freePBX Asterisk problem

**dswartz** · 05-14-2008 08:56 AM

John, one gotcha you may have missed: the outbound context MUST now be called voipo.com. I had been using VOIPO for outbound and VOIPO_in for inbound and even with all the other changes, still no joy. I noticed Ray had the outbound context as voipo.com and changed to that and was good to go.

**dswartz** · 05-14-2008 09:09 PM

Well, this sucks. Just tried making a call, and it went through, but then I got an email from the script that alerts me when a trunk is down. Checked, and sure enough, no voipo. I have NOT changed anything...

Looking at 'sip show peers', codeblue is showing as unreachable, so maybe something's going on. I'll check again tomorrow.

Update this AM: still down. codeblue.voipo.com is still showing as UNREACHABLE. At least inbound still works

**john** · 05-15-2008 07:44 AM

Shows UNREACHABLE for me also this morning.

VOIPoNorm · 05-15-2008 08:00 AM

John,

Can you please open a ticket at http://support.voipo.com

Please include the sip.conf context you are using to connect.

Also, please include the SIP phone number you're using to connect.

Regards,
Norm

**dswartz** · 05-15-2008 08:31 AM

Norm, I opened a ticket too. The sip info is exactly the working one I PM'ed you...

**fisamo** · 05-15-2008 10:05 AM

I was just able to place a few outbound calls (via DISA) on my Residential line. I used the outbound trunk settings posted earlier in the thread, copied below:

host=codeblue.voipo.com
username=919249xxxx
secret=(secret, of course)
type=friend
insecure=invite
qualify=yes
nat=yes
port=5060
fromuser=919249xxxx
fromdomain=919249xxxx
context=voipo

I have not played with other settings (that I recall) to significantly change them from the default PBX-in-a-Flash setup. (I recently backed up my Trixbox, then over-wrote with PIAF to see if I like PIAF better.) However, I'll see if I can sanitize and retrieve my entire config files so I can post them.

VOIPoNorm · 05-15-2008 10:12 AM

Hi,

I think there may be Asterisk (and FreePBX is just a wrapper around Asterisk) is sending OPTIONS messages. These messages were being rejected by some enhanced security we put in place.

The "qualify" setting turns this "feature" on: http://www.voip-info.org/wiki/view/Asterisk+sip+qualify

The problem here is that we need to authenticate these messages. This is going to take some more investigative work to resolve. In the mean time, I have relaxed this particular security setting.

Thanks for your help in this matter.

Regards,
Norm