search for: dialtone

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 st...

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 done (immediate=no) and still it opens the zap trunk even when theres no dialtone and shows that "zap/3 answered"...

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? Thanks. Karlos.

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

...hardware. In a hope of getting better 'chan_skinny' support (to attempt using a Cisco 7920 IP phone) I built asterisk 1.2.40 on this box. Initial tests verify that our previous dialplan is working (iax2 trunks, register sip phones, registering withour SER box, etc. all work but there is no dialtone on the ixj fxs port. Here is what I tried: (in each case asterisk invoked with '-vvv -C name_of_config_file) 1. temporary fresh install of 1.2.40 using sample configs; in dialplan (uncommented extension ext. 1265); in 'phone.conf' uncommented the following: [interfaces]...

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.

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 corr...

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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists...

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 dialtone to the user and wait him dialing? Thank you, S...

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 that allows them to dial a normal number, to also provide a normal dialtone. Shoul...

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 phone? _________________________________________________________________ Help STOP SPAM with the new MSN 8 and get 2 months FREE* http://join.msn.com/?page=features/ju...

...S 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, not a dialtone provided by Asterisk. Al lists wrote: > Correction, on FXO port not FXS, > second, read his email first: > "Also, how it will be possible to assign an dedicated > line (connected to FXO) to an > button on the Polycom IP Phone or Broadtel IP Phone, > so if use...

...sly 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 SPA3000 PSTN Line. You can call, navigate de IVR, log in into our app, and then when you go to DISA, and DISA plays the dialtone... whatever you dial is not recognized... This was REALLY odd... so I made a network capture with Ethereal, and... the SPA actually STOPS sending the RTP Events after the second dialtone... To verify this, I created an IVR which played the dialtone, and verified that it was tr...

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 apropiate tone of call progress. Does anyone has some ideas about this? Ing. Marco Antonio Castillo Chief Design Engineer Van Der Kaaden IT Cons...

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

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

In AU we have a number of different dialtones defined for various purposes. >From indications.conf: au <ringcadance> 400,200,400,2000 au dial 413+438 au busy 425/375,0/375 au ring 413+438/400,0/200,413+438/400,0/2000 au congestion 425/375,0/375,420/375,0/375 au ca...

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 ----...

Hi there, i have a Problem with dialtone and a TE401P Card. I swear I surfed the wiki, the mailing list and google for 4 hours and did not find the solution, can you help me ? In Germany I have an E1-Line and an Alcatel 4200 PRO PBX. Without using asterisk I dial the "0" on an Alcatel Phone and have the local exchange dialtone...

...l(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel) exten => _91866NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel) exten => _91800NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel) Dialing 91800nxxxxxx does attempt a call to 1800nxxxxxx via iax to iaxtel, but I do not get dialtone after the 9. I presume I have to get * to send dialtone via rtp (or is there some sip packet that will tell the ata to continue dialtone)? How is that done to a sip extension? The calls to iaxtel don't seem to be working. I'm also not able to get Dial(IAX2/guest@misery.digium.com/s@defa...