similar to: serious dtmf recognition problem.

Hi, I'm getting unreliable dtmf recognition (it works fine for 4-5 digits, errors (duplicates) on more), when transferred inband from gsm gateway to NT port of quadbri under bristuffed Asterisk..... Since Asterisk is claimed to have good dtmf recognizer, I suspect there are some settings to workarouned... I've tried dtmf relax, but didn't help, so I suspect gain settings.... Is

Hello everyone, I have problem with DTMF recognition when calling from PSTN, my Asterisk box won't read DTMF tone at all. I've tried use cellphone, normal telephone and voip lines, nothing worked. softphone to softphone within extensions are ok. I'm a newbie at this, can anyone point me out where to look? I'd really appreciated. Thanks a lot Nate -------------- next part

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

Is it possible to turn off DTMF recognition (and all transfer services etc.) pending on CallerID (or FXS channel)? Some of the FXS channels I will setup soon, is going to work exactly like POTS. It will be used by people not knowing their within Asterisk. They will be pretty confused when "Transfer" is playbacked in the handset. :)

Hi, in asterisk 1.6.2.16 I get spurious DTMF recognition over SIP from an Audiocodes. I think the DTMF recognition is the Audiocdes' fault, the Audiocodes log seems to say so as well, but I want to make sure, and fixing the Audiocodes is not an option in this particular case - don't ask. Can someone explain to me what the following means *exactly* [Feb 10 21:15:40] DTMF[2538] channel.c:

Hello, I am experiencing a strange problem and I am wondering if anyone may have some pointers as to how to overcome it. I have an account with VoipTalk here in the UK which I have connected to my Asterisk server. VoipTalk supports IAX2 and SIP and I have connected to my Asterisk box using both methods. The problem is when I dial into my Asterisk box via my VoipTalk incoming PSTN phone

I have been having problems dialing into Verizon conferencing using my * system. If i dial using a POTS line directly, the dtmf codes for the conference room are recognized with no issues, however, verizon doesnt recognize the keys when i press them as being a valid code. I remember something a while back (unfortunately - lost all my IMAP email) - Something about verizon lines and dtmf and

Can anyone recommend some phone sets that are adsi compliant and work well with asterisk?

Thanks for your responses - it looks like I have the following options, in order of ease: 1: Modify and recompile app_record.c Change line 471 https://github.com/asterisk/asterisk/blob/master/apps/app_record.c#L471 from status_response = "DTMF"; to status_response = dtmf_integer; Pro: Free, easy Con: Have to remember to edit module each time a new Asterisk update comes out 2:

Thanks Jurijs, Yes, in fact I'm already using that, and it works fine. The problem here is that I cannot find a way of recording speech AND listening for a DTMF digit being pressed as an alternative. That's where the problem lies. J.

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.

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 >

Hi, I have successfully installed libunicall and mfcr2 in Venezuelan variant. We are able to receive, make calls without any problems. When we place a call from a snom sip phone through chan_unicall and press a dtmf (either using info, rfc2833 or inband) Asterisk seems to get the order of playing dtmf but a garbage is heard at the other side. I have been trying to narrow down the problem on

Briefly: I want to be able to have "press or say (number)", with Asterisk listening for a spoken number, but accepting a DTMF digit, too. I'm posting everything I found so far, here, partly to show working, but also in case anyone else finds it useful. So, moving on.... This looked hopeful for a moment until I realised that it doesn't do DTMF:

Which speed recognition products will also recognize DTMF? In other words, I want to say "Please speak or dial the conference number". Does Vestec allow that? LumenVox? Any other way?

I have a couple of DID's with LiveVoip and am having major DTMF issues on incoming calls. I am connecting to them through IAX using ULAW. When someone dials one of these DD's (from a landline) they are for the most part unable to navigate the IVR menu successfuly. I would say the failure rate is greater than 80%. For example if the caller presses 5 sometimes * will see the DTMF as 55 or

I'm getting down to the last of my * issues ... After calling a bank, or cc processing center; you have to enter your social security number, or the cc number - followed by the # key. The lovely * voice responds "transfering" I'm sorry that was an invalade selection. Sometimes the IVR on the other end still gets the digits and proceeds; other times it breaks the IVR on the

Hello list, 'Just wondering if anyone can point to a very light-weight and easy to incorporate into Asterisk (v. 11.x) to handle a minimal set of responses, like: 0 - 9 yes no (maybe * and # for some people) The idea is that within an IVR menu, the caller could respond by speaking to the typical IVR options, like: For Archie, press or say 1 now For Veronica, press or say

Dear all i have a TE205P connected to an Asterisk 1.2.18. Yes i know, the version is old but since now the system was stable and i don't have the necessity of an upgrade. The system provide an IVR service that: 1) receive the call 2) verify the queue length 3) hangup if queue length is > 1 4) put the call in the queue othervise Then, there is an AGI php script that 1) verify the queue

A new RFC was published today, RFC 3601: Abstract: "This memo describes the full set of notations needed to represent a text string in a Dial Sequence. A Dial Sequence is normally composed of Dual Tone Multi Frequency (DTMF) elements, plus separators and additional "actions" (such as "wait for dialtone", "pause for N secs", etc.) which could be