similar to: *@Home .6 adding a outside number to a group

Actually, I love my install of AAH 0.6. When something is not available in AMP I just dive into the configs and correct it. Most of the little things ARE available in AMP though so those times are few... W -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of David Sent: Monday, March 21, 2005 9:03 PM To:

Ok, Here is what I got working: A call comes in from a Zap line. 5 SIP extension ring if nobody picks up, the call is transfered to a cell phone number. That works. I not want to add a playback of a file ("Please waite while you are being transfered") before transfering the call to the cell phone. How can I do this? Andre

I'm trying to see if this is even possible. When you dial ext 2000 I want it to ring my sip phone for 20 sec then call my cell and let it ring for 10 sec if I do not pick up the call on my cell I would like it to go back to * and leave a voice message for me. Here is what I have so far in my extensions.conf Everything works except the call will not go back to * after the 10 sec rule has

Ok, I thought the point of asterisk@home was that it automatically detected the X100P board and configured it correctly. Is this incorrect? You still need to modify /etc/zaptel files? And not just using the AMP configurator. There is no mention of this on the Asterisk@home webpage. Can anyone who has actually used ast6erisk@home confirm this one way or the other? Thanks, Dean

Is there a configuration difference for clone X100P cards versus "compatible"? I have a similar problem to what David Shaw posted earlier today. 0.5 installed OK, but mine just with one X100P clone. Default config files, edited zapata.conf per the FAQs so it includes the line channel => 1 without the semicolon. Any outgoing call attempt returns "all circuits are busy"

Out of the blue extension 100 will ring once. This will happen 3-4 times a day. I have checked the logs and no incoming calls. I have extension 100 and 101 going to a SPA-2000 ATA. Any ideas on this?? Thanks, David

Hello I saw Ver1.2 is out. Whats new? Thanks for the hard work, David

Folks, First off, this is messy, and I hope someone will be kind enough to help me clean this up (the part added to extensions_additional.conf). You've been warned! For those of your using AMP or A@H, there has been a lot of talk about how to route incoming calls to different places based on which trunk is ringing. The standard answer is that you can only do this by using DIDs,

I've installed a TDM400. Having a go with AMP. I would like incoming calls to be put throuhg to an extension (at my desk) and a mobile (cell phone) at the same time. Whichever picks up, gets the call.. This should be possible with AMP (call groups, 200|201|0*0408xxxxxx), but it didn't work, so I have created a custom-incoming in extensions-custom.conf What is happening is, The extension

I have a client who is looking at replacing their PBX, and I'm interested in putting together an Asterisk solution for them. One feature that would really, really get their attention is if I could do the "Vonage" thing, where if a PSTN caller dials a direct extension (coming in over PRI) both the user's deskset _and_ an external number (their cell phone) would ring, with

I have a very strange problem I'm hoping someone has encountered already. I've scoured the internet for an answer to this one. My phone company provides no disconnect supervision. Hence I'm forced to use the busydetect feature. I have a TDM400 with two FXO ports. If I call from an internal extension to a landline and then hangup the landline Asterisk detects the busy signal

Hello All, I installed Asterisk@Home .5 last night. I was able to configure some extensions for the house and they work fine. I just can't make inbound and/or outbound calls. The Flash Operator Panel shows four external icons and my new extensions. I have four X100P and two Broadvoice sip accounts. Thanks, David -- David Shaw <asterisk@ke6upi.com>

Two problems, possibly related: Here's the configuration...my Asterisk box has a TDM844B in it; port 1-4 are FXS, 5-8 FXOs. Here are the config files: /etc/zaptel.conf: -------- fxoks=1 fxoks=2 fxoks=3 fxoks=4 fxsks=5 fxsks=6 fxsks=7 fxsks=8 loadzone = us defaultzone = us -------- /etc/asterisk/zapata.conf: -------- [channels] language=en usecallerid=yes hidecallerid=no

Dear R-helpers I have a problem related to the use of NLME I think is simply a matter of getting the nlme coding correct, but i cannot get my brain around it I am analysing some 24 growth curves of some cells , and i wanted to say that there are significant differences between the curves in two parameters that describe the pattern of growth. these parameters are from a logistic (r & k)

All: (This time with a subject line!) I am looking to move cell phone providers. I have acquired the new cell phone and LOVE my new number but want to keep the old number as well. The new provider only will allow me to use one number or the other. They will port the old number if I want, but will keep my new number if I ask them to port the old one. Where can I go to get the old number

Hey All, I have an Asterisk box connected to a Nortel Option 11C via a T1. In the Asterisk box we have a Sangoma A101C and in the Nortel we have a TMDI card. The Nortel is also hooked to the PSTN via a T1 on a different NTAK09 PRI card. I've included the Zapata.conf and zaptel.conf files below. Our issue is that when a call is sent over the tie line between the two systems, the audio on the

I'm still trying to learn some parts of Asterisk, so sorry in advance for the dumb question! How do I set up an extension to dial out to the PSTN through my ZAP interfaces? I want the ability to have a ring group that will ring all of the phones in an office and then ring cell phones if nobody answers. I'm sure this is simple to do but I'm at a loss. I have tried the following

Hi, I'm having a problem with Verizon Wireless, I'm hoping someone here knows the right way to phrase the trouble report so it gets to someone at Verizon who can solve the problem. We have DIDs that simultaneously ring on voip lines, and Cell numbers. Verizon voicemail is turned off. Every thing works the way it's supposed to, UNLESS one of the cellphones is turned off, or in a

We need incoming calls to simultaneously ring SIP phones, and a cell phone which is called via a SIP or IAX trunk. When the cell phone answers we'd like a brief prompt played (e.g. "press # to accept call") and if # is pressed connect the incoming call to the cell phone. ZAP trunks have some of this functionality with the call confirmation option, but we must use SIP or IAX trunks.

Hello everyone, Having recently gotten Broadvoice inbound DTMF to work (thanks Greg)...I am now running into a frustrating problem...when a call comes in to the BV number via a cell phone (tested with 3 different cell phones; albeit all on T-Mobile) the beginning of the IVR welcome audio is cut off. A call placed via a landline phone over the PSTN to the BV number does not exhibit the problem.