[Asterisk-Users] RE: Need More Simultaneous Voice Channel Capacity on Asterisk

[Matt Roth]

Tadepalli, Hari K wrote:

>(OK - sorry for a 3rd attempt. I see that my message came up with no line   breaks in the first two attempts). 
>
>We are testing Asterisk (1.2.5, configured for an IP PBX) for the number 
>of simultaneous multiple VoIP calls supported. Whenever we increase the 
>number of SIP end points over 250, or the equivalent of 125 caller-callee 
>pairs, our testing fails. In fact, adding even one additional pair of 
>end points over the 250 makes all end points fail. I have pasted the 
>console diagnostics posted by Asterisk below. 
>
>Is there any inherent limitation either in Asterisk or Linux system 
>resources that restricts the SIP end point count to 250? I though 
>Asterisk is an open source SW with no restriction on the number of 
>SIP endpoint seats. As I could see, my CPU has plenty of slack 
>(idle time) left when tested with 250 SIP end points. Hence the 
>desire to increase the simultaneous channel capacity of Asterisk. 
>
>If you have configured your Asterisk IP PBX to serve more than 250 
>SIP end points, I would appreciate some help. 
>
>Thanks, 
>
>Hari Tadepalli
> 
><><><><><><><><><><><><><><><><><>
>Intel Corporation
>Communications Infrastructure Group
>Chandler, AZ
><><><><><><><><><><><><><><><><><>
> 
>//////////////////////////////////////////////////////////////////////// 
>Everyone is busy/congested at this time (1:0/1/0)
>    -- Executing Hangup("SIP/1193-4934", "") in new stack
>  == Spawn extension (from-sip, 1449, 2) exited non-zero on 'SIP/1193-4934'
>    -- Executing Dial("SIP/1065-5adb", "SIP/1321 at 192.169.200.10") in new stack
>    -- Called 1321 at 192.169.200.10
>    -- Executing Dial("SIP/1193-14a4", "SIP/1449 at 192.169.200.10") in new stack
>    -- Called 1449 at 192.169.200.10
>    -- SIP/192.169.200.10-9ef8 is circuit-busy
>  == Everyone is busy/congested at this time (1:0/1/0)
>    -- Executing Hangup("SIP/1065-5adb", "") in new stack
>  == Spawn extension (from-sip, 1321, 2) exited non-zero on 'SIP/1065-5adb'
>    -- SIP/192.169.200.10-f482 is circuit-busy
>  == Everyone is busy/congested at this time (1:0/1/0)
>    -- Executing Hangup("SIP/1193-14a4", "") in new stack
>  == Spawn extension (from-sip, 1449, 2) exited non-zero on 'SIP/1193-14a4'
>    -- Executing Dial("SIP/1193-7d31", "SIP/1449 at 192.169.200.10") in new stack
>    -- Called 1449 at 192.169.200.10
>    -- Executing Dial("SIP/1065-487a", "SIP/1321 at 192.169.200.10") in new stack
>    -- Called 1321 at 192.169.200.10
>    -- SIP/192.169.200.10-807e is circuit-busy
>////////////////////////////////////////////////////////////////////////////////
>_______________________________________________
>--Bandwidth and Colocation provided by Easynews.com --
>
>Asterisk-Users mailing list
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>  
>
Hari,

Try increasing the open file limit prior to running Asterisk by issuing 
the following command at the shell prompt:

[root at asterisk ~]# ulimit -n 65536

If that doesn't fix the problem, edit "/etc/asterisk/logger.conf" so 
that it contains the following line:

messages => notice,warning,error,debug

This will capture diagnostic information to Asterisk's "messages" file 
(usually "/var/log/asterisk/messages").  Rerun your tests and compare 
the contents of "messages" during normal operation and failure.  Be 
warned, if you turn on debugging at the CLI there will be a *lot* of 
information.  To keep things simple, I recommend trying to diagnose the 
problem without debugging on the first couple of runs, then turning it 
on if you need more information.

You can also share the contents of the "messages" file with the list by 
pasting it directly into an email, or using <http://www.pastebin.ca> if 
the contents are very large.

Good luck,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer