Hi All,
- I have Digium cards and given that the archives point out the Digium
cards drop packets does anyone know what hardware would not do this?
(i.e. Allow me to send outbound faxes)
- If there is still an issue with the wctdm driver, does anyone know
which asterisk/spandsp combo would
Hi All,
With spandsp.0.0.2 pre20 installed I can't seem to send faxes with
tx_fax over a Zap channel (POTS). rx_fax works just fine so no issues
with libtiff and (presumably) libxml2.
Basically I get 'slow carrier up' and 'slow carrier down' together with
accompanying beeping noises until
Thank you. But, I am fairly certain that I am using the proper
versions. They certainly match what is in the README!!!
Perhaps I have some contagion from a previous install, but finding it
seems to be defeating me.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
on.
However, in my case, it moves directly to dialout whether
is entered or nothing is entered.
Can anyone enlighten me please?
JC
Use the shell command ! to exit to shell.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ryan Parlee
Sent: Sunday, April 04, 2004 3:56 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Unabled to exit console
No matter what I try, Asterisk won't let
Well, I hope it works because we are installing one in a new office
tomorrow! Ill let you know how it goes.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thomas
Mangin
Sent: Wednesday, March 31, 2004 5:02 PM
To: [EMAIL PROTECTED]
Subject:
PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] incoming SIP calls drop on pickup.
Sounds like a codec mismatch to me. I had a similar
problem with ICH.
On Mon, 29 Mar 2004 19:23:15 +0100, jc wrote:
Hi All,
I have an annoying problem. Out going SIP/sipphone.com
calls work fine
it further??
Thanks
JC
what does this mean and how can I debug it
further??
Thanks
JC
What version of asterisk
are you using, and what version of the SNOM firmware?
--Ernest
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jc
Sent: Tuesday, March 30, 2004
10:20 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
Exception flag set - snom200
Sorry I forgot
I had the same problem. It turned out that X was hogging all the
resources available. Running without it, the problem cleared up.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Bridges
Sent: Monday, March 29, 2004 9:09 AM
To: [EMAIL PROTECTED]
a recording.
The debugs dont produce an obvious error.
Thanks
JC
I have an 8 channel isdn30e coming in from BT. Can
anyone point me to sample zap*.conf that will work. Thanks JC
I have the following problem in playback:
When any sound file is played back, it is garbled for a few
seconds and the following error displays:
Sched_settime: Request to schedule in the past?
After about 5 seconds, the sound clears up and the error
stops.
What gives???
For some reason, in an otherwise working * installation, I
cant hear any of the sound files. It doesnt seem to matter what
phone make or codec I use. Any help would be appreciated.
JC
I have a simple * setup with a couple of SNOM200
installed. I can make IP calls and internal calls fine. But, I
cant hear any of the asterisk sound files on playback. Any
ideas
No, I don't have any errors. Perhaps it is a codec thing.
jc wrote:
I have a simple * setup with a couple of SNOM200 installed. I can make
IP calls and internal calls fine. But, I can't hear any of the
asterisk sound files on playback. Any ideas
are there any errors on the CLI? I had
Hello all, I am using voicepulse DID's to receive
calls via IAX to and asterisk IVR dial plan I have put together. The problem is
after 3-5mins the system cant pickup the DTMF tones I am sending... I have tried
different telephones... It just repeats menu options over and over I have to
If you want to test it, once you include Zapateller in your dial plan, place
an incoming and block your caller id (from the phone your testing from) and
it will do its job.. to block incoming calls that dont produce a caller id.
J.C.
- Original Message -
From: Chris Albertson [EMAIL
Is Voicemail2 still able tohave
connectivitywith mysql.
I know the example was in voicemail.conf where you
specified dbhost,dbname,dbuser... etc.. , but dont
see any more in the latest CVS download. I am aware
of the asterisk-addons feature. I have installed the add-ons and am using Mysql
I am sure mostly anything is possible with
asterisk, but I would definitely recommend you buy a X100P if you want to
connect to PSTN., check it out at their website
at http://www.digium.com/index.php?menu=wildcard_x100p
for the price its worth all the troubleshooting you'll have to go
I have been using Mandrake 9.2 and it has been totally stable and haven't
had any problems with installations of asterisk. I stopped using RH9 because
of the upcoming
end of their support.
J.C.
- Original Message -
From: Tilghman Lesher [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent:
- Original Message -
From: Tilghman Lesher [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, January 22, 2004 9:09 PM
Subject: Re: [Asterisk-Users] Variable to play all gsm files in a directory?
On Thursday 22 January 2004 16:43, jaycard wrote:
Is there a variable that can be
[EMAIL PROTECTED]
Title: Leterhead
You have to input your info in your sip.conf -- its in your examples
Checkout this site for examples www.fnords.org/~eric/asterisk
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David J Carter
Sent: Saturday, November 01,
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