Hi, I was running asterisk 1.4, but recently upgraded to 1.6 (for fax support) at the same time as moving from Ubuntu hardy to I have a single TDM400P rev I with two fxo and two fxs modules, these were working perfectly for years on Asterisk 1.4 using Zaptel drivers with Oslec. Now I've moved to 1.6 so I am using Dahdi. Distribution is stock ubuntu package. After several hours (perhaps 24 or so, not nailed it down precisely) incoming calls are not answered and outgoing calls get dial_exec_full. Incoming calls are reported to either A:just ring and ring, or B:get an engaged tone. Strangely when this happens asterisk DOES see the incoming call in situation A, but fails to answer. What tests can I do to resolve this as it is very inconvenient as we are missing a lot of calls? At the moment I have a terminal open all the time with verbose=10 and debug=10, sadly this log is not written to the logfiles so is lost when the terminal exists (perhaps there is a way round this, I don't know) Shutting down asterisk and restarting dahdi removes the problem for another day. Asterisk is version 1.6.2.5-0ubuntu1 Dahdi is version 2.2.1 Any help appreciated. I am at a complete loss what to do, except go back to the old 1.4 server. Cheers, Jason.
On Tuesday 17 Aug 2010, Jason Morgan wrote:> Hi, > > I was running asterisk 1.4, but recently upgraded to 1.6 (for fax support) > at the same > time as moving from Ubuntu hardy to > > I have a single TDM400P rev I with two fxo and two fxs modules, these were > working perfectly for years > on Asterisk 1.4 using Zaptel drivers with Oslec. > > Now I've moved to 1.6 so I am using Dahdi. Distribution is stock ubuntu > package. > > After several hours (perhaps 24 or so, not nailed it down precisely) > incoming > calls are not answered and outgoing calls get dial_exec_full. > > Incoming calls are reported to either A:just ring and ring, or B:get an > engaged tone. > > Strangely when this happens asterisk DOES see the incoming call in > situation A, but fails > to answer. > > What tests can I do to resolve this as it is very inconvenient as we are > missing a lot of calls?Have you got any extensions defined that aren't physically connected to anything? I replaced an ancient Asterisk version with a bang-up-to-date 1.6 version I built myself, and was getting similar symptoms to what you describe. It seemed not to be freeing up channels it was trying to associate with non-existent devices. I made sure that every entry in sip.conf had a corresponding phone plugged in somewhere, then went through the dialplan and removed all references to anything that wasn't mentioned in sip.conf. (And there were a few.) It seems to have stayed up since then ..... -- AJS