Tadepalli, Hari K
2006-Apr-03 17:22 UTC
[Asterisk-Users] Need More Simultaneous Voice Channel Capacity on Asterisk
Hi, 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 <><><><><><><><><><><><><><><><><> ?
Tadepalli, Hari K
2006-Apr-03 17:25 UTC
[Asterisk-Users] RE: Need More Simultaneous Voice Channel Capacity on Asterisk
(Sorry for resending. I forget to attach screen dump from Asterisk console in
the first attempt).
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@192.169.200.10") in new stack
-- Called 1321@192.169.200.10
-- Executing Dial("SIP/1193-14a4",
"SIP/1449@192.169.200.10") in new stack
-- Called 1449@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@192.169.200.10") in new stack
-- Called 1449@192.169.200.10
-- Executing Dial("SIP/1065-487a",
"SIP/1321@192.169.200.10") in new stack
-- Called 1321@192.169.200.10
-- SIP/192.169.200.10-807e is circuit-busy
////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
Tadepalli, Hari K
2006-Apr-03 17:28 UTC
[Asterisk-Users] RE: Need More Simultaneous Voice Channel Capacity on Asterisk
(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@192.169.200.10") in new stack
-- Called 1321@192.169.200.10
-- Executing Dial("SIP/1193-14a4",
"SIP/1449@192.169.200.10") in new stack
-- Called 1449@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@192.169.200.10") in new stack
-- Called 1449@192.169.200.10
-- Executing Dial("SIP/1065-487a",
"SIP/1321@192.169.200.10") in new stack
-- Called 1321@192.169.200.10
-- SIP/192.169.200.10-807e is circuit-busy
////////////////////////////////////////////////////////////////////////////////
Tadepalli, Hari K
2006-Apr-04 08:33 UTC
[Asterisk-Users] RE: Need More Simultaneous Voice Channel Capacity on Asterisk
Thanks for the suggestions. I have done both the following:> echo "8192" > /proc/sys/fs/file_max > ulimit -n 8192That has cured the problem. I could now run with 255 channels. - Hari -----Original Message----- From: Tadepalli, Hari K Sent: Monday, April 03, 2006 5:26 PM To: 'asterisk-users@lists.digium.com' Subject: RE: Need More Simultaneous Voice Channel Capacity on Asterisk (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@192.169.200.10") in new stack -- Called 1321@192.169.200.10 -- Executing Dial("SIP/1193-14a4", "SIP/1449@192.169.200.10") in new stack -- Called 1449@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@192.169.200.10") in new stack -- Called 1449@192.169.200.10 -- Executing Dial("SIP/1065-487a", "SIP/1321@192.169.200.10") in new stack -- Called 1321@192.169.200.10 -- SIP/192.169.200.10-807e is circuit-busy ////////////////////////////////////////////////////////////////////////////////