Displaying 7 results from an estimated 7 matches for "phpmechanic".
2005 Feb 13
6
Who makes these phones?
Message: 1
Date: Mon, 14 Feb 2005 09:53:36 +1100
From: "PHP Mechanic" <oliver.bode@phpmechanic.com>
Subject: [Asterisk-Users] Who makes these phones?
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID: <08d401c5121e$dbea4750$0200a8c0@oliver>
Content-Type: text/plain; format=flowed; charset="iso-8859-1"...
2004 Sep 10
1
Call Parking Problem
Hi,
I'm unable to pick up parked calls after they are transfered.
I get the "transfer" message when I press # and then I'm told "701" The
extension I'm dialing goes to the on hold music. I'm disconnected, I hang
up, dial "701" and I see this message on the console "Everyone is
busy/congested at this time"
I just have the default
2005 Feb 22
0
How do I do this ?
...s@lists.digium.com] On Behalf Of C F
Sent: Wednesday, 23 February 2005 12:37 AM
To: PHP Mechanic; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] How do I do this ?
FOP
http://www.asternic.org/
On Tue, 22 Feb 2005 22:40:32 +1100, PHP Mechanic <oliver.bode@phpmechanic.com> wrote:
> I wish to initate calls from a web interface, by clicking on a link
> and then connecting to the automatic outgoing call by picking up an analogue phone.
>
> I've got one fxs and one fxo and I wish to automate the call using a
> call file (which I can do now)....
2005 Aug 02
2
port forwarding ip to ip sip calls
Hi,
I've got two pa1688 phones that I want to set up to communicate between
branch offices without a gatekeeper. Both phones will be behind a
firewall and I want to use port forwarding so the phones can communicate.
I tested the phones behind a firewall on the same network segment and
there were no problems at all using sip. However, I then moved the
phones into situ and port forwarded
2005 Mar 18
15
Meetme2 compilation problem
Hi All,
I am trying to compile meetme2 in my asterisk box and getting the
following compilaton error. Please help me to sort it out.
cc -fPIC -c -o app_dial.o app_dial.c
In file included from app_dial.c:14:
/usr/include/asterisk/lock.h: In function `ast_mutex_init':
/usr/include/asterisk/lock.h:317: `PTHREAD_MUTEX_RECURSIVE' undeclared
(first use in this function)
2005 Oct 03
1
no audio on fxo line
Hi,
I got back from two weeks away and appear to have lost audio on my
tdm411 fxo. Everything was working properly when I left. I checked the
logs, config files and can't see any problems, the zap channels and
modules are all loaded properly, asterisk starts without probs and
everything looks sweet on the colsole with -vvvvvvvvc when I make calls,
but I just don't hear a dialtone or
2004 Aug 26
0
Can't make zaptel on red hat 9
Hello,
I've followed the instructions here:
http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation, but I get the
following when trying to make zaptel:
I've added the relevant symlinks:
[root@asterisk src]# ls -ld /usr/src/linux*
lrwxrwxrwx 1 root root 14 Aug 26 22:50 /usr/src/linux ->
linux-2.4.20-8
lrwxrwxrwx 1 root root 14 Aug 27 2004