similar to: Dialtone for Software phone?

Displaying 20 results from an estimated 9000 matches similar to: "Dialtone for Software phone?"

2005 May 18
3
connecting a sipura sip device to asterisk before dialing any digits
I would like the ability for a sipura sip device to instantly connect to an asterisk server as soon as the sipura sip device goes offhook and before any digits are pressed. This way asterisk can provide the dialtone and the dialplan. This also allows me to play a greeting to the phone before giving them a dialtone. Is there any way to do this, like possibly having the sipura device dial a
2005 Jul 20
1
Play Dialtone - get digits
I'd like to write a snippet of dialtone that plays dialtone and collects a specific number of digits into a variable. Sort of like READ but with a generated dialtone. Naturally, I want the dialtone to stop playing after the first digit. I can't find this anywhere. Only thing I can think of is a no-password DISA. Is this the correct method? Is there a better one? </edg>
2006 Mar 22
3
Remote dialtone
Hi, I have two asterisks connected via IAX2 trunk. The first * use dial prefix 2XX, the second one 3XX. Calls routing works OK. But I don't know how to get dialtone of remote asterisk pbx. I'd like to get dialtone of asterisk #2 after dialing 3 and dialtone of asterisk #1 after dialing 2. I know something about DISA but I'm not sure if it is a right way. Can you give me advice?
2005 Mar 08
4
Nortel ATA not passing dtmf tones to fxo
I am trying to integrate a Nortel Norstar system with an Asterisk service using a TDM04B card (4 fxo). So far everyting works from Asterisk to Nortel. The problem is when someone dials from the Nortel PBX to the Asterisk server. Asterisk answers the call and provides a dialtone (with DISA) but appartently the DTMF tones are not passed to asterisk and the call cannot proceed. This only
2005 Jan 13
2
How to present a dialtone to a dial-in user?
Hello, Here's what I'd like to do: call my Asterisk box from a phone, hangup after a few rings, then Asterisk calls me back and presents a dialtone, than I can dial any valid number in the context the call originated. I've done it with CAPI (thanks to the script on http://www.junghanns.net/asterisk/page14.html), I'd like to do it with H323. Problem is, how to present a
2005 Aug 12
3
TDM400P FXO channel hookstate always "Offhook" & outbound digits sent before provider dialtone
I have an Asterisk@Home 1.3 server (Asterisk 1.0.9) and recently added a TDM400P with (1) FXO card on port 4. Inbound calls are always successful but outbound calls fail 75% of the time with intercept messages from my dial tone provider that include "we're sorry, your call did not go through", and "we're sorry, when placing a local call it is now necessary to dial an area
2003 Apr 29
3
Can you invoke an app before dialtone?
say I needed to send a broadcast message that I wanted every user to hear when the pick up thier phone? can I "Play,message" on a line just before they get dialtone? or maybe after they dial before ring? how about a "ringdown" to a voicemail box and on end return them to thier line for the dialout? can * do ringdowns? when a user picks up an extension it automagically
2006 Mar 13
2
DISA & SPA3000 issues
Hi, These days I run into something quite odd. I have an A@H that was modified to meet our requirements. We have a completely funtional DISA which we use pretty much all the time. I works flawlessly with incomming SIP calls from several providers, IAX calls from FWD and with ZAP. Recently we came out with a situation where it doesn't work... with a
2004 Jul 05
2
Problem with BRI_STUF / direct connected ISDN-Phone
Deutsche ?bersetzung folgt / German version following ===================================================== Hello, i have Asterisk running with 2 ISDN-Cards. One AVM Fritz for connection to german ISDN and one HFC-compatible-Card (NT mode) for connection to ISDN-Phone (later: ISDN-PBX). Here is my actual installation: ISDN -> Fritz - ASTERISK ? HFC-NT <- ISDN-Telephone If i pick up my
2004 Sep 17
1
AW: dial '0' for outside line and get a dialtone...
> I'd like to create the following: a user picks up the phone > (gets a dial tone), dials '0' for an 'outside' line, gets a > second (different?) dialtone, and is able to enter an > external phone number. Klaus-Peter Junghanns has something like this on his page: http://83.137.99.170/jn/relaunch/asterisk/page19.html It didn't work for me correctly so I
2005 Jan 04
1
dialplan question - how to dial an * extension to get an outbound dialtone?
First, please forgive me if this is a total newbie question, I've only just begun to scratch the surface of asterisk. I currently have a dialplan set up to let me dial a specific extension, authenticate the user, then have * dial a hard coded/programmed overseas number. What I would like to do is set up my dialplan to have an extension that offers up an outbound dialtone allowing the
2005 Oct 17
1
How can I get a dialtone calling from outside...
Hi all, How can I configure, in extension.conf, to call and extension and have a dialtone so I can compose a number to dialout? Basically, I want to be able, when I am out of the office, to call in my asterisk box and then dialout from it. Regards, Francois
2005 Feb 27
1
DISA and a long delay; ideas?
Hi, I have just setup a DISA setup whereby people can dial in, authenticate, are given a dialtone and can then call out. Everything works however there is a 10 second delay after the user enters the number and presses #, until the system does anything. Here is the relevant section from my extensions.conf: -------------- next part -------------- An HTML attachment was scrubbed... URL:
2003 Jun 15
3
Voicemail and DISA fixes
I've commited changes to Voicemail2: * Handle properly when being left a message while checking VM -- this should fix the "saving to your inbox" issue too, at least in principle. And to DISA: * Properly handle extensions with multiple matches and "dots" Please let me know on or off list about any feedback you have regarding these changes. Mark
2005 Feb 20
8
Simulated dialtone like in other PBX
Guys.. Im new to asterisk but is it possible to simulate a dialtone for example, in other PBX when you pick up the phone you can hear a certain dialup, which is the PBX dialtone, and when you hit 9, you can hear the PSTN dialtone, is this possible? __________________________________________________________________ Anton Krall
2004 Apr 06
1
SIP phone registering problem
I am clearly doing something ridiculously wrong. Running Asterisk 0.7.2 on FreeBSD 5.1, I have SIP soft phones which are unable to register. They keep trying and then time out. With the sip debug on in Asterisk nothing is logged. Here is the trace from one of the phones (kphone): (192.168.100.13 is kphone, 192.168.100.3 is Asterisk) sipclient: sending: 21:47:45.454
2005 May 10
2
DISA
We are using DISA with local SIP users. The user enters in a 2 digit code then they get a dialtone and the phone dials out. The problem is that the calls waits 10 seconds after the outgoing number is dialed, no matter what I put for the timeout values. Anyone else using DISA that has run into this? exten => _2X,1,Answer exten => _2X,2,DigitTimeout(2) exten =>
2005 Aug 03
4
Transfer to outside line.
Finally got everything up and run with the help of Manny Wise last night. So I am setting up my digital assistant and getting down to the task I need this box to perform the most. I need to have a custom app that I can call that will take me pressing 2 at the menu and have it transfer the call to a offsite phone number utilizing my Zap Trunk. I'm sure someone has done this already. Anyone want
2006 Jun 05
2
DTMF and DISA
Hi Folks, I'm trying to test out Asterisk overall. I'm having some problems with DTMF. Currently I'm playing with DISA, but I'm worried this will happen when I get to implementing AAs etc. I have a free SIP trunk from IPKall that I'm trying to make work. I'm able to receive calls, and I've now setup and extension with DISA and a password. I connect ok from the
2004 Jan 06
3
Policies - deny some nubers
Hi, I had asterisk installed, ISDN-adapter, some x-lite software-phones and I can call betweens the softphone- and 'normal' phones during the ISDN-card. 2 questions now 1) Is it posible to create policies, so that some SIP-users can dial ALL numbers, and some SIP-users not are allowed to dial eg. 900xxx-numbers, 30xxxxxx (mobilphones), 40xxxx(long distance) and if possible on time