[asterisk-users] RTP and Signalling Dropping
Jon Farmer
jon at bctech.co.uk
Tue Apr 19 11:09:43 CDT 2011
Hi
I have a weird issue with a new 1.6.2.17.2 box.
At random intervals it just stops responding to RTP and signalling
(both SIP and IAX observed). All calls in progress lose audio both
ways although the console shows the call legs still in progress. No
signalling can be sent or is received. It is as though the server
drops of the net for those protocols. I can still navigate the
console. Killing an restarting Asterisk is the only way to bring it
back. I can see nothing in the logs to indicate what is happening.
The server is dual homed network one interface on a public address and
the other interface on a private subnet that the phones sit on.
It can do 100's even 1000's of calls before the issue happens and then
BAM it drops off. The box is handling between 1500 - 3000 calls a day,
mostly SIP and IAX with a small percentage of DADHI.
Anyone any ideas what is going on or where to look next?
Regards
Jon
--
Jon Farmer
Tel 07795 118140
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