Hello,
I was using 1.6 asterisk for a long time.
My configuration is as follows. SOme of my users(analogue ones) are on
ericsson pbx which is connected to asterisk via e1 interfaces.
And asterisk is dialing out via a sip trunk.
Ericsson has a setting for prefixes as minimum digits and
On Thu, Mar 29, 2012 at 01:58:39PM -0400, sean darcy wrote:
DAHDI 2.6.0, dahdi show status
Description Alarms IRQbpviol CRC
Fra Codi Options LBO
Wildcard TDM400P REV I Board 5 OK 0 0 0 CAS
Unk 0 db (CSU)/0-133 feet (DSX-1)
Hi
I suspect that my telco set-up is acting funny and I want to use spectral
analysis to confirm the culprit :)
What is the best way to generate Fax tones from a dialplan and then record
them at the other end? Also, where can I get a list of the all the tones
and duration which are used in Fax.
Hi
We are trying to accept inbound calls from a SIP provider who sends us calls
from various IP's within a given subnet but they are failing every time with
the following message on the console.
chan_sip.c:20006 handle_request_invite: Call from '' to extension
'destination-number' rejected
Your understanding of the problem seems incorrect. The problem seems due to
the extension not available in your dialplan. Please check carefully in
which context the call is placed and if the extension is defined in that
context.
Maybe it can be useful to define a _X. extension to catch all not
On Monday 02 April 2012, Mark Farmer wrote:
Hi
We are trying to accept inbound calls from a SIP provider who sends us
calls from various IP's within a given subnet but they are failing every
time with the following message on the console.
chan_sip.c:20006 handle_request_invite: Call from
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini
Sent: 02 April 2012 13:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk ACL
Your understanding of the problem seems
On 2 April 2012 14:06, Mark Farmer mark.far...@gagenetworks.com wrote:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro
Dardini
Sent: 02 April 2012 13:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
No, I don't do transcoding, I've disabled all the codec except for the g729.
But in my last test I've found out what is the problem (not yet how to
solve it)
I make all my calls through a php agi, this old script works well on
asterisk 1.4 and I want to move on 1.8.
Just for test I've created
Hi
it's possible into Asterisk 1.6.x to limit a call at 120 mn ?
after 120mn, hangup and the customer call a new time
thanks
olivier
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
Thanks but i read:
; The maximum number of concurrent calls you want to allow
Not limit the duration of a call ;=)
Le 2 avril 2012 16:55, Bakko asannu...@gmail.com a écrit :
Hi,
look at maxcalls parameter on the asterisk.conf file.
regards
El 02/04/2012 16:46, Olivier CALVANO
have you tried the L parameter in the dial command?
* *L(*x[:y][:z]*)*: Limit the call to 'x' ms, warning when 'y' ms are
left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are
optional. Numbers must be integers- beware of AGI scripts that may
return long integers in
are you by chance using the a2billing script?
On Mon, Apr 2, 2012 at 5:43 PM, Syco syco...@gmail.com wrote:
No, I don't do transcoding, I've disabled all the codec except for the
g729.
But in my last test I've found out what is the problem (not yet how to
solve it)
I make all my calls
On Mon, Apr 2, 2012 at 7:44 AM, Mark Farmer mark.far...@gagenetworks.comwrote:
Hi
** **
We are trying to accept inbound calls from a SIP provider who sends us
calls from various IP’s within a given subnet but they are failing every
time with the following message on the console.
Hi
A couple of weeks ago I asekd how to setup a fallback numer and one of
the reply I received was to se GotoIF and ${DIALSTATUS}.
I succeeded in making it work for a single fallback number (i.e. the
operator), but I want to extend it in the following manner:
2000-2099 - fallback to 2000
On Mon, Apr 2, 2012 at 7:05 PM, Paolo Supino paolo.sup...@gmail.com wrote:
Hi
A couple of weeks ago I asekd how to setup a fallback numer and one of
the reply I received was to se GotoIF and ${DIALSTATUS}.
I succeeded in making it work for a single fallback number (i.e. the
operator), but I
On 04/02/2012 08:35 PM, Warren Selby wrote:
On Mon, Apr 2, 2012 at 7:05 PM, Paolo Supino paolo.sup...@gmail.com
mailto:paolo.sup...@gmail.com wrote:
Hi
A couple of weeks ago I asekd how to setup a fallback numer and one of
the reply I received was to se GotoIF and ${DIALSTATUS}.
17 matches
Mail list logo
