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Summary:ASTERISK-08915: asterisk loosing networking
Reporter:frawd (frawd)Labels:
Date Opened:2007-03-01 12:16:39.000-0600Date Closed:2007-04-04 10:53:36
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) fulllog.txt
( 1) sip.conf
Description:I'm experiencing this issue every few days, at least all LAN and WAN SIP phones (for I'm only using SIP) become unreachable in Asterisk, and DNS queries don't work.
The server still has complete network conectivity and can reach all IPs and do DNS queries as usual.

A restart or "stop-start" doesn't solve anything (it restarts with the log uploaded), and a complete machine reboot it required.

****** ADDITIONAL INFORMATION ******

OS: debian etch
kernel: custom based on debian 2.6.18-XEN kernel
Comments:By: Joshua C. Colp (jcolp) 2007-03-05 14:31:58.000-0600

Asterisk uses the standard posix APIs for DNS lookup. I doubt it's specifically Asterisk's fault and doubt there's anything we can do... but just out of curiosity - what's the contents of resolv.conf?
By: frawd (frawd) 2007-03-07 16:14:55.000-0600

Thanks for answer,
Does Asterisk need DNS lookups for SIP phones in LAN (without any externhost parameter) that register to it?

I understand that external SIP accounts would fail in case of a DNS problem like this one (it also delays startup by 20 seconds...):
WARNING[27919]: acl.c:243 ast_get_ip_or_srv: Unable to lookup 'sip1.mitelefonovirtual.net'

But the strange thing local SIP accounts (host=dynamic) all become UNREACHABLE, and that sip1.mitelefonovirtual.net actually resolves on the server when trying a ping in the same moment.

resolv.conf (has its own bind server):
nameserver 127.0.0.1
nameserver 62.14.63.145
nameserver 62.14.2.1
By: Anthony LaMantia-2 (anthonyl) 2007-03-09 03:17:47.000-0600

do you have srvlookup set in your sip.conf?
By: frawd (frawd) 2007-03-09 04:40:50.000-0600

no srvlookup set (unless the default is to set it), i attached my sip.conf to the bug just in case.
By: Olle Johansson (oej) 2007-03-13 04:39:25

The log file does not contain SIP DEBUG, there is no way we can analyze what's going on without it and it's required by the bug guidelines... Please add one if you want us to be able to help you explain what's going on here. Thanks.
By: Serge Vecher (serge-v) 2007-03-26 12:58:39

1) Prepare test environment (reduce the amount of unrelated traffic on the server);
2) Make sure your logger.conf has the following line:
  console => notice,warning,error,debug
3) restart Asterisk with the following command:
  'asterisk -Tvvvvvdddddngc | tee /tmp/verbosedebug.txt'
4) Enable SIP transaction logging with the following CLI commands (1.4/trunk commands in parenthesis):
set debug 4 (core set debug 4)
set verbose 4 (core set verbose 4)
sip debug (sip set debug)
5) Reproduce the problem
6) Trim startup information and attach verbosedebug.txt to the issue.
By: frawd (frawd) 2007-04-04 10:46:01

Thanks for everything and sorry for having forgotten the sip debug.

Since I upgraded my server to 1.4.1, then 1.4.2, the problem has not appeared. It was totally random in 1.4.0, and happened once a week approximately.

I believe that some change between 1.4.0 and 1.4.1 resolved the problem, as I didn't upgrade any other program nor change any configuration in the server.

I think you can close this bug, except if you want a sip debug to know what the problem was, in that case i can try to come back to 1.4.0 and wait to check if it  reproduces.
By: Serge Vecher (serge-v) 2007-04-04 10:53:35

If it is fixed in 1.4.2, that's "good enough" reason to close this bug. Thanks for reporting back.