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
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Tags: asterisk, audio, sip, trixbox
I am running into the same issue. remote SIP softphone(X-lite) gets registered and i can call my local softphone. Once the call gets established, I don’t see any audio passing through. Its been few days already. I see whole bunch of documents on the internet but none is comprehensive enough to resolve no-audio issue.
Hi,
So, were there any steps above that you were having issues with that you didn’t quite know what todo?
have you tried with using your IP instead of a dynamic DNS?
Have you made sure that the SIP and RTP ports are set to forward to the trixbox?
Kind regards,
Shayne