Maybe a hardware problem? What does zttool and ztcfg -vvv say? Is Zaptel
running?
-- Original Message --
From: OCOSA ListAcct [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Date: Sat, 14
Hello, I'm deploying asterisk on a comtrend mips adsl router, I'm aware of the
dependence of libncurses, so I compiled ncurses 5.6 for that platform, As you
must Know this devices are not resource wide and flash memory especially,
after ncurses compilation I have a /usr/share/terminfo with 1,6
Is there any way to set the targeting ip that is sent out in the
dundi answer (to my public ip or any other where i want to receive the
call)?
Change your mapping in dundi.conf to reflect your true public IP rather than
using ${IPADDRESS}.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur
Yeah thats what I thought I found everything running so I just upgraded
and fixed the problem.
Otis
Anthony Francis wrote:
Maybe a hardware problem? What does zttool and ztcfg -vvv say? Is Zaptel
running?
-- Original Message --
From: OCOSA
APC makes a two line unit. PTEL2. But it's two lines in one jack.
Another - www.ablecom.com is a bit more Pro
Just do a google and take your pick.
joe a.
On 7/14/2007 at 10:17 PM, Todd H [EMAIL PROTECTED] wrote:
I lost one channel on an FXO module on a Sangoma A200 card due to a
Joe acquisto wrote:
APC makes a two line unit. PTEL2. But it's two lines in one jack.
I have always been a fan of Triplite. They use old tech when
appropriate. I am big on Line Conditioners and UPSs with line
conditioning. Of course power in my house is really bad. Anything big
kicks on
Hi guys,
I'm in the process of setting up an Asterisk server over a satellite
connection to allow people on a remote island to place and receive calls
over the pstn.
What are the ideal settings I should use in iax.conf for the optimal
operation over satellite besides the normal options for the
I have a client that is using SIP over satellite with G729, VAD and Jitter
buffer. The calls are coming in great.
- Original Message -
From: Tom Moore [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Cc: [EMAIL PROTECTED]
Sent: Sunday, July 15, 2007 5:50 PM
Subject:
Sip would probably work well in a single phone situation, but what I'm
trying to do is use multiple phones over a single trunk connection.
Using sip though do you have a few seconds at the beginning of each call
where the audio is not clear?
On our link when I tried a sip phone the connection was
Tom,
The 2.5 second latency is probably what is causing it. I do not know how
long it takes for the call to stabilize but I am sure that if it took 2.5
seconds that I would of heard about it already. You have to look in to why
in the initial 30 seconds the latency is at 2.5 seconds. I know that
after recompilling asterisk (trunk-r75109) after system (mandriva
cooker) update (new glibc 2.6, gcc 4.2.1),
sound starts very choppy, when codec translation is performed,
if translation isn't needed, it sounds OK
any idea? until update, everything worked fine.
I'm using ztdummy as clock source.
asterisk-1.4.7, Fedora 7, intel emt64 - nocona:
== Parsing '/etc/asterisk/alsa.conf': Found
ALSA lib pcm_dsnoop.c:558:(snd_pcm_dsnoop_open) unable to
open slave
[Jul 15 10:12:23] ERROR[24420]: chan_alsa.c:365
alsa_card_init: snd_pcm_open failed: No such file or directory
[Jul 15 10:12:23]
Tzafrir Cohen schrieb:
On Sat, Jul 14, 2007 at 01:23:35PM +0200, Christophorus Laube wrote:
Hi list,
I am searching for a possibility to do a certain call transfer method
which is called path replacement in QSIG. But I want to do that in
DSS1 (EuroISDN). If my asterisk does a call
Doug Lytle wrote:
Looks like mail is getting held up between INXS.digium.internal and
lists.digium.com
Here's what I get:
---cut---
Received: from lanai.amooma.com ([127.0.0.1])
by localhost (lanai.amooma.com [127.0.0.1]) (amavisd-new, port 10024)
with ESMTP id DYhd6TXxmG6B;
Hi!
I found an old feature-request bug in Zaptel which seems relevant:
http://bugs.digium.com/3554
Not sure if this means that the feature is supported. Maybe ask Mathew
Fredrikson or Digium support.
by the way: Is this call deflection or ECT etc. only possible to be
executed at ring
On Wed, 11 Jul 2007 19:57:05 +1000, Bill Maidment wrote
On Wed, 4 Jul 2007 17:37:29 +0200, Christian Victor wrote
I have the same problem. My mail sent yesterday around 20:00h and it still
not arrived at the list. Sent from germany by the way.
Christian
email delays here are about 8
app_valetparking listed here http://www.freeswitch.org/asterisk_stuff/
Indicates support for Asterisk 1.4. The documentation listed suggests an
install like so:
cd /usr/src/asterisk
cp contrib/scripts/astxs /usr/bin/
cd apps
wget http://www.bkw.org/app_valetparking.c
cd ..
astxs -install
Hi Ira -
In the end my issue would seem to be I did something out of order,
though I thought I did it right, and as much as I tried the software
only gave meaningless to me messages. I didn't understand and I don't
think I've seen it said before that I should run .\configure every
time I do
18 matches
Mail list logo
