SIP via TCP?

Is it possible to use TCP instead of UDP for SIP traffic with extensions?

Our PiaF/Asterisk/FreePBX installation is here for no other purpose than to act as a source of VoIP traffic for testing stuff that we write. We need a source of TCP SIP.

Asterisk has supported TCP SIP since 1.6.

“Asterisk has supported TCP SIP since 1.6”…


I installed the version of PBX in a Flash that was current at the time, which was late last year. It has Asterisk version 1.4, which therefore is what I’m running.

Some things aren’t clear to me:

  1. How do I upgrade Asterisk? Is it possible to do it from within FreePBX? If so, I haven’t managed to find out how/where.

  2. Is FreePBX compatible with Asterisk 1.6? If so (and I realise that the answer is probably yes), what minimum version of FreePBX? Again, I haven’t managed to find where the information is.

FreePBX has supported Asterisk 1.6 since version 2.5

Searching Asterisk 1.6 revealed that answer.

As far as upgrading Asterisk in PBXiaF I don’t know the process. You can always download the current version of Asterisk at

WIth PBX in a Flash, there is a very active forum

Basically, you need to type several commands at the command prompt in Linux:

update-scripts - will get you the latest PIAF scripting for the system.
update-fixes - applies current fixes to PIAF and related applications
update-source - will get you the latest and greatest Asterisk 1.4 and related stuff.

Taking your system from 1.4 to 1.6 Asterisk is a little more complex. Research the PIAF forums for information.

Regardless of what else you do, be sure you’ve upgraded your FreePBX to the latest version (2.7) and update all the modules. Back up all your configuration files with the FreePBX backup.

OK, I’ll see what the PIAF forum has to say.

Just one clue - does upgrading mean building Asterisk from source?

OK, thanks for your help. If only I had a spare server…

The good news is that the Asterisk system I have is only just above minimal; it consists of 5 extensions, no trunks, no external connections. There can’t be a lot of extras!

I’m torn between your from-scratch method, and the build-on-the-target-machine method I read about in the PiaF forum, which keeps the existing system safe by the simple expedient of renaming the (few) directories that contain all Asterisk’s files - code and configuration. One of them will work :slight_smile:

There is no clean way in PIAF to upgrade from 1.4 to 1.6 at this point. When you install a PIAF ISO, it is either for 1.4 or 1.6. You can manually upgrade Asterisk to 1.6 but all your PIAF scripts and extras will still be for Asterisk 1.4.

When you are going to 1.6, the best thing is to download the PIAF ISO for 1.6 and burn it to a CD. Back up your system and be sure to run a FreePBX backup and put it someplace where you can restore it later.

Then I’d load the new PIAF disk and start from scratch. You can then restore the FreePBX backup and have most of your work done. You just need to modify the /etc/amportal.conf file to reflect ZAP2DAHDICOMPAT=true so your old ZAP stuff can be converted to DAHDI.

When I’m doing something this radical, I like to keep the original system untouched and build from scratch on a backup server for testing. None of this is for the faint of heart.