search for: toning

I followed the how-to for R2 that coppice wrote on the webpage http://www.opencall.org/installing-mfcr2.html, now asterisk has channel unicall and when i test an incoming call I get the following errors. astbox:~/mediatel/libmfcr2-0.0.1# asterisk -r Asterisk 1.0.0, Copyright (C) 1999-2004 Digium. Written by Mark Spencer <markster@digium.com>

Maybe its just me but I find this question a little confusing, the tone duration should have no impact on tone recognition and typically in my experience the duration of the tone is defined by how long the user holds down the button !? > -----Original Message----- > From: James Sizemore [mailto:james@deny.org] > Sent: 22 August 2003 17:33 > To: asterisk-users@lists.digium.com >

I'm losing call files in /var/spool/asterisk/outgoing because * isn't able to detect the busy signal. The call file looks like this: Channel: Zap/g2/3036701917 MaxRetries: 1000 RetryTime: 60 WaitTime: 45 Application: TxFAX Data: <filename>.tiff|caller Using the "|caller" parameter, TxFax injects the fax tone (CNG) onto the line. With the CNG tone, asterisk is unable

Hi all, I would like to ask please if there is a way to play a ring back tone from asterisk when the customer try to make a call...I already added the ringing function to the context in extensions .conf and it work perfectly...But the issue that the asterisk server is stoping playing back his own ring back tone as soon as it detect a ring back tone coming from the carrier side... Is there a way

I'm going to be hooking FXS lines on a TDM400 to a PBX which doesn't drop line voltage at the end of a call, so I'm going to have to use busy detection. A few questions - The tones are taken from the tones specified by the zone in zaptel.conf, right? Which tones cause hangup? The PBX may not use the national standard tones. Does anyone have any suggestion for how I can

Few weeks back local telco introduced option of custom ring tones. I am not talking about your phone ring tones but about ring tones you hear in your headset while phone is ringing on the other end. If I understand correctly, ringing tone is generated localy on asterisk if you are connected to phone network with E1/T1 connection. Which means that instead of regular beep-beep tone we could send

dear Hackers, i have a voicetronix Openswitch card, and i have been finding it very dificult to get it to work with asterisk. i intend to connect 8 ports to the PSTN and 4 as station ports. problem 1: On running asterisk all i get at first i get : event[9=>[11] station OFF hook] on vpb/1-12 even [12=>[11] loop drop on vpb/1-12 event [12=>[11] Tone detect:GRUNT event [2=>[11] Dial

For some reason Asterisk is producing a US ring tone when it should be an Australian ring tone. I am using ztdummy and do not have any cards installed. My configuration is as follows. I am using Trixbox 1.2.2. Can someone please guide me into the right direction? zaptel.conf loadzone = au defaultzone = au zapata.conf [channels] language=au indications.conf [general]

I'm experiencing touch tone recognition issues when calling some outside phone systems. For instance, if I call my Nextel phone, and try to press * to enter my voicemail, Nextel's system does not "hear" the DTMF tone. I've also experienced other outside phone systems for which I am unable to use their touch tone menus. Oddly, this isn't the case with all outside systems.

Does asterisk have a way in the dialplan to generate tones? Say I want to play a tone 300Hz for 3 seconds. Can I do that? If not, can I use some system command to generate the wav file then just have asterisk play it? Jerry

Hello ! I 'm using a TE405P with a HW echocanceller module attached on it. dahdi version is dahdi-linux-complete-2.2.0.2+2.2.0. As far as I know, the fax tone detection is done on the FW board. How can I verify that the echo canceller has been turned off ? When I do a cat /proc/dahdi/1 for span 1 I see still the VPM450 entry near the channel although this was a fax call with CED tone.

I have the following setup: AnalogPhone1--TDM400P-ASTERISK---via SIP---Softswitch--------POTS Phone2 When I call from AnalogPhone1 to Phone2 I hear a ringing tone and all is well. When making a call from Phone2, I get a dial tone but after dialing the number I hear nothing (no ringing tone). On Asterisk console it says that a call is coming in and that it is ringing Zap/2. I can also hear the

I am wondering if it is possible for an extension that is served by a zaptel device to revert to dial tone once a call disconnects. For instance, if I make a call to another extension, talk with them, and THEY hang up, can I then be presented with a new dial tone rather than a congestion tone? Further, can an extension be set up so that, once the call goes back to dial tone, if the user does NOT

Oh geez nobody respond, I guess I have to be more descriptive. I'm install Asterisk and FXO FXS Lite dev kit from Digium everything has installed and I am getting an error message that my sound card is not enabled. I'm using a Dell Optix (whatever). Everything seems to be in order. I connect an analogue telephone to the FXO port? and try to dial. Dial tones work however not dial tone.

Does anyone know if it is possible to use the following disconnect tone setting with an x100p card? Disconnect Tone: 350@-19,440@-19;4(.25/.25/1+2) This tone was written for a Sipura SPA-3000 for a Panasonic KX-TD1232. The Panasonic does not support disconnect supervision, so this tone is the only thing that will detect a disconnect. It is not a standard fast busy or offhook tone.

Hello all, I've played with background and play sounds apps and googled around and asked the list before to no avail. Does anyone know of a way to have tones played during the call progress stage of the call? We (especially on some international circuits) get up to 5 seconds of silence before the phone starts ringing or is busy. I don't want to force "R" on the Dial app as

Hi all, I have a Diva Server V-BRI-2 card, which support, as written in reference guide: Extended tone processing (human talker detection, generation and detection of country-specific tones) I would like to detect human speech and fax tone with asterisk. I think that the diva card transmit a DTMF code when detecting voice, but chan_capi doesn't receive this DTMF code. I verbose it

On systems even key systems it is customary to have an 'internal' dial tone. Since Asterisk simply ignores the 9 and keeps the tone going it is hard to tell for some 'new users' if they can make a call. My first idea was to change the generated dial tone via source. Then if the user presses 9 go to a different context where I would record about 30 seconds of the normal dial

Im trying to configure voicemail, but asterisk doesnt respond to dtmf codes. I uses kphone with g711u codec (I've tryed the others one) and in sip.conf I configure dtmfmode=rfc2833 (I've tryied inband and info). Asterisk seems not to "see" the tones. Could somebody help me? Thanks

Good day, I am hoping that someone can assist me with a work around. I have an IVR system that I am attempting to connect asterisk to, however the IVR was written Some time ago, and requires tones to be approximately 2 seconds in length. I am using SIP Phones, Is there any way to adjust the Length of the Tones that asterisk sends out the audio path? Are there any phones that would allow