On Fri, Jun 30, 2017 at 03:15:21PM +0200, Stefan Viljoen wrote:
> Hi all
>
> I'm trying to limit the maximum concurrent calls on my Asterisk to try and
> mitigate another problem I posted about earlier.
>
> I've edited /etc/asterisk/asterisk.conf
>
> And uncommented this line, and put a value
You should try to limit it in your sip trunks (is you are using SIP
trunks, of course)
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
On 30 June 2017 at 15:41,
This limit is only valid for inbound calls:
Sets a maximum number of simultaneous inbound channels. No limit is
set by default.
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
Hi all
I'm trying to limit the maximum concurrent calls on my Asterisk to try and
mitigate another problem I posted about earlier.
I've edited
/etc/asterisk/asterisk.conf
And uncommented this line, and put a value of 60 in there:
maxcalls = 60
in an effort to limit my
Dear list
anyone know wich is the limit of maxload into asterisk.conf ?
Also the meaning is related to RAM ? or CPU ?
Regards Andrea
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--- Vieri [EMAIL PROTECTED] wrote:
I set uniquename = MYHOST in asterisk.conf (under
[options]) so that my uniqueid data shows up as
MYHOST.time.seq.
First of all, I would like to know if uniquename (or
sysname?) will still be valid across future *
versions
(mainly 1.6).
Secondly, is
On 3/18/08, Vieri [EMAIL PROTECTED] wrote:
--- Vieri [EMAIL PROTECTED] wrote:
I set uniquename = MYHOST in asterisk.conf (under
[options]) so that my uniqueid data shows up as
MYHOST.time.seq.
First of all, I would like to know if uniquename (or
sysname?) will still be valid
I set uniquename = MYHOST in asterisk.conf (under
[options]) so that my uniqueid data shows up as
MYHOST.time.seq.
First of all, I would like to know if uniquename (or
sysname?) will still be valid across future * versions
(mainly 1.6).
Secondly, is there a way to specify uniquename as an
I was previous using Asterisk 1.2.9.1 and decided to get some real servers
outside of my house. It was time for Asterisk 1.4.4.
I figured since all the conf files were in /etc/asterisk form the old box,
i'd just copy tha directory over to the new server. My SIP DID AGI stuff
worked, except
this message is basically tells you asterisk is not running.
can you check and see if asterisk is running and present in memory?
something like
ps -ef | grep asterisk
On 10/20/07, Dominic Son [EMAIL PROTECTED] wrote:
I was previous using Asterisk 1.2.9.1 and decided to get some real
servers
astrundir = /var/run
Change this to astrundir = /var/run/asterisk on 1.4 server and chmod
/var/run/asterisk to 777 . make sure u create that directory as well .
On 20/10/2007, Al lists [EMAIL PROTECTED] wrote:
this message is basically tells you asterisk is not running.
can you check and see
On Sat, Oct 20, 2007 at 11:57:05PM +0530, Jaswinder Singh wrote:
astrundir = /var/run
Change this to astrundir = /var/run/asterisk on 1.4 server and chmod
/var/run/asterisk to 777 . make sure u create that directory as well .
chmod 777 (or even 666) to the control socket (asterisk.ctl)
awesome. it worked. thanks guys.
On 10/20/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sat, Oct 20, 2007 at 11:57:05PM +0530, Jaswinder Singh wrote:
astrundir = /var/run
Change this to astrundir = /var/run/asterisk on 1.4 server and chmod
/var/run/asterisk to 777 . make sure u create
Why there is no asterisk.conf.sample file?
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Adrian A wrote:
Does anyone know what exactly the option
transmit_silence_during_record in asterisk.conf does? Is this
useful
for voicemail recording?
Could the option be named any more explicitly? It does _exactly_
what
it says it does.
Some providers terminate the connection
Steve Totaro wrote:
Thanks for the useful answer to this Leif. Yes, it does _exactly_ what
it says it does but the usage of the option remained elusive to me.
Well, I was being in an ultra-pedantic mode when I answered your
question... which was 'what does it do', not 'when would I need
Adrian A wrote:
Does anyone know what exactly the option transmit_silence_during_record in
asterisk.conf does? Is this useful for voicemail recording?
Could the option be named any more explicitly? It does _exactly_ what it
says it does.
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Adrian A wrote:
Does anyone know what exactly the option
transmit_silence_during_record in asterisk.conf does? Is this useful
for voicemail recording?
Could the option be named any more explicitly? It does _exactly_ what
it says it does.
Some providers terminate the connection if nothing is
Hi,
Does anyone know what exactly the option transmit_silence_during_record
in asterisk.conf does? Is this useful for voicemail recording?
Providers such as Broadvoice and Vonage hang up the channel after 30
seconds because they think the other party is not there when someone
records a voicemail.
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