In the PEER details of the Trunk in FreePBX:Change dtmfmode=rfc2833 to dtmfmode=inbandOr add it if isn't there...Why?The device that you press the key on will generate the DTMF tones. - If Tried adding the line qualify=yes to my PEER trunk settings. ... Honee (Honee) 2015-07-13 03:51:33 UTC #16 Hi All, I experienced the same issue. Those of us who have been around a little while CAN often supply a quick fix if we are in a position to see, first hand, what the problem ACTUALLY is.
I have an inbound route with DID Number _123456XX that forwards to an IVR with Direct Dial = Extension enabled. If you call from your mobile to a regular analog phone line can you hear the dtmf when you press the keys? monotek 2008-07-08 15:26:46 UTC #6 Thanks for your help. Or permissions are wacky.
Press one to reach xxx, press 2 to reach xxx and so forth? They were good suggestions. I've asked Sipgate if they could disable the voicemail for one of my 'troublesome' accounts just to help prove/disprove my suspicion - I'll see what they say.
sgaul (Sgaul) 2014-12-12 09:06:19 UTC #3 so that means I have to tamper with extensions_custom.conf to do that? Log In IVR in freepbx not working General Help linuxman 2013-03-24 12:47:11 UTC #1 I setup a IVR in Freepbx and when I call the number that has the IVR I Taylor (#1) Account Manager at 10:05 PM Labels: asterisk, dtmf, dtmfmode, inband, inbound, ivr, peopleline, trixdox 2 comments: Nico van JaarsveldMarch 28, 2012 at 1:49 PMTHANKS it is workingReplyDeleteDiener Maick PiskeApril Asterisk Dtmf Not Working The only thing I could NOT test in this round was your suggestion to use a "host-port type setup rather than using a registration string." I do not understand how to
fskrotzki 2008-07-03 19:16:53 UTC #2 check with your trunk provider for dtmf signaling. Asterisk Ivr Dtmf Not Working Trunk Config (sanitized)?Router type-brand, using a POTS line? Final note: I tend to think you're close with the "one-way audio" diagnosis, but if it helps at all, VOICE audio for a call works both ways for calls originated inbound It's fairly easy to tell when a new member takes my suggestions and really runs with them, often making surprising progress.
Set an announcement on your IVR (I uploaded the recording, just forgot to set it as an announcement) and the error message should go away. Freepbx We Have Not Received A Valid Response What is the make and model of firewall that you are using? #15 leemason, Sep 19, 2013 briankelly63 Expand Collapse Guru Joined: Nov 14, 2008 Messages: 1,226 Likes Received: 267 INBOUND calls audio passes outbound, but not inbound. #20 youreontheair, Sep 19, 2013 (You must log in or sign up to post here.) Show Ignored Content Page 1 of 2 dicko 2014-12-12 08:45:44 UTC #2 look at the [from-did-direct] context in /etc/asterisk/extensions.conf and use it as a template for a custom context you can build to do that.
but the DTMF tones somehow don't pass, or get 'heard') The logs show things like: CSeq: 20943 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Found RTP audio format 0 Found RTP All I did was, set up my announcement - changed direct dial to extensions - left everything else but changed my invalid destination and timeout destination to terminate calls. Freepbx Dtmf Not Working and I zeroed in on looking for anything related to DTMF (because AUDIO works two-way... Freepbx Dtmf Logging linuxman 2013-03-25 16:58:23 UTC #6 Hi I did what you said to do and it did not work.
no change. It appears that tones are getting repeated. Often carriers will block inband dtmf tones or at best provide a short burst so you rely on the fact that the keypress is passed along the data channel. linuxman 2013-03-28 18:10:25 UTC #14 This is my Trunk setting: type=peersecret=Passwordusername=Usernamefromdomain=sip.voipwelcome.comhost=sip.voipwelcome.comoutboundproxy=sip.voipwelcome.comnat=yescontext=incoming&from-trunkcaninvite=nodisallow=allallow=ulaw,alawinsecure=port,invite brandonmc 2013-03-28 18:41:31 UTC #15 In your trunk settings, you can specify auto like this: dtmfmode=auto You can also set Freepbx Dtmf Mode
It has an announcement set for "dial 1 for this extension, dial 2 for this extension..." but when the user dials any number, a pause and a "invalid response" message is but do not work on all three numbers at the same time. The system does not respond to the DTMF. no change.
It does seem strange that it is allowing voice packets through but possibly not allowing (or mangling) DTMF packets. Freepbx Ivr Direct Dial Not Working Is direct dial on and working, if allowed? Even tried coding it into sip_general_custom.conf (upon reload of the Asterisk SIP settings in the GUI, it threw an error and recommended clearing the settings from sip_general_custom.conf).
Does the IVR work if you use a phone connected locally to the PBX? Discussion in 'Help' started by youreontheair, Sep 16, 2013. Might be better for you to explain what you're trying to do, so we can understand more about your IVR. Freepbx Ivr Direct Dial Extension SO he actually dialed 12345678-78.
Please elaborate? I only get problems when I am calling using my Linksys pap2t adaptor. mustardman 2012-12-26 18:35:05 UTC #9 This is a good beginner explanation of inband vs out of band with some recommendations and tips in case you don't know the basics.http://www.voipmechanic.com/dtmf-issues.htm You didn't But of course he could also enter extension 79 and than 79 rings after the enduser dialed 12345678-79Isn't it possible that just the last 2 digits after the actual number (see
To confirm a couple things in the meantime: -- when i use my mobile to call someone else's IVR, it works perfectly. -- when i temporarily route UDP inbound ports 10,000-20,000 I have a Sipgate Trunk which works fine for inbound and outbound calls. Like (I think it was Dicko) said, check with your provider as to which form of DTMF you have to send. It's not picking up the DTMF tone or something.
If it helps, I can add that DTMF from a softphone does work great locally on the network.
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