similar to: DTMF

I am having a problem trying to do inband DTMF passthru via asterisk. My setup: PSTN gateway MAXTNT v11.0 SIP (T1 PRI/NT2) Asterisk HEAD or v1.0 makes no difference (I am using HEAD mostly) 12/10/04 and 01/17/05 (no difference) CAC ABII-T100P to/from analog lines to/from asterisk BTW, I have used a ABI and it works just like the ABII with asterisk. What I am seeing is: I make a call from a

Hi all! I'm experimenting a strange problem in my Asterisk PBX: I've got an Asterisk pbx in the office: I dial an external number; the dialled number answers me correctly, but as soon as I press the '*' button (i.e. to navigate through the menus or to enter a password) my Asterisk box put me on hold. (CLI transcription follows: -- Executing

Hello, I did this post a long time ago but never solved the problem, so i'm trying again after something like 10 months, hopefully i'll find someone that found a solution ;-) When i call an external number that sends audio before call has been answered (like some PBX of public offices do here in italy), strange things happen: I'm using chan_capi, with Early B3 active, i can listen

I'm hoping people can suggest some ideas for debugging a problem that I'm having with DTMF. Unlike most of the DTMF problems reported here, it has nothing to do with Asterisk interpreting DTMF. My problem is with the synthesis of DTMF tones on outbound calls on a PRI connected to a TE412P card. I'm running * 1.4.10.1 with Zaptel 1.4.4. It is important to note that these problems

Hi, After some testing I've found out that my client's hardware recognizes DTMF only if digits are sent 50ms apart with 50ms of tone duration. This was tested using a test device which generates DTMF. Now asterisk doesn't do it by default because digits going out from Asterisk are not being recognized. Using command sendDTMF, I can control inter-digit duration, and using

Hi, This error I got it just when I gonfigure zaptel support isdneuro 31 channels. But if I configure zaptel to support T1 and just 24 channels I have no problem. ######################################################################## #### # # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # loadzone = it span=1,1,0,ccs,hdb3,crc4 bchan=1-23 dchan=16

Hi, is there a possibility to tell zaptel or Asterisk to modify the DTMF sensibility? The problem what i have is that the Asterisk don't get all Numbers which the analog-FAX dial, let say the FAX dial 123456789 the Asterisk get to number 24679. I think that can be to DTMF Tone duration or the Frequenzy. so you got yna idea what it could be? Thx for helping me. Bye Timm

Does anyone have any idea what would cause distorted audio but ONLY for DTMF tones coming in over our analog lines. (The analog interfaces are X100P's). I have carefully adjusted the gains in the zapata.conf using a local test line after trying various settings with no gain or just random gain settings. RelaxDTMF has no effect. I set up a monitor command in my dial plan to capture

Hi all, I have been looking at the code for both the zaptel driver (wctdm.c/wcfxs.c) and the asterisk channel driver (chan_zap.c) trying to figure out how much of this that has been implemented. So far I can see that the current stable 1.0.9.1 zaptel driver don't have the SETPOLARITY ioctl that would be required to properly signal the Swedish/Dutch CLIP, but the 1.2 beta1 has this

Hi all, I'm having a problem with * being very finicky about the length of DTMF key-presses during menus, voicemail, etc. Basically, short (<100 ms) tones are ignored, anything between 100ms (or so) and about 300ms is correctly detected, and anything >300ms is interpreted as multiple presses of the same key. This is terrible for callers who are trying to get to the correct

I am currently using Asterisk with G.711 codecs and in-band DTMF for several Cisco 7960's and an Audiocodes GW. When allowing out-of-band DTMF, I could use voicemail menus and anything else on Asterisk that required DTMF but I could not get the DTMF relayed out of the GW. Has anyone verified that this works between 2 SIP devices? If so, I would be interested in your settings. Also, I would

Hello, I am trying to make my Asterisk to recognize CallerID from incoming PSTN calls. I am using a TDM400. CallerID in my country's PSTN is based on DTMF. Some information from my configuration files: /etc/zaptel.conf: loadzone=nl defaultzone=nl /etc/asterisk/zapata.conf usecallerid=yes cidsignalling=dtmf cidstart=polarity As displayed on the console, my Asterisk cannot detect the event

Hi, I am having this issue that with one of the Asterisk servers, on zaptel hardware, that DTMF tones are 10-30 Hz too high than the upper limit for any DTMF digit frequency. This is causing problem with the equipment on the other end which is trying to recognize the DTMF digits. I tried different cards from the same vendor but it didn't help. For example for digit 1 where it should generate

Hi helpers, I seem to have a problem of intermittent DTMF tones being played during a conversation. Eg: Extn 100 takes an inbound call and all is fine. Except, at an undetermined time the person on extn 100 will here a DTMF tone for no apparent reason (it's not the caller pressing buttons). The caller doesn't hear the tone - only the called person. The call itself progresses normally.

Actually I am using Asterisk 1.2.7.1, zaptel 1.2.5, libpri 1.2.2 I ve tried many values for rx/txgain togeher with echocancel and relaxdtmf. The detection is not working with call file, manager originate and not with the dial command to the mobile. I have no ideas left. I got it sometimes to work if I use a specific channel (i.e. Dial(ZAP/14/...) But with the same vaules on a second call there

When dtmfmode is set to inband for SIP, and i originate a call from sip out to the PSTN, I can hear the DTMF digit twice in the audio stream. Once very briefly and once for normal duration. Our Theory: While Asterisk is parsing the DTMF, for a fraction of a second, while the end user generated DTMF is being detected, the DTMF is passed inband. Once the DTMF is detected Asterisk silences it

Hi, I have asked this question months ago - i have "toggled down" all DTMF Recognizations in my Asterisk (no more features etc) and found more people which recognized the same problem, but i cant find any help for them and me. The Problem (short as possible) : In a randomly call in my business day some unit in my Asterisk System sends an randomly DTMF Tone, like "A"

Hi, is there a way to enable dtmf detection on zap channels? I am trying to pickup, play a ringtone and the dial out. I.e. exten => s,1,Wait,1 exten => s,1,Answer exten => s,2,Playtones(dial) exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,10 exten => _X,1,StopPlaytones exten => _X,2,Dial,Zap/g8/BYEXTENSION|10

Hi list! I have an Asterisk server (1.2.14) connected to a E1 line via a TE410P, and some PAP2NA connected to it. The PAP2 DTMF configurations is set to INFO and Asterisk to INFO too. At first, is INFO method different from RFC2833?? Well, I have two problems. The first is that when I place a call to outside, via E1 trunk, sometimes I get some DTMF tones and I'm sure nobody hit any key. Seems

I've been using one of the popular asterisk ISO distributions for a couple of years and DTMF had always worked. I recently switched to another asterisk ISO distribution, and outbound DTMF is no longer working. After doing a bit of digging, I noticed that the new distribution wasn't setting any sort of dtmfmode at all, anywhere. The old distribution had rfc2833 set in sip.conf