Trixbox, Asterisk – External sip clients, no audio

admin — Mon, 06 Apr 2009 03:32:38 +0000

When trying to connect an external sip client to my home Asterisk (Trixbox) after setting up the extension with nat=yes and qualify=yes I could connect and register to the machine, you can check this by using ‘sip show peers‘ from the asterisk CLI

To resolve this issue, I manually edited the sip_nat.conf file, below is my configuration.

nat=yes
externhost=XXXXXXX.dyndns.org
localnet=192.168.1.0/255.255.255.0
externrefresh=120

If you have a static ip, you can use externip=xxx.xxx.x.xx instead of externhost

Also, in my sip.conf file I have the following.

bindport=5060 ; UDP Port to bind to
bindaddr=0.0.0.0 ; (0.0.0.0 binds to all)
rtpstart=10000
rtpend=20000

You can edit these files via the config editor built into trixbox, or from an SSH terminal using Putty with the easy to use nano text editor

You may also need to point the following ports to your asterisk / trixbox from your router

SIP port: 5060
RTP Ports: 10000 – 20000

Nokia Sip Client Settings – Asterisk, trixbox

admin — Mon, 06 Apr 2009 03:30:17 +0000

Nokia, has very limited information and support in regards to the sip client, asterisk and nokia phones ‘took hours’ to figure out!

So, here is a working configuration, tested with Nokia 6100 Navigator and Nokia N95

Got into the Menu -> settings -> Phone Settings – > Connection -> SIP settings

Profile Name: Your desired profile name
Service Profile: IETF
Default Access Point: Your service provider access point, in my case ‘Planet 3‘
Public user name: sip:206@domain.com

Public username is in the format ’sip:ext@ip/domain’ ext is the extension setup on your asterisk server, the IP is the public IP of your asterisk server, if you do not have a static IP, use dyndns.org service.

Use Compression: No
Registration: When needed

You can set this to ‘Always On‘ if you want your sip client ready when you are

Use security: No

Proxy server

Proxy server address: sip:domain/ip
Realm: asterisk
Username: ext ‘Asterisk extention’
Password: pass ‘Asterisk extentsion password’
Allow loose routing: No
Transport type: UDP
Port: 5060

Registrar server

Registrar server address: sip:ip/domain
Realm: ip/domain
Username: ext ‘Asterisk extention’
Password: pass ‘Asterisk extentsion password’
Transport type: UDP
Port: 5060

Have fun Hopefully everything will register for you with those settings, you will need to have the appropriate ports in your router and your sip_nat.conf and sip.conf configuration corrected as outlined in the bottom of my previous post Trixbox – External sip clients, no audio

Linux, windows, asterisk, vmware » sip

Trixbox, Asterisk – External sip clients, no audio

Nokia Sip Client Settings – Asterisk, trixbox