Hi
I'm looking for some help configuring an asterisk 1.2.8/SIP connection with Freespeech.

Does anyone have an Asterisk set-up I can try other than the one here on the forum...

Some background if you are interested:

I have had a successful Asterisk@home/Trixbox 1.x connection to Freespeech for 13 months. No issues.

Following a hard drive failure I upgraded the device and installed Trixbox 2.2. Actually there was so much to learn, and future recovery to discover that I actually build several versions of Trixbox 2.2, and asterisk 1.2.18, 1.2.19 and 1.4.1 before doing a clean install of Trixbox 2.2 with asterisk 1.2.18.

With all of these installs I have been able to succesfully create trunk SIP connections to a variety of iTSPs ..except Freespeech. But I want to use Freespeech because we generally find that the call quality is better... our qualify results are certaily much better than others.

And I do not know why we are failing. I've followed the setup format here on this and the .co.uk forums, I've used sip debug peer freespeech to try to see what is happening.. I've used FreePBX to config and I've edited the conf files directly.

The asterisk CLI shows clean SIP headers, but the asterisk log shows them deformed.

We could dial in, but after all the attemted debugging we can't even do that.

Any ideas would be appreciated

Regards
David

The same thing is happening to me. Did you manage to get this resolved?

I had inbound and outbound working (outbound had a few issues calling 076 numbers but was working for everything else)
After reinstalling I cant get inbound anymore.