Hi all,
I face a strange problem. I'm in France using Completel as operator for
the E1 line. I move a client from ccm to Asterisk keeping the 2811 gateway.
Set up is complete, outgoing and incoming calls are sended to the 2811.
The problem is that in 90% of the time I get a 500server error
Hello;
Can someone advise me what is the maximum number of users (IP Phones) that can
be supported by asterisk 1.8 or later?
Regards
Bilal--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
The better question is, maximum number of users (IP Phones) can your
hardware support. I have * deployments with 300-600 phones. - works fine.
though concurrent calls has never seen more then 244. Also at this point I
have to ask, for this to be any concern to you, you must either A, Make
tons of
You can have tens of thousands of phones as long as no one makes or receives
any calls J. The better question to ask is how many concurrent calls have
people been able to make. The quick answer is it depends on many things.
John
From: asterisk-users-boun...@lists.digium.com
How did the system behave with 244 calls? I've been able to make 1,024
concurrent faxes (which tend to use more resources than audio calls) in the
lab. The problem I had was after the faxes were transmitted, things couldn't
keep up and kept dumping core. Two things were going on, (1) the CDR
To be honest I am not sure, I pulled the data from a cacti graph shortly
before posting my reply. I imagine it all ended well, as I don't recall
hearing complaints on quality. I have noticed the core dump issue mixed
with * and SQL a few months ago. We had about 120-150 users in a
conference, that
We have a machine with a quad core 'Intel(R) Xeon(R) CPU E5-1410 0 @
2.80GHz' running asterisk 11.2-cert with ingress and egress all sip.
Fastagi running as a daemon (written in perl) performing cdr updates at
call start, answer and call end together with a query when a call comes
in to get
Have you ever checked out the app_konference module? You can check it out
here. http://sourceforge.net/projects/appkonference. I have a customer who
routinely hosts 100+ users in a conference without issue. We've had very
good results so far. We're hoping to eventually hit 500+ users in the future
Yeah. I started looking at this a few weeks ago. I am going to do a trial
deployment in the new year. Where are you located in the world?
Regards,
Keith Sloan
Voice Operations Center
Vianet
705-222-9996 X7203
1-800-788-0363 X7203
kei...@vianet.ca
On Wed, Dec 18, 2013 at 12:05 PM, Tech Support
Central Maryland, USA. About an hour NW from Washington, DC.
John
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Keith Sloan
Sent: Wednesday, December 18, 2013 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hello. I have a problem with the configuration of a remote extensions.
Calls are truncated at 20 seconds.
I got my my NAT firewall properly configured. Here I attached my debug in
CLI: http://pastebin.com/gh34E69f
Thank you!
--
Allan Porras
http://allanPorras.com http://www.AllanPorras.com
Calls dropping after 20 seconds is often directmedia enabled when it should not
be enabled or RTP keepalives enabled when they should not be enabled. Dropping
around 20 mins is often Session Timers being enabled when they don't work for
the specific environment.
-Original Message-
Thank you Eric for your reply. How Can I fix it?
In server side, I opened RTP ports.
On Wednesday, December 18, 2013, Eric Wieling wrote:
Calls dropping after 20 seconds is often directmedia enabled when it
should not be enabled or RTP keepalives enabled when they should not be
enabled.
here's a checklist...
First, RTP port range not port forwarded correctly on the NAT router (check
rtp.conf).
Then, on sip.conf:
externip not correctly setup (it should be the public IP of the NAT
router)?
nat setting not enabled for any outbound trunk and the extensions (nat=yes)
?
localnet
Rodrigo, thanks for reply.
1- RTP ports is forwarded correctly on the NAT router.
2- externip is my public ip.
3- All my extensions have nat=yes by default.
4- localnet is setup.
5- canreinvite is disabled.
It could be a codec mistake?
On Wed, Dec 18, 2013 at 2:58 PM, Rodrigo Borges Pereira
1.4 1.6 1.8 11.6.0
All compiled and all running on debian 6 or 7
On 16 December 2013 12:27, Dotan Cohen dotanco...@gmail.com wrote:
On Mon, Dec 16, 2013 at 12:41 AM, dotnetdub dotnet...@gmail.com wrote:
Always has cleared the entire line..
Interesting, thanks. From where is your
What version of Asterisk?directmedia=no should be used in versions of
Asterisk 1.8 and later.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com
Sent: Wednesday, December 18, 2013 4:23 PM
On 12/18/13, 3:09 PM, alp...@gmail.com wrote:
Hello. I have a problem with the configuration of a remote extensions.
Calls are truncated at 20 seconds.
I got my my NAT firewall properly configured. Here I attached my debug
in CLI: http://pastebin.com/gh34E69f
When the call is setup I see
Hi,
Trying to properly broadcast / relay DTMF digits to other confbridge users, but
does not appear to work. Goal is to have a conference user be able to receive
the DTMF, so it has the effect of being 'broadcasted.'
I have the following set up in 'confbridge.conf':
dtmf_passthrough=yes
From
Hi Bilal,
Assuming you have the latest hardware, sufficient memory, cpu, etc...
The key to determine the maximum number of users comes down to the
office type, RTP path, network interface, and primary codec used.
First we need to determine the over-subscription rate, how many people
will be
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