similar to: A Simple Question

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:

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:

> > I've never seen stats, but it's probably a safe assumption that the > > majority of IP phones are sitting next to a PC and the additional > > expense has been incurred because "people want a phone that looks and > > works like a phone". That's certainly been my experience far > > outweighing any technical issues with quality or reliability

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:

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 I have a problem with dtmf recognition an analog lines connected to Sangoma A200. The digits (in most cases the first one) are doubled and so my IVR is useless. I tried to adjust the rxgain, toneduration and relxing the dtmf but nothing worked. I also noticed one thing it only happens during the background application: exten => s,1,Background(soundfile) exten => 111,1,Dial(SIP/111)

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.

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'm running Asterisk 1.8 from Debian. I have some analog phones connected via a TDM400P. I'm testing them with these simple extensions: exten => 600,1,Answer() same => n,Festival(This is an echo test) same => n,Festival(Hang up or press pound when you are done) same => n,Echo() same => n,Festival(Good-bye) same => n,Hangup() exten

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

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.

CALL FOR PAPERS The 11th International Conference on Computer Vision Theory and Applications ? VISAPP 2016 Website: http://www.visapp.visigrapp.org/ February 27 ? 29, 2016 Rome, Italy Regular Papers Paper Submission: September 17, 2015 Authors Notification: November 12, 2015 Camera Ready and Registration: November 27, 2015 Position Papers Paper Submission: October 29, 2015

CALL FOR PAPERS The 11th International Conference on Computer Vision Theory and Applications ? VISAPP 2016 Website: http://www.visapp.visigrapp.org/ February 27 ? 29, 2016 Rome, Italy Regular Papers Paper Submission: September 17, 2015 Authors Notification: November 12, 2015 Camera Ready and Registration: November 27, 2015 Position Papers Paper Submission: October 29, 2015

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 have fedora 6 and i am trying to istall a guest OS with virtual machine manager Unfortunatale, virt-manager gives "The console is currently unavailable What is wrong? Can anyone help? _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

Hi all, I`m a newbie and I need any help!! I want to istall xilinx under win and after the command wine setup.exe Iget the following message: fixme:module:CreateProcessA (D:\ce\jre\1.2\bin\java.exe,...): HIGH_PRIORITY_CLASS ignored fixme:pthread_kill_other_threads_np FIXME:pthread_rwlock_rdlock FIXME:pthread_rwlock_unlock FIXME:pthread_rwlock_rdlock FIXME:pthread_rwlock_unlock

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 >

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. :)

The first release of Voiceglue is now available. Voiceglue provides a VXML interpreter using Asterisk telephony and the OpenVXI VXML parsing suite. It is released under the GPL, and thus compatible with Asterisk and OpenVXI licensing. The first release is available at the project website: http://www.voiceglue.org There is also a mailing list for those interested in continued evolution of