Displaying 6 results from an estimated 6 matches for "carnt".
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cant
2003 Jul 11
1
FreeBSD /etc/passwd errors
...unds = oe6-fetch-no-newmail outlook-idle
auth = default
auth_userdb = passwd
auth_passdb = passwd
auth_user = dovecot-auth
auth_username_chars =
If its somthing simple please point me in the right direction as I feel increadbly dumb because im sure its something to do with the passwd file but just carnt get pass the last hurdle!
Many thanks in advanced.
James.
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2004 Sep 19
1
Dial 0 to outbound
Hi Folks.
I see that can put 0 to call out using a x101p (zaptel) or even a pstn service.
Thats great, but when press the 0 i just dial then the numbers to call out.
There is any way after hit 0 (ear) the line sound ??
I know it's just a style way put some users, really like it !!
So after hit 0 to call for example a pstn the user will ear the line sound
to dial out.
I read lot's of
2003 Dec 21
4
First version of the ActiveX version of DIAX (0.1.0) available for download
Hi all,
A first basic version of DIAX as an ActiveX can be downloaded from:
http://www.laser.com/dante/diax/activediax.zip
There is only one small file (diax.ocx) and a readme.txt with the usage
instructions.
For the moment you can only place authenticated (or not) calls and there is
no feedback (ring, messages, etc)
Put this simple thing on your web page and you will be able to dial from any
2004 Jan 20
1
PSTN Gateway
...se: 355 - Release Date: 1/17/2004
>
> ---
> Outgoing mail is certified Virus Free.
> Checked by AVG anti-virus system (http://www.grisoft.com).
> Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004
>
>
> --__--__--
>
> Message: 2
> From: Carlos Arnt <carnt@intellissence.com>
> To: <asterisk-users@lists.digium.com>
> Date: Tue, 20 Jan 2004 02:39:09 -0200
> Subject: [Asterisk-Users] X101P CallOut Big Problem.
> Reply-To: asterisk-users@lists.digium.com
>
> <html><head><meta name=3D"Generator" content=...
2003 Oct 29
3
VMAIL.cgi
2004 Sep 17
1
Canreinvite=???
Hi, everyone !
Looking at this explanation :
"When SIP initiates the call, the INVITE message contains the information
on where to send the media streams. Asterisk uses itself as the end-points
of media streams when setting up the call. Once the call has been accepted,
Asterisk sends another (re)INVITE message to the clients with the
information necessary to have the two clients send the