Connect to SIP VOIP using Asterisk Mobile Phone

This article is using the open source PBX Asterisk to setup a SIP account properly on the mobile phone. Use Nokia Series N81 to write this mobile configuration. Some of other mobile series will have different settings, therefore the configuration for different SIP servers may be similar.

Things you need to do is enable voip internet phone at your mobile phone
1. Your mobile phone must have WIFI connection that can connect to wireless access point .
2. Wireless account point connected to internet with open port UDP 5060
3. Asterisk / SIP Server username password

SIP Server /Asterisk Settings

We can use public VOIP or we can create our own SIP Server. In the next article, I will write more technical things about how to configure your own SIP Server / Asterisk. This article is written when the SIP Server is already installed

Create a new SIP account on the Asterisk / SIP Server

Add following setting to sip.conf (the username “agushalim” is just used for example!):
username=agushalim
type=friend
secret=SIP
qualify=no
port=5060
notransfer=yes
host=dynamic
context=from-internal
disallow=all
allow=alaw

SIP realm

The realm for digest authentication is set defaults to “asterisk”. Otherwise please verify your active realm in sip.conf.

Mobile Phone Settings

Navigate to Tools - Settings - Connection - SIP settings

Open the Options menu and select Add new - Use default profile :
Profile name : Asterisk
Service profile : IETF
Default access point : your WIFI / wireless access point
Public user name : sip:agushalim@SIP Server(SIP : username@domain or IP address)
Use compression : No
Registration : Always on
Use security : No

Proxy server
Proxy server address : sip: SIP Server (SIP : IP address of your asterisk / sip server)
Realm : asterisk (or realm of your asterisk)
Username : agushalim (Your SIP username)
Password : SIP (Your SIP password)
Allow loose routing : Yes
Transport type : UDP
Port : 5060

Registrar server
Registrar server address : sip: SIP Server (SIP : IP address of your asterisk / sip server)
Realm : -
Username : agushalim (Your SIP username)
Password : SIP (Your SIP password)
Transport type: UDP
Port: 5060

Now, navigate to Tools - Settings - Connection - SIP Settings / Internet telephone settings and create a new profile in the Options menu with the following settings:
Name: Default
SIP profiles: previously defined profile
Default call type

To switch between normal GSM calls or VOIP calls, navigate to Tools - Settings - Call - Default call type.

Select Cellular to make normal calls to the phone number or Internet to use VOIP to call the number or address.