los Chavez wrote:
On Mon, 2007-10-22 at 15:35 -0400, Rurouni Alucard wrote:
Saludos Carlos,
Como vas a recibir las llamadas via SIP, puedes especificar el IP del
host que te enviara las llamadas, por ej.
Este es un bloque que tengo definido en el SIP.conf de uno de mis
servers para enrutar la
Saludos Carlos,
Como vas a recibir las llamadas via SIP, puedes especificar el IP del
host que te enviara las llamadas, por ej.
Este es un bloque que tengo definido en el SIP.conf de uno de mis
servers para enrutar las llamadas internacionales y a telefonos moviles
utilizando un proveedor de
Hi there,
I have use the H.323 module that comes with asterisk-addons and i
consider it (so far) VERY stable for my needs.
Im talking about 10,000 minutes at month , + or - , and never had a
crash or something bad about it.
Personally, i recommend it,
--
J. P.
rakh at slackware-es dot com
Hi Roy,
Look I dont know why u specify 'zap/1-1', but i do things like this on
my agi scripts a lot of times:
...
$stdin= fopen('php://stdin', 'r');
$stdout = fopen('php://stdout', 'w');
$stdlog = fopen('/tmp/outPUT.log', 'a');
...
fwrite($stdout,"EXEC DIAL \"Zap/g2/1809220355
When using Grandstreamg Handytone ATA everything works fine incoming/ougoing
but when using Linksys SPA 2002 ATA 'sip show peers' marks those extensions
as "UNREACHABLE" and can't receive calls, but they can call out.
Any Idea about possible reasons ?...
Rurouni,
Did you check your mpg123 version ?, asterisk needs
a specific version in order to work...
- Original Message -
From:
Richard Reina
To: asterisk-users@lists.digium.com
Sent: Wednesday, June 07, 2006 6:02
AM
Subject: [Asterisk-Users] Music On Hold
not wor
