I just added a new verison of Asterisk to the repo. It has a bunch of fixes for load related crashing.
I just added a new verison of Asterisk to the repo. It has a bunch of fixes for load related crashing.
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Did this change the run as user?
I just applied this update and when I restarted asterisk the asterisk process was running as "nobody" instead of "asterisk". I could have sworn that it was running as asterisk previously. All of the previous log files are owned by asterisk...
I'm trying to track down a copy of the previous RPM to confirm or deny this, but it's eluding me.
the asterisk process should
the asterisk process should always run as user asterisk and group asterisk.
I just checked a machine I have here running the latest asterisk RPM and it seems ok
ps -ef
/usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c
you can see that on a ps command asterisk has been start with the user asterisk. What make you think yours is running as nobody?
one way Audio Problem
Hi,
We are using TrixBox ver:v2.8.0.1 & have a one way audio problem.
The current trixbox setup as follows:
Trixbox -> firewall-> Router and sip trunk registered with sip server sucessfully.
another sip trunk used by Xlite softphone which are the behind the same firewall & router.
when i call to trixbox extension from Xlite softphone using external trunk line, signalling is ok but no audio transmited from Xlite to Trixbox extensions.
I have observe from wireshark that RTP packets transmitted from Xlite softphone are going towards the WAN IP but RTP packets transmitted from Trixbox extn. towards the Asterisk server.
could any one help me or provide the useful link to solve this problem.
Ashwani
Look at your firewall
Look at your firewall configuration to see if you have forward the rtp ports from your external firewall ip to the trixbox server. The RTP ports are UDP 10000 to 20000. About the extension sending the rtp packets to the server, thats a normal behavior asterisk will always be in between. Its not a pure SIP server
regards,
Ivan