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Posted on January 16, 2016 by Paul

Asterisk to UAX Junctions - Update

Over the Christmas period, I spent quite a lot of time giving out Asterisk config a good going over, trying to identify the root cause of it not always releasing junctions at the end of a SIP--UAX call. In the process, I found and fixed several issues.
" We now pause slightly before dialling, to play a little nicer with slow line finders.
" Whenever we use Dial to send a call out to the UAX, we explicitly jump to the (h)angup extension, so that we can trap the end of the call
" We're now a bit more thorough about logging what we're doing as a call progresses. This isn't a "fix" as such, but it does help debugging intermittent errors
" Our DAHDI config was using Loop Start (ls) rather than the preferred KewlStart (ks) signalling. ks is supposedly a bit better at disconnect supervision
" We upgraded from Debian Wheezy to Debian Jessie. This gives us an upgrade to a more recent version of asterisk
" In sip.conf we now set an rtp timeout so that we end the call if we don't receive any RTP (audio) traffic for 60 seconds. This should help to end calls which haven't been torn down properly
" We turned off "SIP ALG" on the BT Business Hub router as this seems to be causing SIP packets to occasionally go astray
" We were rotating the logs a little aggressively and were only keeping a couple of weeks. This has been expanded so we now keep several months - which should make retrospectively investigating issues easier.

With all that done, we decided last week to reinstate one of the junctions from Asterisk to UAX. By only enabling a single junction, we can easily detect when it gets stuck (as any subsequent calls will get NU from the asterisk). It's been back up for a week, and so far it's working well. I'm still keeping an eye on it, and will continue to do so for a while before we re-enable the second junction.

Posted on March 1, 2016 by Paul

And we're offline again (mostly)

The Asterisk is currently offline (mostly). Some classes of call can be completed in some circumstances (eg calling the speaking clock on 400 works). Some calls complete but with no audio (I believe calls between our home phones do this). All calls out to the strowger network fail. Calls in from the sipgate dial in number get as far as the voice prompt on the asterisk box itself, but can't dial out into the strowger and seem to have the same audio problems (ie if I call my voip number I get no audio).
My remote access has stopped working, so I can't troubleshoot it in detail from here, detailed troubleshooting will have to wait until I'm able to get on-site. Unfortunately that's not until Saturday 9th April. Updates will follow on here as and when I know anything.

Posted on March 20, 2016 by Paul

Asterisk back online - 2016-03-19
I managed to get to site yesterday with Peter, and the asterisk is now back on line and working again.
The fault turned out to be with the BT router at Norchard (again!) This time it was a new failure mode I haven't seen before. It had the port forwarding rules configured, but for some reason had decided it didn't want to actually use them any more. I took the port forwarding off and put it back again and everything sprung back to life.

Posted on March 22, 2016 by Paul

It seems that the status page stopped working back in December. I've fixed it, and the graphs are working again now.

Posted on July 14, 2016 by Paul

Downtime - fixed

It looks like at about 6pm on 12th July 2016, our external IP address changed at Norchard (so much for "BT Business - Static IP") but for some reason our dynamic DNS didn't catch up. So the asterisk has been effectively offline since then. Service was restored at around 4pm on 14th July when I noticed it was down! My phones are still offline (including the C*Net and SipGate numbers) due to some technical glitches at home, but that's unrelated!

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