[Asterisk-Dev] spike / asterisk hang?

[Matt Hess]

We have a steadily growing dial plan for our users on a sparc netra t105..
I have noticed that audio going through the asterisk (stable) system 
sometimes becomes choppy (cuts out and restores after a few seconds) 
when new calls are being handled.. on the system running a vmstat 1 I 
watch the traps sys counter jumps up really high from the normal run stats..

An example dialplan entry for a user is as simple as:
exten => 201,1,Dial(SIP/201,,t)
(we've got a few hundred of these but note that we only have around 15 
calls active at any given time)
because of the transfer requirement media goes through the asterisk server..
we are using sip on both side of the asterisk server ulaw codec is 
preferred at the endpoints.

Here's a vmstat at 1 second intervals .. at the time there are 8 active 
calls:
 procs   memory        page                    disks     traps         cpu
 r b w      avm         fre   flt  re  pi  po  fr  sr sd0 cd0  int   
sys   cs us sy id
 0 0 0  42288 868840     7   0   0   0   0   0   0   0  367   469   61  
0  0 100
 0 0 0  42288 868840     7   0   0   0   0   0   0   0  395   802   92  
1  0 99
 0 0 0  42288 868840    12   0   0   0   0   0   0   0  377   554   64  
0  0 100
 0 0 0  42288 868840     7   0   0   0   0   0   0   0  359   476   62  
0  0 100
 0 0 0  42288 868840     7   0   0   0   0   0   0   0  364   468   62  
0  0 100
 0 0 0  42288 868840     7   0   0   0   0   0   0   0  375   459   61  
0  0 100
 0 0 0  42288 868840     7   0   0   0   0   0   0   0  364   480   61  
0  0 100

But on processing a call vmstat's sys traps and cpu cs counters both  
jump way up..

 procs   memory        page                    disks     traps         cpu
 r b w      avm         fre   flt  re  pi  po  fr  sr sd0 cd0  int   
sys   cs us sy id
 0 0 0  42280 868848     7   0   0   0   0   0   0   0  709  1429  160  
3  1 96
 0 0 0  42288 868840    11   0   0   0   0   0   0   0  659  1395  168  
0  0 100

This issue has steadily gotten worse with the addition of more dialplan 
entries like the one above..
Note that as long as all calls are active and setup (nothing happening 
involving dialplan) audio is perfect.. only when dialplan processing 
happens does asterisk seem to hang up for a second..

Am I going crazy or am I near the mark in troubleshooting this.. and if 
so (near the mark and not crazy) then how can I help improve the situation?



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