Displaying 10 results from an estimated 10 matches for "minchang".
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minchan
2006 Apr 15
3
FreePBX in Production systems?
Is anyone using FreePBX in production level systems because I'm just
wondering if its stable enough to use. Currently I'm editing my own *.conf
scripts but it sure would be nice if there were some sort of web interface
for other people to use. The only thing holding me back is the stability of
the FreePBX package... Any comments on this? Thanks in advance.
Regards,
Min Chang
2006 Mar 03
2
Does an entry in AstDB stay after reboot?
I set up a call forwarding script in extensions.conf which uses the
AstDB but I'm wondering if I reboot the server, will the entry in
AstDB still reside?
What the script does is when a call comes in, it check to see if there
is a null value or a call forward number. If null, it will call the
local office connections. If there is a number, it calls that. Now I
just need to know if I reboot
2006 Apr 14
0
Ztmonitor shows RX is always on. FIXED.
...India is pretty loud b/c the RX
bounces off the charts, changed it to (-4) and all my problems are solved.
Regards,
Min Chang
On 4/14/06, Kyle Sexton <ks@mocker.org> wrote:
>
> Have you tried putting a Hangup in your extensions.conf?
>
>
>
> On 4/13/06, Min Hwan Chang <minchang@gmail.com > wrote:
>
> > Details:
> Asterisk 1.0.9
> Zaptel 1.0
> Dell P3 1ghz with X100P Clone
> Location: India
>
> This is an interesting issue where when I open up ZTMonitor, it shows the
> RX as being on. It seems that Zaptel doesn't know to hang up the li...
2006 Apr 13
1
Ztmonitor shows RX is always on.
Details:
Asterisk 1.0.9
Zaptel 1.0
Dell P3 1ghz with X100P Clone
Location: India
This is an interesting issue where when I open up ZTMonitor, it shows the RX
as being on. It seems that Zaptel doesn't know to hang up the line so after
a couple of hours when the telecom cuts the line, everythign stops working.
Things I've tried include playing with the zaptel.conf, trying zaptel
v1.2(with
2005 Jul 05
1
Help with Cisco 7905G corrupted image!!
...pia C3 Linux (Rob Thomas)
----------------------------------------------------------------------
Message: 1
Date: Tue, 05 Jul 2005 19:25:06 -0500
From: Eric Wieling aka ManxPower <eric@fnords.org>
Subject: Re: [Asterisk-Users] Best BootRom & SIP Code for Poly600?
To: Min Hwan Chang <minchang@gmail.com>, Asterisk Users Mailing List
-
Non-Commercial Discussion <asterisk-users@lists.digium.com>
Message-ID: <42CB24E2.7060802@fnords.org>
Content-Type: text/plain; charset=us-ascii; format=flowed
1.5.1/1.5.2 also support disabling call waiting too,
Min Hwan Chang wrote:
>...
2005 Jun 09
0
Getting a Quintum AS200 to connect with Asterisk using SIP?
Does anyone have any instructions on how to connect a Quintum AS200 to
Asterisk as a SIP phone?
I currently have the AS200 sitting in a remote office and would like
to use it as a SIP phone. It would register with Asterisk.
Overall, I've had problems getting this work but I'm hoping that
someone on this list might have some experience with this!
2005 Jul 22
1
Caller logging in to call out IAX line?
Hm, I'm wondering if its possible for someone to call in the POTS
line, dial an extension, then be able to dial a number of their
choosing out the IAX line?
So let's say I'm here in california and I dial into the office. Dial
8888 which gets me a message saying please enter the number you'dl ike
to call. At which point I dial 7983487 to dial someone in Austria over
IAX. Is this
2005 Aug 09
1
Incoming call #2 sent to VM immediately when already on phone with incoming.
I'm having this problem where if the phone is ringing from
IncomingCall #1, IC#2 will be immediately sent to VM. Is there
somethign wrong with my dial plan? I currently have 4 incoming lines
going into a TDM400 with the group set to g0.
Could it be that the way I've set this up, if any of the phones are
busy, it goes immediately to VM?
exten => s,1,Answer()
exten => s,2,Wait(1)
2009 Jan 27
0
SPA-3102 in India - Problem dialing out PTSN
Good morning,
I've been having some problems getting the SPA-3102 working properly in
India. Specific problem is that calls from the Asterisk server out the FXS
port is failing. When trying to make calls, I'm getting this message:
[Jan 26 23:00:31] NOTICE[2136]: chan_sip.c:13774 handle_request_invite: Call
from '' to extension '66200' rejected because extension not found.
2005 Jul 25
2
DISA disconnects
DISA is currently disconnecting when I dial 8888 to access DISA.
Below is my extensions.conf file from A@H and some lines which shows
the disconnect. Should DISA be loaded as a module in modules.conf?
When I do a 'show applications' i see that DISA is there. Help!
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;Asterisk CLI as I placed a call from cell into the system.
Playing