search for: dtmfs

...proxy ( which does NOT do RTP, only SIP ). I called asterisk from my hard phone ( gxp2000 ) by dialing 22. I see the console DTMF messages indicating the DTMF was sent or received. ( I forgot to keep this output ). I than watch the console DTMF output on asterisk-pri and it showed about half the DTMFs. The pager that was called showed the DTMFs that appeared on the asterisk-pri console. So somewhere between the two machines, the DTMFs have disappeared. So I ran TCPDump on asterisk and saw that close to half of the DTMF events were never sent. tcpdump -i eth0 -n -s 0 dst asterisk-pri-ip -vvv -w...

Hello A user seems unable to type DTMF in our Asterisk IVR menu. Can this be due to their phone or PBX that disables DTMFs when a user is off-hook? Thank you.

We need to conference together a call center agent, a customer and a third party IVR and send DTMF tones from the agent to the IVR. MeetMe has been eating our DTMFs so we'd like to try app_conference. Has anybody setup such a configuration in app_conference and how did you configure it? -HJC

Hi, I've just installed DAHDI at two PBXs as follows: *PBX-1 PBX-2* FXO ------------- FXS When I try to send calls from PBX-1 to PBX-2 I just receive the message: "Starting simple switch on 'DAHDI/1-1" It seems like if PBX-1 couldn't send DTMFs, but when I set immediate=yes at chan_dahdi.conf at PBX-2 dialplan is executed at PBX-2 but nothing is heard at PBX-1 I'm using a TDM400P card, configuration files bellow *chan_dahdi.conf* [channels] usecallerid=yes hidecallerid=yes callwaiting=yes threewaycalling=yes transfer=yes echocancel=...

Hi I've got an Asterisk box connected to a Siemens Hicom 300 using a Digium TDM400. The Hicom provides the calling extension as DTMF at the beginning of the call followed by two *, as in 3425** when 3425 calls my extension, I can hear all 6 tones if I have a handset connected but using Asterisk's Read application straight after Answer() Asterisk usually only gets the last *, sometimes the

On Mon, Nov 17, 2008 at 10:20 AM, Tony Mountifield <tony at softins.clara.co.uk > wrote: > > If I do this from an NEC digital extension I get 141496920000, but if I > do > > it from an NEC POTS extension I get 1942124000 > > That looks like when you pick up the analogue phone and dial 9, it > immediately opens the outgoing line and sends the 141 acces code, but >

Thanks very much for the reply Joshua! So I guess that setting dtmfmode=auto would be the safest choice in order to strip out the DTMFs from the recording, right? Cheers! Patrick Wakano On Tue, 1 May 2018, 19:36 Joshua Colp, <jcolp at digium.com> wrote: > On Mon, Apr 30, 2018, at 11:23 PM, Patrick Wakano wrote: > > Hello list, > > Hope you are all doing fine! > > > > I have stumbled over some piec...

Hi ! Did anyone had issues/managed to solve issues with DISA over Zap channels on * 1.0.X (STABLE) ? I have a situatuion where DTMFs that should be recognized in DISA work over SIP channels and do not work over ZAP channels (Zap channels are on TE110P) I have in default context: exten=> 299,1,DISA(no-password|default) and I have SIP extension 200 in [default] and I have Zap trunk which context=default. When I call 299 fr...

Would anybody mind if the the following command options where added to MeetMeAdmin? 0 - 9, * and # I'm considering hacking the code to add these commands to play the DTMFs to the specified user as tones and hope that the SIP or IAX channels then work with these correctly. -HJC

...risk strips the DTMF from the audio stream when configured for inband, so internal stuff can react to the DTMF and so the other side does not hear the tone unless they are using inband (in which case it is regenerated)"* So my questions are, what are the cases in which Asterisk regenerates the DTMFs? Does it cause the recording to have the tone as well, or is it only transmitted to the other leg without being generated to the recording file? Also, what if one or both legs are RFC2833? From my tests the RFC2833 events never show up in the recording, but I just want to confirm that this is alway...

Hi, I'm having two problems. First - I'm using the xten x-lite program to communicate with asterisk, and everything works fine except that DTMFs are not transferred. I've set DTMFMODE to inband on both the sip.conf file and the x-lite configuration, and still it doesn't work. Anyone had this problem before>? Second thing: I get a WARNING:[1209214400]: File dsp.c, line 1198 (ast_dsp_process): unable to detect process 2 f...

Greetings, What is the recommended settings for using SPA-3000's FXO port for dialing out to PSTN in regard of the DTMF? The voip lan contains SPA-2100 and SPA-3000, with all fxs/fxo ports registered to the Asterisk box with unique username/passwords. The inbound PSTN DTMF works excellently, e.g. people calling from PSTN into the * box are able to pick IVR items with DTMF reliably. The

Hi everyone, Could somebody enlighten me on this one? I have configured my asterisk to run on oh323 using codec g729. Incoming calls are working okay. But the thing I want to work is say pressing some options, say dial 1 to go to voicemail or dial a certain number to dial a specific extension. I have a config for this and tried calling from a normal PSTN and is working. But i just can't seem

Hello! I have asterisk updated from CVS on 31/8/2004 with sample configuration. I have just changed the sip.conf to register asterisk with sip proxy in out intranet. Then I can successfully make call to asterisk and go to demo IVR, but no response to dtmfs. I try to make call from several sip phones: Cisco7960, Ata186, Snom200. All of them send telephone-event in INVITE, but asterisk answers with no telephone-event in OK. Only Sipura3000 "manages" to get answer with telephone-event in OK and that's why asterisk detects dtmfs. I try to...

...extremely reasonable not sending any audio when a PROGRESS message is received on a PRI channel (isn't it an early-media session or one-way audio session?), nevertheless some Italian IVRs expect the user to select the proper option by sending DTMF. Now my asterisk box understands correctly the DTMFs on the caller SIP channel, but it doesn't forward them on the Zap channel. Are there any configuration or any other way to let the asterisk forwards the DTMFs to a zap channel in progress? Any suggestions are very appreciated, Thanks for your attention, Regards, Francesco PS I'm using As...

Hello! I've configured Music on Hold in asterisk, the only, most certainly, stupid problem I have is, which DTMFs to send to activate and deactivate it. If I use the cli, I can establish a call with originate. With the "misdn send digit" command I can send a number of digits to the other party. But what are the combinations to put the other one on hold? Or do I have to use a completely differen...

Hello! I'm still wondering, how to park a call with an ISDN line. The setup is the asterisk server only, controlled via the CLI. I can originate a call and I can tell asterisk to start the JACK application. But I can't then park the call. I tried it with sending DTMFs with misdn send digit, no luck. I had a look at the CLI, but didn't stumble upon a command to park the call. What's the procedure in these circumstances? Can anybody please help me? Kindest regards Julien -------- Music was my first love and it will be my last (John Miles)...

Hello everyone. I've got this really annoying HFC Cologne card (or however I should call it - a single channel ISDN card based on the HFC chipset). It wrongfully detects lots and lots and lots of incoming DTMFs, to the point the card is not usable. Here's a sample out of CLI: P[ 1] I IND :DTMF_TONE oad:206361 dad:520101 P[ 1] --> mode:TE cause:16 ocause:16 rad: cad: P[ 1] --> facility:FAC_NONE out_facility:FAC_NONE P[ 1] --> info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0 P[ 1]...

...: X-lite (build: 1084) Calling and get calls on PSTN from X-Lite is no problem. We only get sound from PSTN to X-lite. Never from X.-lite to PSTN. The soundmeter on X-lite shows activity ... (not muted, correct device...) When pressing numbers while having these silent calls in x-lite is playing DTMFs at the PSTN phone side. sip.conf: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to allow=all [1*phonenumber*] type=friend username=NAME secret=testpass auth=md5 nat=no host=dynamic reinvite=no canreinvite=no dtmfmode=inband callerid...

Hi, I have encountered a DTMF issue. My scenario: Access carrier-----sip----> Asterisk-1.4.25.1-----sip---->CiscoGW-----ISDN----->TDM Switch the access carrier sends to Asterisk out of band (rfc2833) dtmf, Asterisk forwards it with SIP INFO method to Cisco gateway, but on TDM switch every digit is duplicated. Is it possible that the carrier sends inband along with rfc2833? Kind