I was wrong AT&T is NOT blocking my SIP

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


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.





Is AT&T blocking SIP

Last night I installed a SIP client (CSipSimple) on my phone, and it worked perfectly on my wireless network.  So I said to myself, what about 3G.  I gave it a few test and while I get the inbound and can make outbound calls.  There is lack of voice data being sent.  This was a major let down.  So I went to my SIP server and made changes in hopes to get this working, without any change in behavior.


This morning I attempted this once more, with the same not so desirable results.  So then I said “let’s try this via VPN.  I then setup a PPTP VPN connection from my phone only using the 3G (H+) network, I then setup the a new SIP Connection pointing to my internal IP of my SIP server and like magic I was registered and able to accept inbound calls and make outbound calls.


So now I convinced that AT&T is blocking part of my SIP Call.   Well wait, perhaps I should try something else.   And that is what I did.  I connected to a remote network, and setup a SIP client on a computer to connect to my SIP server via the internet.   I was able to register and accept / and place calls with full voice also.   I now conclude that AT&T is blocking parts of SIP.


After some Google searching, it seems that AT&T allows SIP on data network, but its only seems to be for those with iPhone plans.  Well isn’t that nice.


So there it is, I will use my SIP connection to make calls via VPN, it introduces an extra step but it works.  Most of all its secure, so I guess it’s for the best.