similar to: making * more like a normal pbx (cisco ata-186)

When a user dials 999 he is always asked for the mailbox and has to enter his mailbox number and password. As I understand this shouldn't happen because the CALLERIDNUM is passed over to VoicemailMain. It's annoying to have to enter the number everytime ... The voice mail configuration is read from MySQL. We are using the CVS version from a few days ago. Extract from extensions.conf:

Hi All trying to compile asterisk under linux kernel 2.6.6. Currently under zaptel get the following error make linux26 Link /usr/src/linux-2.6 to your kernel sources first! make: *** [linux26] Error 1 as going from the readme. is 2.6 not compatiable with asterisk and should I go back to 2.4.26. Also has anyone got the sipura 3000 working with asterisk, both fxo and the fxs ports on the

Hello, I have modified the VoiceMailMain application to satisfy the request of the "local users", i.e., my wife. She lost patience with too many options (we have one mailbox, so we don't need to forward messages, or reply to messages, or file them in 6 different folders...) So the modified app says "Message 1", reads the message, "Message 2", reads...

Hello, Over the past several weeks, we have been having an intermittant problem with our deployment of a TDM400P card (4 fxo). ?We have tried many things, and the problem still re-occurs. The Problem: Occasionally (every 48 hours), the TDM400p card will stop answering incoming calls on ALL fxo ports. ?Attempts to send outbound calls on any Zap channel will result in hearing a loud

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Robert Withrow > Sent: Tuesday, June 15, 2004 12:32 PM > To: Asterisk-users > Subject: RE: [Asterisk-Users] making * more like a normal pbx (ciscoata- > 186) > > On Mon, 2004-06-14 at 19:34, Reid A. Forrest wrote: > > I've

once u press 9 is there a way to make it so it restores dial tone, like most pbx's do? so dial tone , 9, dialtone, then ur local num -- Gafachi.com - referal code hunter81 instant iax termination - 2 cents a minute Also they have a great referal program, tell them jacob, hunter81 sent you -------------- next part -------------- A non-text attachment was scrubbed... Name: not available

I have a brand-spanking new TDM31B (3 FXS, 1 FXO) and when I start wcfxs (the only module that recognizes the card) from Zaptel 0.9.1 I get: Zapata Telephony Interface Registered on major 196 Freshmaker version: 63 Freshmaker passed register test Module 0: Installed -- AUTO FXS Module 1: Installed -- AUTO FXS Module 2: Installed -- AUTO FXS

I've searched the lists but I didn't find anything exactly like this. I have two Grandstream BT101 phones connected to an Asterisk. Periodically, for reasons that I can't determine, one or the other (or both) of the BT101s decide(s) to go on permanent busy. Dialing that phone gives: -- Executing Macro("SIP/24567-7856", "dialphone|SIP/27654") in new stack

Hi: I'm attempting to run "The Neverhood" on FreeBSD/wine-2002.10.07 no windows config, which, according to the application DB, should work. When I run the setup95.exe program the Neverhood installer screen, and when I hit "INSTALL" I get: bash-2.05b$ wine d:setup95.exe fixme:cdrom:CDROM_GetIdeInterface not implemented for BSD fixme:reg:GetSystemInfo not yet supported on

I think I've configured everything to have stuttered dialtone on my analog phones (it works fine for my SIP phones). But I still don't have it:-( I'm using asterisk 0.9.0 and zaptel 0.9.1. In voicemail.conf: [local] 21111 => 987654,Robert Withrow,email@here.com In zapata.conf [channels] ... mailbox=21111@local channel => 1

Hi all, I tried to add a new user to a Samba share, I did all the usual: made a Unix account on the Unix server for the user which was the same as the user's window's UID. Put her name as a valid user for that directory in server:/etc/opt/samba/smb.conf, and stopped and started server:/sbin/init.d/samba server; nothing worked. Interesting observations: 1- After performing the steps

No, DON'T use mpg321, it will not work. Make sure you're using mpg123. See the wiki. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Rich Adamson > Sent: Friday, October 01, 2004 12:13 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re:

Hi people, I have a Cisco ATA 186 which understands only the SCCP protocoll, therefore I am a pure beginner and I hope that anybody of you could help me. How will I configure the ATA which has 2 analog ports? For any support I would kindly thank you Tamer

Thanks, Mark! Here's a summary of what one needs to do in order to get NAT working with Asterisk. Please note that I have a Cisco ATA-186, and your experience may be slightly different based on the equipment you're using. You'll need to have a CVS updated version of Asterisk as 2003-03-06 ~2:00 PM EST. NOTE: This currently works for outbound calling only, not inbound. In other

Does anyone have experience with ATA-186 Dialplans? If I was doing this in Asterisk I would use the following patterns: 911 _9NXXXXXX _91XXXXXXXXXX _NXXX _*XX I tried the following in my ATA-186, but it doesn't seem to work: 911S|9.r6S|91.r9S|^1.r2S|*..S Does any one have any ideas/suggestions? Thanks in advance, --Eric

Is it just me or do others have a problem with the ATA-186 de-registering? Every couple of hours, if I don't make use of the ATA connected line, I find that I have to unplug and let the ATA reboot. After that it is good to go for awhile, but eventually I have to repeat the process. My ATA sits behind a NATd firewall, any ideas what might cause the de-registration? Kim C. Callis

I use cisco ATA 186 ( Version: v2.16 ) with sip protocol and asterisk as pbx. I need feature called as 'three way calling' or 'transfer with consultation'. Registering,calling and 'blind transfer' work fine. Is this feature provided by sip clients or by asterisk itself ? What I have to configure in ATA and what keys I have to press on my phones ? Three way calling is

Hi All, I'm looking at setting up a Asterisk system, and hope to use ATA 186's with it. Im in Australia, and am getting mixed answers to if its the I1 or I2 i need, does anyone have any experience with using ATA 186's in Australia Also, can anyone recommend a good place to obtain these locally? Cheers, Steven

The details for the Australian cisco ATA186 are below: > -----Original Message----- > From: Tony Du [mailto:tony.du@action.com.au] > Sent: Tuesday, 8 July 2003 4:31 PM > To: 'Adam Goryachev' > Subject: RE: [Asterisk-Users] ATA 186 in Australia > > Hi Adam, > > I sold a Cisco ATA186 I1 2 port adaptor (Cisco code: > SW-SMH-UL-ATA-2P)to you on 16/10/02) >

I have been using the ATA-186 with good success (with exception of the fact that you have to recycle it from time to time). The one thing I haven't been able to do is to figure out how to make use of parking, transfers, etc. My cordless doesn't seem to pass DTMF, so I haven't had any success with parking. As for transfers, I can do a flash and get dial tone, but then what are my