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Version: Baidi441

N906iL - Asterisk (Ubuntu) - Sipphone.com

call cheap from your home-pbx to anywhere.

This is a setup note for connecting Docomo N906iL to a IP-callout-PSTN service. To build a wireless IP-phone.


What you need on this note.

  1. IP-Phone (softphone or hardphone, whatever. I used N906iL here, as it is most powerful handset in Japan)
  2. Wifi Enviromnent with Internet connection.
  3. Ubuntu PC, as it is a greatest enviroment for building a server.
  4. Gizmo account, and callout credit if you want to reach PSTN worldwide.

Visual map of what you do here.

IP-phone = N906iL in this case. It connects to Asterisk using SIP.
Asterisk behaves as an SIP-client, against "proxy01.sipphone.com".
proxy01.sipphone.com connects you to PSTN worldwide.

Step1) Building a simple IP-PBX with Asterisk.

  • 1.1) On a normal Ubuntu PC, just type "sudo apt-get install asterisk"
  • 1.x) Reflecting changes, and see if it is working.

Step2) Connect your N906iL to Asterisk.

  • 2.1) Set up your N906iL
Hope you are done with WLAN settings. Below are just for "SIP settings"
** replace 10.0.0.3 with your Asterisk's IP address.
SIP address: 906@10.0.0.3
SIP Server: 10.0.0.3
Presence Server: 10.0.0.3, port:5060  ** maybe not necessary...
SIP Sequence: Auto-Basic
Digest Authorization: ID=906, Pass=906
  • 2.2) Edit sip.conf
    First, "sudo gedit /etc/asterisk/sip.conf" and add below to the bottom of the file.
[906]
type=friend
username=906
secret=906
host=dynamic
disallow=all
allow=ulaw
qualify=yes
  • Then, add one line below to [general] section.
    register => 1747xxxxxxx:yourpassword@proxy01.sipphone.com
  • 2.3) Go to step 1.x, if your asterisk log says like "906 is registered", you are done with this step.
  • 2.x) By default, Ubuntu-asterisk setting have a demo call at "600" number. Just type "600" and call thru WLAN on your N906iL. You will hear a echo test navigation.

Step3) Connecting your Asterisk to Gizmo cloud.

  • 3.1) Edit sip.conf
[proxy01.sipphone.com]
type=peer
disallow=all
allow=ulaw
allow=ilbc
dtmfmode=rfc2833
host=proxy01.sipphone.com
fromdomain=proxy01.sipphone.com
qualify=yes
fromuser=1747xxxxxxx
username=1747xxxxxxx
secret=PASSWORD
canreinvite=no
nat=yes
context=outbound
  • 3.2) Edit extensions.conf
    "sudo gedit /etc/astrerisk/extensions.conf"
    and find [default] section. Add a line below.
    include => outbound
    Delete "include => demo" line. This will make all of your call to be made thru Sipphone.com.

    And also, add these to the bottom of the file.
[outbound]
exten => _X.,1,Set(CALLERID="YOURNAME" <1747xxxxxxxx>)
exten => _X.,2,Dial(SIP/${EXTEN}@proxy01.sipphone.com,20,r)
exten => _X.,3,Hangup
  • 3.3) Go to step 1.x, if your asterisk log says like "sipphone.com is registered", you might be happy.
  • 3.4) Call "17474743246" for echo-test. This will assure you that you are reaching Gizmo/Sipphone.com cloud.
  • 3.5) Call any number on the list.... To Japan PSTN, try "0081-x-xxxx-xxxx".
  • 3.x) In case of error, see "sudo cat /var/log/asterisk/messages" or go to chapel to pray.

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yngd
yngd
Network Engineer at V-zon
Tokyo
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