Hi Asterisk Users,
I would like to handle about 250 simultaneous (calls agents only) calls
with PRI or a SIP trunk with the following configuration
Dell R710
Dual Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz or Single
Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz
On 11-02-01 05:22 PM, Juan David Diaz wrote:
I would like to handle about 250 simultaneous (calls agents only) calls
with PRI or a SIP trunk with the following configuration
Dell R710
Dual Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz or Single
Intel® Xeon® X5650, 2.66Ghz, 12M
Tuesday, February 1, 2011, 11:22:30 PM, Juan wrote:
I would like to handle about 250 simultaneous (calls agents only) calls
with PRI or a SIP trunk with the following configuration
Dell R710
Dual Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz or Single
Intel® Xeon® X5650,
That's quite possible. We handle around 100 similtaneous calls(PRI +
SIP) with a decent dell server with only 4gb ram.
On Wed, Feb 2, 2011 at 6:22 AM, Juan David Diaz juanch...@gmail.com wrote:
Hi Asterisk Users,
I would like to handle about 250 simultaneous (calls agents only) calls
with PRI
On Sun, 9 Dec 2007, jorain wrote:
Thanks for your replies.
1.. Our connection mainly for voip, occasionally used for surfing
websites.
2.. We are using codec g711u for local calls through TE120P, and g729
only if making international calls through our sip provider, which only
allow g723
shapping would you
recommend?
5.. How many users can we expect to use voip(with good quality) with 512kbps
outbound connection?
Regards,
jorain
Date: Fri, 7 Dec 2007 10:27:36 -0500
From: C F [EMAIL PROTECTED]
Subject: Re: [asterisk-users] asterisk performance
To: jorain [EMAIL
by 3rd call do you mean over the internet?
if the answer is yes, then I wouldn't be surprised. another thing what
codec are you using?
On 12/6/07, jorain [EMAIL PROTECTED] wrote:
Hi all,
We are using
- a dell sc440(Single dual-core intel xeon 3040, 1.86GHz,1066MHz front size
bus 2MB cache)
Hi all,
We are using
- a dell sc440(Single dual-core intel xeon 3040, 1.86GHz,1066MHz front size bus
2MB cache) as the asterisk server
- dell 400sc(Intel P4) as a SER server
- digium isdn card, TE120P at Asterisk server
- Bandwidth: 2Mbps/512kbps
All SIP Phones are registered to SER server,
Your 512k outbound bandwidth will tend to be the defining factor in
call quality here.
Does your connection only gets used for voip? Or is it shared with
other uses?
Can you use more compressed codecs? G729 will quadruple you call
capacity.
What sort of QoS and traffic shaping do you use?
2007/12/7, C F [EMAIL PROTECTED]:
by 3rd call do you mean over the internet?
if the answer is yes, then I wouldn't be surprised.
Oh my god!
If it is over internet and you get crap quality.. you have to be surprised..
It is depends by Latency (Traffic congestion, Network congestion) and
Packet
On Tue, 29 Aug 2006, Nick Hoffman wrote:
On Tue August 29 2006 04:39, Greg Boehnlein [EMAIL PROTECTED] wrote:
On Mon, 28 Aug 2006, Andrew Kohlsmith wrote:
On Monday 28 August 2006 13:02, Greg Boehnlein wrote:
I've pushed over 1,000 concurrent calls this way using the SIPP
program
On Sat, 26 Aug 2006, Kelvin Williams wrote:
If Asterisk was used to set up and tear down calls, and using canreinvite
allowing the RTP to pass from end-point to end-point, how many calls could
Asterisk handle at once?
I've pushed over 1,000 concurrent calls this way using the SIPP program
On Monday 28 August 2006 13:02, Greg Boehnlein wrote:
I've pushed over 1,000 concurrent calls this way using the SIPP program
for SIP performance testing. There was some tuning that needed to be done,
but it worked. Never went that far in production, though.
May you share some of your tuning
On Mon, 28 Aug 2006, Andrew Kohlsmith wrote:
On Monday 28 August 2006 13:02, Greg Boehnlein wrote:
I've pushed over 1,000 concurrent calls this way using the SIPP program
for SIP performance testing. There was some tuning that needed to be done,
but it worked. Never went that far in
On Tue August 29 2006 04:39, Greg Boehnlein [EMAIL PROTECTED] wrote:
On Mon, 28 Aug 2006, Andrew Kohlsmith wrote:
On Monday 28 August 2006 13:02, Greg Boehnlein wrote:
I've pushed over 1,000 concurrent calls this way using the SIPP
program for SIP performance testing. There was some
If Asterisk was used to set up and tear down calls, and
using canreinvite allowing the RTP to pass from end-point to end-point, how
many calls could Asterisk handle at once?
I ask because I have been utilizing OpenSER but find myself constantly
needing Asterisk to do this or that, and
Hi All,
This is a performance update. I have built appliance type servers
with the following specs:
Motherboard Asus P5MT-M
Memory 1Gig DDR2
No hard drive, running in Ramdrive but using Sandisk Compact Flash to
hold compressed image and /var directory
Processor 3.2 Gig Pentium 4, HT Turned Off
Is this with Asterisk in the RTP stream? Is it doing any transcoding?
-Original Message-
From: JR Richardson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 18, 2006 9:34 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk Performance 350 Concurrent
Asterisk was in the RTP and no transcoding, straight Ulaw g.711.
--
JR Richardson
Engineering for the Masses
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How much can be the load (How much register and
calls Asterisk can Handle simultaneously by asterisk) and what will be the
performance of Asterisk (Call Quality) if all the users are on SIP only and uses
same Codec, I have all three codecs loaded G.711, G.723, G.729) without media
support
PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, February 18, 2005 4:15 AM
Subject: [Asterisk-Users] Asterisk Performance in comparission of SER
How much can be the load (How much register and calls Asterisk can Handle
simultaneously by asterisk) and what will be the performance
I'm not sure it answers all your questions but there is ast-stats from
http://areski.net/areski/index.php?
option=com_contenttask=categorysectionid=5id=70Itemid=54
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Title: Asterisk performance monitoring
Hello,
Has anyone used any 3rd party web based software to get performance information out of Asterisk?
Looking for CPS, call setup times, voicemail database utilization etc
Cheers
Keith.
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