similar to: Detect Dialtone

hi everybody, Is there SIP client which work with Asterisk and can be embedded in a HTML page ? Thanks Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031029/226a8b1b/attachment.htm

Hi, I see applications for signalling busy, congested, ringing, progress etc, which I understand can be provided either in or out of band. But all I want to do is generate a dialtone. This obviously can only be done in band. There is code for generating the tones when you have a physical line, like the alsa channel, or a zap channel. But I'm just thinking of if they've selected an option

Is there a way to generate an old-fashioned dial tone with Asterisk? I'm thinking of one that sounds like: http://www.seg.co.uk/telecomm/dialtone.wav -- Phil Reynolds o ____ mail: phil at tinsleyviaduct.com |L_ \ / Web: http://www.tinsleyviaduct.com/phil/ (_)- \/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95

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

I need to select a line from the Zap group channel using the SIP Phone (not FXO and not FXS ports). ignorepat does not work? Also, what is the method to let the second dial tone has another tone frequency? Regards Bilal ---------------- No, ignorepat is for FXS ports (FXS ports use FXO signaling). Also, ignorepat does not apply to SIP phones, because SIP phones provide their own dialtone,

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

extensions.conf has ignorepat => 9 exten => _9X.,1,Dial(Zap/G2/${EXTEN:1}) The first user to try it asked if instead of keeping the same dialtone after pressing 9, if I could play a different dialtone. Can this be done? I'm running asterisk 1.0.0 in case that matters.

Hi, I'm having a little problem when trying to make a call from asterisk. I connect a SIP phone to asterisk, and in the asterisk box I have a TE110P card connected to a E1. When a SIP client makes a call through the E1, I received no dialtone in the SIP client. In the same manner, when somebody from the POTS network makes a call to a SIP client (through * and the E1) he doesn't receive the

Hello, I'm running * on a very basic configuration. I have a Wildcard TE405P with the first T1 connected to a PRI line and the remaining three to Adtran TA750 channel banks with FXS modules. I successfully configured everything to work with a couple of Swissvoice IP10S handsets (MGCP) and analog extensions connected to the channel banks. The problem I'm having is that when I pick up any

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

Hi everyone! 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. How do I implement this in extensions.conf...? Regards, Evert

Is there a way to reduce the volume of the dial tone on a zap channel? I don't want to reduce the audio volume on calls so txgain in zapata.conf will not work. I am having problems with asterisk not recognizing the first dialed digit from an analog phone about 8-15% of the time. Once the dialtone goes away, the digits are always recognized. Any other thoughts on how to solve this are also

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?

Hello, I created a new application for myself called Dialtone() by modifing res/res_indications.c file. It can be used as such: exten => s,4,Dialtone(30|${CALLERIDNUM}) exten => s,5,Playback(time-exceeded) exten => s,6,Goto(s|1) It will stutter if you have new voicemail and you have passed the mailbox number as I did above. It will stop dialtone the moment you press a key

Hi List, I cant seem to find the setting that changes this! You put a person on hold, they are on hold like normal, but after a few seconds the phone will then start having dialtone coming from the speakerphone, really weird!! Anyone know how to fix this? I see where it could be nice, but in this case, we just want them on hold is all, no dialtone! Any help would be great! Thanks! Ron

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>

I just set up a Polycom 500 on *. Every few calls I make, the call connects and the receiving party can hear me (thru Broadvoice), but I still get ringing on my end, as if they never picked up. * logs look just fine. Does any one have any suggestions? Thanks. ________________________________ Adam S. Robins Executive Vice President & CIO PHARMACENTRA, LLP 5901B Peachtree Dunwoody

So I have an X100P, and a TDM10B both working (at least I think they are). The drivers have been loaded and ztcfg -vv shows no errors in the configuration of two channels. When I run asterisk -vvvc and pick up my phone (plugged into TDM10B), I don't gear a dialtone. in phone.conf, I have [interfaces] mode=dialtone format=slinear ... Shouldn't that produce a dialtone when I pick up the

> 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

Hi, Is it possible with asterisk to deliver a dialtone to a software phone, such as kphone? I'm able to dial, but the silence seems to confuse my users :) thanks, lane