similar to: Incorrect DTMF detection in Asterisk 1.8

Hi, it seems that there is a bug in asterisk's dtmf mode autodetection. Assume following sip.conf: [sipprovider] disallow=all allow=g726 dtmfmode=auto DTMF does not work. It seems rfc2833 mode is chosen despite it being obvious that this cannot work! The following configuration is necessary to get DTMF to work: dtmfmode=info In my opinion, this behaviour is counter-intuitive. I am using

Hello all, When I generate an out-going call from *, the DTMF detection is not working ? ASTERISK --> GW AS5300 --> PSTN But the DTMF is working correctly when it's an incoming call. PSTN - -> GW AS5300 - -> ASTERISK Well, I tried with the 3 dtmfmode of asterisk inband, rfc2833 and info, no way !!! Is it normal that asterisk try to setup the outgoing-call using ULAW ? if I

Hi. I am having problems with asterisk detection dtmf properly in asterisk 11. I am up to rev 390229. Now, when coming in off a did we have with Velocity, the dids work fine, but from extensions often it misses digits -- I can type *4 and it will miss the 4. Often, if I type quite slowly things will work properly. All dtmf modes are set to rfc2833. Strangely enough, I did not notice this with

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

Hello list, I'm having some troubles with DTMF tones. When pressing numbers on a Snom phone, the DTMF-signal takes too long. I have the following in sip.conf : dtmfmode = rfc2833 which works well for Grandstream, Yealink and Cisco phones. But not for Snom. Snom support tells me I should use SIP info. Is it possible to have something like this : dtmfmode = rfc2833, info ?? Because

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

Problem: Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that Asterisk can detect and use. It worked in 1.0.5, but has not worked since. This has been verified on SoundPoint IP 300's and SoundPoint IP 600's. Workaround: It used to be that for DTMF to work, I had to set the mode in sip.conf to "inband". Without making any configuration changes on the

Hi.. I just wondering why DTMF are not recognized by aterisk on incoming calls from my SIP provider ... ANy suggesteions ?` /Mike

I got no resonses to this and some funny bounces so I'm trying again. First of all Merry Christmas. Second, my first problem with my provider not staying registered with our server was my fault. We moved our server room and I restarted the test system and the production system causing them to ping-pong back and forth registering with our provider causing random problems, they are both

I am trying to connect to a vocal server from an asterisk server. A call is received via iax2 to my asterisk server. I then initiate a SIP connection to the vocal server. everything works great except dtmf doesnt work. A cisco 5300 can connect to this vocal server and do dtmf without a problem. I have my dtmf set to rfc2833 in the general section of the sip.conf . I can confirm that the

Hello list, Hope you are all doing fine! I have stumbled over some piece of dialplan code in which apparently they were trying to avoid recording the DTMF tones in the wav file. It is really messy and I am not sure if this really works. So after a bit of research I found this comment ( https://community.asterisk.org/t/asterisk-dtmf-record/65040) in which it is said: *"Asterisk strips the

Hi, I'm trying to connect Asterisk 1.4 and Cisco CallManager 5 using SIP Trunk without MTP (media termination point). Howerver, Cisco 79xx phones do not support RFC2833, they always notify CCM5 via SKINNY channel no matter where they send RTP to. For non-MTP trunk there's Out-of-band DTMF support in CCM5 called "kpml". I wonder if Asterisk can support it. I found an

We are experiencing intermittent DTMF problems here, with the following setup: ITSP -> PIX -> Asterisk (g729, RFC2833 for DTMF). I am running Ubuntu server 10.04, but Asterisk is compiled by us and not installed from the software repository. Essentially, DTMF works for some time, but at some point it simply stops and the point at which it stops appears to be random. Using RTP debug, I

Hi buddies, I encountered DTMF issue when I tried to place call from x-lite to a sip conference serice,here is the diagram. X-lite---->Asterisk--->Cisco SIP proxy---->SIP Conference service The Call can be established,and I can hear from x-lite the prompt of the conference,but when I input any digits,nothing happened,the conference service did not recognize my input.At the same time,in

For efficiency & reliability, when SIP transmits DTMF as non-audio data, it uses RFC2833 or INFO. My question is - (not knowing much about IAX2) - when IAX2 transmits DTMF as non-audio data - is it also using RFC2833 and/or INFO, or it it using some other IAX2-specific mechanism with its own name? Thank you, Brent -------------- next part -------------- An HTML attachment was scrubbed...

Hello, I have one problem and I need to disable dtmf (disable rfc2833, info and inband) on one (other peers must support dtmf) SIP peer . Is it possible? Workaround would be use g729 codec with dtmfmode=inband. Maybe there is better solution? Thanks for help. -- Pagarbiai / Best Regards, Giedrius Augys -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi again, I am trying to get my DTMF to use RFC 2833 rather then inband, so that I can utilize lower bandwidth codecs through my FXO. After much tinkering I was able to get my gateway (wellgate 3701A) configured to a point where I have some success, but no real joy. I have configured the RTP Payload type (or RFC2833 Payload type) to 101. I don't have a clue what this means, but I took

I am using viatalk as my voip provider and they use dtmf=rfc2833, but asterisk is not seeing any of the dtmf. I am using CVShead as of 8/26/05. Nothing in the logs indicates a dtmf is being seen. If I use my pots line it sees it fine. Any assistance would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici

hello i want to kick participant in a meeting by pressing the digit on sip phone.when i entry the meeting ,no matter how i press the button,the dtmf does not work. here is my dialplan and my agi script,and sip.conf [from-internal] exten =>121,1,MeetMeCount(900,CONFCOUNT) exten =>121,2,GotoIF($[${CONFCOUNT}<10]?3:100) exten =>121,3,Authenticate(123456) exten

How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no messages related to DTMF... or if I just do a global SIP debug for that matter.... I am using RFC DTMF but it's not being passed to the PSTN and I need to debug this further. I've tried to increase the verbosity and the debug ('set debug n') and that didn't help either. I assume this is because even