[asterisk-users] CPU Spikes in asterisk connected via IAX trunk
Rajkumar S
rajkumars at gmail.com
Fri Aug 14 02:01:30 CDT 2009
Hello,
I have a 3 server asterisk configuration where one asterisk (say A) (v
1.4.25) has a digiuim card connected to E1 from which calls are routed
to another asterisk server (B) (1.6.0.9) over IAX trunk from which
calls get routed to third server (C) (1.6.0.9) again via IAX trunk.
SIP clients are connected to third server. A is the PSTN termination
server, B runs the menu and AGI and C is where SIP clients connect.
SIP clients can also dial outside and call goes like C -> B -> A ->
PSTN.
I am facing some call disturbance for agents connected via SIP in C.
While investigating I found that CPU usage hits 99% occasionally and
in general CPU usage is very un even. Load average also goes up
correspondingly some times till about 30. It has no correlation with
number of calls. Some times even with about 29 calls the cpu is not
much loaded (10%) but it hits 60% 70% with about 6 calls (12 channels)
some other times.
I am using ulaw through out, (disallow=all; allow=ulaw in iax.conf and sip.conf)
One correlation I found was that when ever agents transfer calls to
main menu (ie to server B) there is a load spike. This transfer again
goes via same IAX trunk as the incoming.
IAX conf in C is:
[general]
bindport = 4569
bindaddr = 0.0.0.0
disallow=all
allow=ulaw
jitterbuffer=no
forcejitterbuffer=no
[a16-q1]
type=user
auth=plaintext
secret=password
context=inbound-calls
qualify=yes
trunk=yes
[a16-q1-a16-in1]
type=peer
host=192.168.79.177
auth=plaintext
secret=password
username=a16-in1
qualify=yes
trunk=yes
IAX conf in B is:
[general]
bindport = 4569
bindaddr = 0.0.0.0
disallow=all
allow=ulaw
jitterbuffer=no
forcejitterbuffer=no
transfer = no
[a16-in1]
type=user
auth=plaintext
secret=password
context=inbound-calls
qualify=yes
trunk=yes
[a16-in1-a16-q1]
type=peer
host=192.168.79.176
auth=plaintext
secret=password
username=a16-q1
qualify=yes
trunk=yes
I am pretty much stumped here. Could IAX trunk be the source of the
problem? Should I switch to SIP ?
thanks and regards,
raj
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