On 6/13/07, Josu Lazkano <[EMAIL PROTECTED]> wrote:
>
> How can I saw the status of the ISDN???
>
...try "misdn show stacks" or "misdn show config". You can also increase
debug level in /etc/misdn-init.conf... Output will end up in
/var/log/asterisk/misdn.log
Cheers,
--
Ex Vito
_
You have only one extension in the [incoming] context and that is
's'. You probably need a different one -- the one the telco sends
you...
Ideas:
1. Try using a generic wildcard such as '_X.' instead of 's', then
check the CLI after incrementing verbosity to at least 3
(B
...not really sure, maybe ChanIsAvail can be of use ?
--
Ex Vito
On 6/20/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Is it possible to force the Dial function to skip to the next priority if it
> doesn't find the server of the called contact within a few seconds?
>
> I know I can use:
>
> Someone has worked with any test to speech software with aceptable
> quality in spanish? Probably in english the text to speech quality
> will be better.
> Witch test to speech software gave you the best results in spanish?
>
Hi Andres,
Check www.loquendo.com out... They have a nice web f
In a dialplan, after i set an autohangup (with AGI) , how could i send a
sound (stream a sound ) into an open channel at X seconds before the
autohangup time get to 0 for that channel?
(Like public phones, that gives u a 'beep!!!' before ur time runs out, just
like that...)
Check the L opti
On 2/15/07, Angel Heart <[EMAIL PROTECTED]> wrote:
cud any one help me figuring out the problem... When the agent in a queue is
engaged, it cannot accept anymore calls, below is the script;
Angel,
Check your queues.conf, specifically the "joinempty" parameter.
See below the relevant part in
Hello List,
The main issue is server selection regarding PCI bus connectivity
for Asterisk solutions. Most offerings on HP Proliants, something
I've been looking into, include PCI-X and/or PCI-e expansion slots.
I know PCI-e is totally different from PCI and PCI-X so, for now,
that's not an iss
...well, thanks Andrew + Thomas, but that's exactly
what I am trying to avoid: knowing by trying ! ;-)
I can't risk spending a few thousand just to reach the
conclusion that Digium's PRI or BRI cards do not work
with a particular system's PCI-X slots/bus... Or, worse,
staying with a dead ca
Hello Asterisk Users,
I guess the subject says the most of it; here goes some more
detail:
- Running Asterisk 1.2.14
- Objective: record all calls managed by a specific queue
- Name those files ${TIMESTAMP}-${CALLERIDNUM}-${UNIQUEID}
Facts:
- If the UNIQUEID chan var is used in the MO
(1st attempt was rejected by postfix @lists.digium.com, here goes the 2nd)
-- Forwarded message --
From: Ex Vitorino <[EMAIL PROTECTED]>
Date: Dec 18, 2006 11:41 PM
Subject: Queue Monitor not mixing if using UNIQUEID in MONITOR_FILENAME
To: Asterisk Users Mailing List
James,
Thanks a lot for sharing the result of those debugging hours ! :-)
I'm now left with two choices to begin with:
1. Replacing the "." with a "-" within the dialplan
2. Replacing the "Ubuntu Server" packaged sox version (12.17.9)
with the most recent (12.18.2) which no longer s
Jay,
I had a similar issue recently... My filename had more than one "."
(dot / period)
and the sox version I was using failed to mix files in such conditions...
If that is your case, try:
- Using a filename with no "."
- Upgrade sox to the latest version which fixes the funny behaviour
Nikola,
Check the maxlen parameter for the queue... Also check the sample
queues.conf distributed with Asterisk source, which somehow includes
queue parameter documentation.
If set, maxlen will limit the number of calls in the queue.
Cheers,
--
Ex Vito
On 1/2/07, Nikola Ciprich <[EMAIL
/var/lib/asterisk/licenses
:-)
On 1/8/07, Xue Liangliang <[EMAIL PROTECTED]> wrote:
Hi, leo, I will try the following solution that seperate
/usr/lib/asterisk/modules in another patition other than drbd, then
register the licenses on both server. not sure where the license key
acutally lies
We've been using logrotate without any issue... We're using
the below quoted configuration. Notice the invocation of
Asterisk's CLI "logger reload" command so as to close the
old files and open new ones.
Cheers,
--
Ex Vito
/var/log/asterisk/messages /var/log/asterisk/queue_log
/var/log/a
On 1/8/07, lenz <[EMAIL PROTECTED]> wrote:
You know that if you rename an open Unix file, it will stay open - i.e. if
you rename the logfile "full" to "full.1", Asterisk will continue writing
to "full.1" thinking it was "full".
The "logger rotate" command forces all log files to be closed and re
Hello List,
This must be an easy one... I'd like to setup a queue
without music on hold - just give the callers the traditional
ringing tones.
However, not setting the musiconhold parameter in
queues.conf does not seem to do the trick: it defaults
to "default" moh class which:
a) Gets p
r any feedback and kind regards,
--
Ex Vito
-- Forwarded message --
From: Ex Vitorino <[EMAIL PROTECTED]>
Date: Jan 2, 2007 1:22 PM
Subject: Trying to understand kernel buffer messages...
To: misdn-asterisk@lists.beronet.com
Hello list,
I've been successfully run
I've been through this once and, IIRC, I'd say your [outside]
context in the dialplan does not have an extension matching
the incoming msn.
Suggestion:
- Recheck your extensions.conf [outside] context
Otherwise:
- Increase misdn debug to level 3
(something like "misdn set debug 3" a
On 1/18/07, Cosmin Prund <[EMAIL PROTECTED]> wrote:
How about the "Digium Wildcard B410P" card? It seems to be Digium, it
has hardware echo cancel and I can buy this in Romania. Is this card any
good?
...well, it's essentially a 4 port HFC based card with builtin
HW echo canceler.
I'd say i
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