similar to: No dialtone in a E1

Hi. I guess the world is full of these, one more won't hurt: "The ogg-cdparanoia script is a wrapper of the programs cdparanoia (http://www.xiph.org/paranoia/), sox (http://sox.sourceforge.net/) and oggenc (http://www.vorbis.com/). The process is simple: for each track, cdparanoia is called to extract the audio to a temporary wav file. Sox is used on it to see if it is normalized.

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?

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

In the next couple of weeks we will be starting the beta phase of our Guatemala POP. If you could wait, welcome. LTenorio -----Original Message----- From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Phil Astin Sent: Sunday, January 16, 2005 6:23 PM To: asterisk-biz@lists.digium.com; asterisk-users@lists.digium.com Subject: [Asterisk-biz]

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

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'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>

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

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.

My name is Stuardo, from Guatemala and i have this problem: We want to stream a local radio, but the bandwidth here in Guatemala is no so cheam ass in USA, so we have a server in USA. What we want to do is this: to stream the local radio from Guatemala to USA, this means that the local machine will only have 1 client. Then we want that the server in USA "rebounds" what it recives

i need to know something in the zaptel configuration as it seems i can configure detecting the busy tone and hangup after number of busy tone counts, that was great but the problem is sometimes the pstn line has no dialtone and when i try to make call it continue dialing while not having a dialtone! while it should say "all lines are busy/congested" how can i configure that?? i already

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

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

Is it possible after dialling a trunk prefix to wait for dialtone (or wait 2 seconds) before continuing to dial the outgoing number ? I need to "hunt" for one of several lines, and when one successfully provides dialtone, to continue dialling.. A simple NO will do, but if there's a workaround, I have not found it. -------------- next part -------------- An HTML attachment was

The Vonage service is offered with a SIP Cisco ATA device for connection to an analog phone. Is it possible to connect the Vonage service directly to the Asterisk PBX bypassing the ATA and FXO card? Are there other services that offer this capability or something similar to IP dialtone? Thanks Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, Sorry, I know I read this somewhere but now I can't find it when I need it. I'd like to force a call to go out one line if we dial '9' first and then the number. Same for '8' only I will force it out a different line. There is a parameter or a method to allow the dialtone to come back after pressing the first 9... but I can't remember how to do it. Anyone know?

How can I get external (telecom local exchange) dialtone on HFC ISDN BRI with bristuff/zaphfc driver? with capi, voip-info say that it should be something like: Dial(CAPI/MSN:b) But with zaphfc, if I try: Dial(ZAP/1/), I just get NOANSWER.

Hi all, I am having a problem on to different boxes and to different channel banks. I can't get dial tone out of either one. I can still send and receive calls but no DT. This is the error that I get: Jun 8 15:23:54 WARNING[5021]: chan_zap.c:6283 handle_init_event: Unable to play dialtone on channel 95 asterisk-1.2.9.1 zaptel-1.2.6 zapata.conf: signalling=fxo_ls context=gene

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,