sample configuration, 192.168.0.1 represents the Asterisk server and 192.168.0.2 is the client running SJPhone.

Configure Asterisk to accept registration and inbound calls in sip.conf like this:

 [mysjphone]
 context=from-sip
 type=friend
 host=dynamic
 username=mysjphone
 secret=blablabla

Next, configure Asterisk to dial the SJphone with an entry in extensions.conf

 exten => 100,1,dial(SIP/mysjphone)
 exten => mysjphone,1,goto(100,1) ; To be able to dial with text, "mysjphone"

SJPhone configuration:


1/ click on the Options button
2/ go to the Profiles tab.
3/ click on 'New'
4/ create a new profile called 'asterisk' with profile type 'Calls through SIP proxy'
5/ use this profile for your asterisk connection with the following settings:

 Register with proxy - checked.
 Proxy domain: 192.168.0.1

Leave the rest of the settings at default. When you hit the OK button, it will ask for

  Account: mysjphone
  Password: <as above>

You can change the account and password by reinitializing the profile.

When it's working, SJPhone's main display shows:

 Status: no active calls
 Default protocol: SIP
 SIP Proxy: registered with 192.168.0.1
 Host address: 192.168.0.2

and Asterisk's console displays:
 Registered SIP 'mysjphone' at 192.168.0.2


Mark Johnston, 2003-11-27
Rob Scott, 2005-01-21

Please note

This configuration does not take NAT into consideration. To do that, read the Asterisk FAQ on SIP and NAT.
In addition, You can view SJPhone/Asterisk setup instructions for newer versions of SJPhone Here or you can find a complete step by step configuration manual on asteriskguru

See Also


Asterisk | Asterisk Configuration | Channel Configuration | Configuration for Specific Phones

Comments

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222

333Question

by AsteriX, Saturday 02 of April, 2005 [09:49:38 UTC]
(:question:)

Can I use SJphone like a H.323 phone in order to dial and receive call through Asterisk?Can I consider Asterisk just like a sort of H323 Gateway?

Thanks 4 all!
AX
222

333GSM and inband dtmf

by , Thursday 03 of February, 2005 [19:23:55 UTC]
I was getting a message about GSM does not support inband dtmf.

I had to disable GSM codec (the default for SJPhone), then I could call the asterisk server from the SJPhone.

I also made Ulaw the top of the Codec list.
222

333External SJPhone Extensions?

by docelm0, Sunday 03 of October, 2004 [16:38:36 UTC]
(:cry:) Here is my problem. I have internal 192.168.1.x network and my * is in the DMZ acting on the outside world. BUT.. I have friends that are connect to my * but when I call them its not sending to there router's IP addy. cause of them being natted.. I get an error when trying to connect. I did the debug and it shows something like this:

Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK04367ddc;rport
From: "2000" ;tag=as5e4b16da
To:
Contact:


When I call the 3000 its sitting on a network outside mine but it cant connect to it. Could someone help?

222

333Subscription does not exist

by , Thursday 19 of August, 2004 [12:28:53 UTC]
I also got this error message and solved the problem unchecking the "Enable STUN usage" option in Options -> Profiles -> in_use_profile -> edit -> compatibility.
OPTIONS SIP messages were being sent periodically from my SIP phone to the Asterisk server and responded with "404 Not found" error messages. This messages are not sent anymore after disabling STUN.

Hope it helps!
 Alberto