We are running Asterisk 1.8.3 on Fedora 14 and on a few occasions, asterisk locks up and the process needs to be killed.
Just before the crash I see this in the log
[Jun 10 16:06:23] WARNING sig_pri.c: Received NOTIFY on unconfigured channel 255/255 span 1
[Jun 10 16:06:31] WARNING sig_pri.c: Received NOTIFY on unconfigured channel 255/255 span 1
Span 1: WCT1/0 “Digium Wildcard TE110P T1/E1 Card 0” (MASTER) B8ZS/ESF
Let me know if more info is required. Anyone else see this before? All help is appreciated
Confirm that you can repeat this on the latest version then:
Compile with thread debugging on. When it freezes, start a new console session and run core show locks. Attach gdb and do:
thread apply all bt
thread apply all bt full
Attach the results to a bug report on issues.asterisk.org.
Thanks for the quick reply. To get to the latest version, Is there an update procedure or do I need to start all over with a new install? If there is an update procedure, is there any documentation that you can point me to?
Assuming installing form source, as long as you don’t do make samples, the configuration will not change if you do a complete build and install.
There is no need to remove the older version first? Just build the new source over the older without creating samples?
Your help is appreciated…
It is safer to clear out /usr/lib/asterisk/modules first, but you will be warned about any obsolete modules.
One more quick question on the upgrade to 22.214.171.124. Is it necessary to update libpri and dahdi-linux-complete as well? Is there a way to revert back to 126.96.36.199 if needed? Am I being to paranoid? We are also seeing another error message as well,
“WARNING app_dial.c: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)” and we are hoping that clears up as well with the upgrade… Thanks again.
In my experience, that error is always the result of a configuration error.
Such as extension.conf or asterisk.conf? We were originally using asterisk 1.4 and started fresh install using 188.8.131.52 then just copied the 1.4 config files over… Thanks again
Normally extensions.conf or sip.conf. It might be the result of the change in the parameter delimiter and/or the handling of double quotes. It generally means that you are trying to dial something that neither looks like an IP address or domain name, nor matches an entry name in sip.conf.
Thanks. So do you think its necessary to update libpri and dahdi when I update asterisk to 184.108.40.206?
Thanks again for all your help David55. I completed the upgrade to 220.127.116.11 and everything went well. I guess it’s a wait and see if we have any more lockups. One issue we are still seeing is when we dial 1-866-222-7056 which is a AT&T conference bridge we here no dialing and see in CLI the following error
– Executing [18662227056@default:1] Dial(“SIP/132-0000001e”, “DAHDI/g2/8662227056,45”) in new stack
– Requested transfer capability: 0x00 - SPEECH
– Called g2/8662227056
– DAHDI/i1/8662227056-1a is proceeding passing it to SIP/132-0000001e
– PROGRESS with cause code 127 received
Any ideas? It was suggested to disable early media which I’m not sure how to do. Should I start a new thread for this?