search for: touchtones

> -----Original Message----- > From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com] > Is the channel physically being hung up before the * tone is heard? Good question. If it is, Asterisk doesn't detect it -- the PBX doesn't support Kewlstart-style disconnect notification. The sequence I hear on the extension, when I plug in an analog phone, is the click of the

Hello: I need to be able to reliably send out touchtone to any calling party who comes into my pbx. The standard things to help with this have been done as far as I know: 1. dtmfmode is rfc2833. 2. The phones themselves are set to rfc2833. 3. allow=ulaw 4. On internal calls between extensions, touchtone works fine. Also, I have reviewed sip.conf with my carriers. Now for the

The legacy PBX I'm working with does in-band disconnect notification -- it sends a * touchtone when the line is hung up. I've been trying to get this to work with Asterisk. I added a * extension to my menu context that plays "Goodbye" and hangs up. This works fine if I manually press *, but it never triggers when I hang up and the PBX sends it. I've plugged in an analog

> -----Original Message----- > From: Paul Rodan [mailto:asterisk@glitch.cc] > I'll ask a stupid question, how does a user hit an alpha > letter from his touchtone? > > I know that the Cisco 7960's support entering alpha letters, > and it could > potentially do it (maybe), but how does the average end user > enter an a b c or d from their touchtone phone?

> -----Original Message----- > From: David Brodbeck [mailto:DavidB@mail.interclean.com] > Okay, the problem appears to be that I'm tone deaf. ;) > > I finally thought to turn on debugging on the channel. The > PBX is sending > "D", not "*". The programmer of the previous voice mail system (whose > configuration I was cribbing from) seems to have

> -----Original Message----- > From: Gilad Ben-Yossef [mailto:gilad@codefidence.com] > I'm prbably stupid, but wont this do what you want? > > > exten => 1,1,Goto(bye,s,1) No, because I wanted to match on "D", not "1". Anyway, I figured it out. The extension was working, but Background() ignores the tones A through D by default. I didn't

> -----Original Message----- > From: Kevin P. Fleming [mailto:kpfleming@starnetworks.us] > David Brodbeck wrote: > > > Anyway, I figured it out. The extension was working, but > Background() > > ignores the tones A through D by default. I didn't realize > this because I > > wasn't waiting for message playback to finish. > > Please enter a

> -----Original Message----- > From: Chris Wade [mailto:clwade@sparco.com] > Brian Roy wrote: > > I think that my PBX does this too. Is there any way I can get the > > Zaptel drivers to disconnect on that tone too? I would love > to replace > > my existing voicemail with * but I can't get my PBX to signal a > > disconnect properly. I have to use

We are using Asterisk 1.0.7. We have this scenario: IAX2 user comes in to Asterisk, dials an extension, and transfers to a SIP user. The dial command is simple, looks like this: exten => 300,1,Dial(SIP/300) Extension 300 is a legacy PBX device operated by touchtones. The user (coming in over IAX2) is trying to drive this PBX using touchtones. But the trouble is, by the time the touchtones go out to the SIP extension, it's being sent as audio, not as RFC2833. An ethereal trace confirms this. DTMF is arriving to Asterisk (via IAX2) as OOB data, but no RFC283...

Hi, I'm having a problem calling international numbers with debian's Asterisk 1.0.7 w/ zaptel 1.0.9 in Moscow. Russia doesn't seem to have touchtone dialing, so pulsedial is enabled on my TDM400P interface. Local numbers work fine, but when it comes to long distance or international, I'm lost. The prefix for these should be 8 (wait for dialtone) 10 (country code) (city code)

I've been looking around for a small external sip fxo gateway, sending emails to possible vendors, etc, and can not seem to come up with anything that fits. Suggestions anyone? (No channel bank & T1 card suggestions, please. I've also just completed an eval of the Mediatrix 1204 which does not support the requirements.) The market between two fxo pstn lines (pair of x100p's) and

Hi, Some users have been reporting a peculiar problem. The are having an issue when they dial out to some multi-level IVRs where you make 2 or 3 touchtone choices and then are connected to a live operator. When the live operator connects the operator cannot hear them or sometimes it results in dead air. With the one-way audio issue, is it possible that something has locked the channel into some

...ith the echocancel and the echotraining settings, and I still can not get the echo to go away. Also, when tring to call out, i always seem to get a message that calls can not be compleated as dialed. However, when I switch to pulsdial=yes, then, all the calls start going out ok, its almost like the touchtones are not being recognized properly. Its hit and miss. I have even tried adjusting the rxgain and txgain. <br />I have a asterisk system at home, and i dont have this echo problem...<br /><br />I really need to get this system up and running, and, im at a loss as to what to try next...

...oint where both my hardphones (ATA's) and my softphones (IAXcomm, or JackenIAX) work perfectly with comedian mail. To me this means that asterisk is properly getting the RFC2833 events from the user agents. BUT, if I try to dial out the FXO, none of my phones (hard or soft) produce working touchtones for a PSTN based voicemail system. Even stranger to me, is the fact that from the phone connected to the FXS on the wellgate I can hear tones(listening on a called line), but they sound kind "rough" at the edges. From the AG168V I hear no tones, but what seems to be "blown out...

I have a client who is using a Shoretel PBX. This PBX apparently does not send DTMF information OOB, but instead sends this inband via the B-channel. This is traversing an Asterisk box via a PRI. The user calls the IVR (1-800-CALL-DHL), receives audio, but is not able to present DTMF to engage the IVR. With some light research it appears that the DSP is not activating until the call is

...th teliax is that im not paying 2 cents per call essentially to deliver a 45 second message. For those that don't know, teliax offers termination at 2 cents per minute 60/1 Bottom line teliax service with voxee pricing would be great if anyone has tested any providers that properly relay touchtones. Mark Adams <http://www.vistaprint.com/vp/gateway.aspx?S=5176697856> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060128/0d0acf48/attachment.htm

Hi. Ever since I bought my Digium 400P with 1 FXS and 1FXO module, once in a while I hear what sounds like a touchtone in my ear on a phone hooked up to the FXS module. This was not heard by the other side, and although it was annoying, it was not too much of a problem till I was using the asterisk 1.4 (rev 46317) and the beta of zaptel 1.4 (rev 1536). Doing this, the toutchtone noises once

...act, I call this system "IKR" (Interactive Key Response), and maybe every system should answer the call with first saying "IKR". Then callers would immediately know when our skills on the common UI would work, without waiting to learn, or mistake it. If the server played a few touchtones, like "4-5-7" (keypad "IKR") while saying "IKR", smart automated clients could detect the system and use it. To complete the interactivity protocol, every spoken digit to be pressed in the numbered menus would also play the digits' DTMF. And the intro to the scope...

...) and > my > softphones (IAXcomm, or JackenIAX) work perfectly with comedian mail. > > To me this means that asterisk is properly getting the RFC2833 events > from the user agents. > > BUT, if I try to dial out the FXO, none of my phones (hard or soft) > produce working touchtones for a PSTN based voicemail system. > > Even stranger to me, is the fact that from the phone connected to the > FXS on the wellgate I can hear tones(listening on a called line), but > they sound kind "rough" at the edges. From the AG168V I hear no > tones, but what se...

Hello All I am looking for a PC based switchboard application. Cisco CallManager has a web attendant console that allows you to use the PC to transfer calls and the like and I was wondering if there was a similar program compatible with *. Thank you in advance Keith D'Atrio -------------- next part -------------- An HTML attachment was scrubbed... URL: