search for: hwan

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

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! -------------------------------------- ;Asterisk CLI as I placed a call from cell into the system. Playing

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

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

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)

...e been wanting something similar. I paid some money for a busy detect routine from newman telecom, but it is not yet done. We'll see what happens. Greg -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Min Hwan Chang Sent: Tuesday, August 09, 2005 6:57 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Incoming call #2 sent to VM immediately whenalready 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 V...

...ess the line in 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 kno...

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

Hi, How do I rename the column labels in the table? For Instance, if I have a table like this, and I want to have the column labels changed from "A1 A2 A3 A4 A5" to "Mike Kate Michelle Paul Young" A1 A2 A3 A4 A5 1 33 44 55 66 77 2 3 4 5 6 7 7 8 9 and my text file location is: ""/Users/MAC/Desktop/data.txt" When I type in

...ne) 8. RE: Epia 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 Hw...

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!

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.