similar to: DTMF control for TDM device?

I hesitate to open up another bug ticket on this (my older one just having been closed), but after having worked for some number of CVS iterations, my Zap TDMXX stations seem to be struggling again to send DTMF clear enough to be "understood" by IVR equipment I interact with. I can use the same channels with either of my SIP phones and things are just fine. I have listened in, and

Hi everybody, I am having DTMF detection problem on DISA with my callback system. For many users, it keeps playing the dialtone even after they have input their number. I have trunk setup to both g729 and ulaw. What could be the reason for this problem. Some users have to dial a few times before the system can recognize their dialed number. -- Zeeshan A Zakaria -------------- next part

Can anyone explain why this is happening? I have a server attached to a phone line that will play a .wav file, then play all the dtmf digits (after it answers the call). If I place a call from a SIP device (like a Cisco 7960 phone) through Asterisk and on to the test server, via PSTN, the .wav file sounds fine, but the DTMF digits are distorted ----->------------->--------------------audio

Hi, I'm curious if there is any other option beside relaxdtmf in zapata , or any where else to tune dtmf detection on TDM400 fxo boards. in one of our sites provider is giving us 4 analog lines out of Adtran router and Asterisk often recognize DTMF wrong. Obviously playing with relaxdtmf was not helpfull. What do we know anout 1.2 and 1.4 DTMF handling diffrences? At this time i'm using

Still grasping at straws trying to solve DTMF detection issues with one of my asterisk servers. This particular server is now running Asterisk 1.4.18.1 and Zaptel 1.4.9.2 in runlevel 3 (console only) with 2 X100P cards. I have tried adjusting channel gains, turning call progress and relaxdtmf on and off, switching echo cancelers, just about everything that Google turns up and I can't

I've searched the site with google, but can't think of the magic words I guess. I got a "swap out" TDM30 today to replace my buzzy one. I swapped it with the older one, swapped out the FXS modules, hooked it up to the computer's power supply, and booted, but the wcfxs driver won't load--it gives me the standard "Cannot init module" types of errors that

Hello all. Please, I'd like to know if somebody can help me with this problem. I have successfully configured a PBX with Asterisk 1.4 and a Digium analog card with 4 ports. This PBX has a lot of incoming and outgoing calls, and works perfect in general, but there are some extrange cases where an incoming call is bridget with an outgoing call, and the caller that is calling TO the PBX can

I sent a query about this before, but have some further information and am hoping somebody has a suggestion as to what to try next to debug this. I'm using an Asterisk box primarily for MeetMe conferencing. There are two sources: TDM via two Q.SIG T1's and SIP phones. Conferencing works fine between TDM channels. But when a SIP phone calls the conference, there's no voice path *to*

Im having a problem getting the 2 cards working together. I added the TDM with 3 FXO and 1 FXS board. The TE405p is working fine. THe TDM registers and identifies the FXO's and FXS but cant get them to go off hook. I hope someone can help. Asterisk seems to THINK its dialing it sends manager status as dialing, and it just sits there with no tones, on the phone dialing out. If I dial,

Hi, everyone: I am developing a system using Asterisk, TDM-400 analog cards, analog lines, and Polycom SIP phones for internal extensions. Initially there was bad echo but after a series of efforts, I've managed to reduce it to a negligible level (it only happens when both parties speak simultaneously, and even there, only for a few hundred milliseconds). From an echo standpoint, things are

Hi Is there a phone comparison matrix I could consult I have a series of features that I would like to evaluate on the most common phones on the market example: dual-ethernet POE / direct power / both number of lines speed dials programmable buttons BLF LEDS Headset plug conference call built in hands free operation display size codecs communication protocol (SIP, h.323) price availability

Hello all, We have the following problem: When calling via iax, the sound is off after a while - most often after about 5 minutes (sometimes later or earlier) - at one end or at both ends. While the channel is up, and packages are still being transmitted, you just can't hear anything. Sometimes you can hear something just a little, but with the voice greatly distorted, sounding like a

Hi all, I have some probleem detecting DTMF send by a GSM phone, I'm using SIP with ulaw. do you know what are the options to improve the detection ? I'm using asterisk 1.05, is the CVS HEAD version had some improvement about DTMF detection? Florian.

Hi, I couldn't find a discussion that specifically addresses this, so here it is. I'm using Speex AEC in my mobile VoIP application to cancel speaker echo. The used version is 1.2rc1 from the website, and I'm compiling with fixed-point. On most occasions, the AEC works very well and cancels most of the echo (combined with the preprocessor). On some devices, where the microphone signal

I''m new to Rails (and liking it!) and I have multiple controllers, each of which has a method which uses a view template which is identical to what the others use. I would like to be able to have just one template to maintain that they can all use. What is the customary way to do this? I tried using render_action to specify the template explicitly, and it "works". But I

Hi all. Why don't Asterisk support inband DTMF with G729? Is there a way to do that!? Are you using RFC2833? Doesn't it a security hole? Thanks. Denis.

Hello All, We recently changed our asterisk system to begin using G.729a as the primary codec. We have a Cisco 1700-series router which connects to the PSTN via FXO ports, along with Cisco 7940 SIP phones. Everything is working great, except... When an inbound caller calls into our system, they hear an IVR. When the caller dials an ext (SIP phone), the ringing progress tone is

I'm trying to come to a decision between CentOS 4.2 and Fedora Core 4 for use on a server. One of the things the server will be serving is X desktops so there are some advantages to Fedora. However, one thing I can't help but notice is that the patch volume for FC4 from Oct 11 2005 thru the present compared to CentOS 4.2 for the same period is about 5 times greater. In fact, since

I'm sure there must be a setting I'm missing somewhere, so I thought I might was well ask here. Conversations are punctuated by sudden replacement of a given syllable or so of conversation with a DTMF tone. I would hope perhaps there's some kind of setting that has to do with the way it detects inband DTMF? I'm pretty sure it's an artifact of this particular ATA; my

Hello, we would like to hook up analog modems behind an Asterisk server, and we're very interested in the experiences that others have made when attempting that. We assume that there are no inherent problems with modems in respect to the Asterisk software, but it appears that the FXO/FXS hardware restricts this kind of a setup to lower data transmission rates, is this correct? Currently,