Does the SIpml5 api support sending dtmf?

[Damian Patrick]

I have successfully connected a website to an asterisk IVR (custom built with Opus codec) using sln48 HD quality prompts.  It sounds incredible.  Now I  just need to build a dialpad on my webpage and tie in DTMF events so users can navigate the IVR.  Please tell me this is currently supported.  I read through all the documentation and found no mention of sending DTMF tones, so I am thinking it's not possible.  

So far I am very impressed by the sipML5 api and webrtc2sip and basically the entire project as a whole, it's absolutely amazing to me.  I just had to add that.  The quality is incredible.

Please let me know if I can send DTMF during a call, thanks.

Thank you.

[Damian Patrick]

Okay, I found an old post where a user simply called the method "dtmf" on the callsession variable.  This has allowed me to send dtmf to asterisk in the form of INFO messages.  However, these messages are responded to by asterisk with a 200 OK, but the DTMF digit seems to be ignored.  Is there a special setting I must use to capture DTMF INFO messages?  If memory serves INFO is not RFC2833, but a different method of sending DTMF.  My memory is hazy in this area.  Thanks for any help.

D

[Mamadou DIOP]

http://sipml5.org/docgen/symbols/SIPml.Session.Call.html#dtmf

For now only SIP INFO is supported.

If you're using webrtc2sip it will be relayed as RTP event.

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[Damian Patrick]

Okay, got it worked out.  In case anyone else has trouble sending dtmf tones to asterisk you just need to set the dtmfmode to info in sip.conf.  To send the dtmf tones from sipml5, simply call dtmf from your sip session variable.

For example:

callSession.dtmf(0) 

Will send a zero dtmf digit during the call.

Thanks,

On Friday, September 13, 2013 4:14:02 PM UTC-5, Damian Patrick wrote: