Well a bit of egg on ones face isn’t a bad thing.  Failure to admit to it is…

So I was thinking about this issue during my date around the house and playing with the kids and when I had some time to review what was going on I got to thinking.  Perhaps this is an issue with just voice because my SIP phone does register, and I am able to answer the calls.  I simply am not hearing any audio and the the call drops off.

So I found an a few posts about what changes others made. And eventually one of the procedures worked well for me.

Doing the following has resolved my issues. And YES!  SIP is working over the AT&T 3G Network.

Sorry AT&T I was quick to blame ya…


Made sure my system knew its name

nano /etc/hosts

look for this line: localhost

DO NOT REMOVE OR CHANGE THAT LINE. On a NEW, place this line: jermsmit.com

But substitute YOUR address, of course.

Add some information to your /etc/asterisk/sip_nat.conf file
If this file doesn’t exist you’ll have to create it, but make sure that the ownership and permissions match those of sip.conf and other files in that directory. You can use the command to create the file if it doesn’t exist.  In my case it did.

touch /etc/asterisk/sip_nat.conf

nano /etc/asterisk/sip_nat.conf

Now edit the file and insert AT LEAST these two lines:



The above localnet line assumes that your local network uses 192.168.0.x addresses, but if it uses something else, make the appropriate substitution.

I use only these four lines, as follows:






Reload SIP

After you have added whichever lines you need in sip_nat.conf, go to the Command Line Interface and type

# asterisk -r

*CLI> sip reload

And hit enter. Alternately you could restart Asterisk, but that will interrupt any calls that are in progress.


Be sure to have opened the SIP and RTP ports to your Asterisk server via your firewall.

You must make sure that you open the correct UDP ports in your router’s firewall and pointed at your Asterisk server. For SIP protocol, open UDP (NOT TCP) port 5060 (SIP) AND ports 10001-20000 (RTP, which must also be defined in /etc/asterisk/rtp.conf, see below).

Check your /etc/asterisk/rtp.conf file

It should contain these two lines like this:



I then saved my config and restarted my services with the command

# amportal restart

After that I placed a test inbound call to my SIP registered device sitting on 3G and I could hear my audio…  I then placed the call on hold, 30 min’s so far, and the default FreePBX on hold music isn’t that bad… However I will be looking for a way to change it

Thanks again for reading the above.