[asterisk-users] setting dtmf mode for a particular peer

[Brian J. Murrell]

On Wed, 2008-04-09 at 17:21 -0500, Brent Davidson wrote:
> Have you tried the using the "SIPDtmfMode" function in your dial plan?

Not sure how I would introduce that with my enum macro, but as a test I
did try it for this particular peer:

    -- Executing [01118668398145 at internal-sip:1] Goto("SIP/1011002206-b7232f10", "18668398145|1") in new stack
    -- Goto (internal-sip,18668398145,1)
    -- Executing [18668398145 at internal-sip:1] SIPDtmfMode("SIP/1011002206-b7232f10", "info") in new stack
    -- Executing [18668398145 at internal-sip:2] Macro("SIP/1011002206-b7232f10", "ringingdial|SIP/18668398145 at voipmich") in new stack
    -- Executing [s at macro-ringingdial:1] Ringing("SIP/1011002206-b7232f10", "") in new stack
    -- Executing [s at macro-ringingdial:2] Dial("SIP/1011002206-b7232f10", "SIP/18668398145 at voipmich") in new stack

But still, with "sip set debug peer voipmich" I see the initial SIP
packets establishing the session but no SIP packets when I press buttons
on my phone.

> It can be used to change the DTMF mode between two points in a call. 

Yeah.  I had noticed it before but was not sure how I would introduce it
given that selection of the given SIP server was more or less random
given that it's an ENUM destination.

> The problem, I would think, would be if your phones are set up to ONLY
> send inband audio then you have to find someway to get audio to
> transcode the DTMF from inband to info.

Oh, damnit.  I thought for sure this phone I was using was configured
for rfc2833 or at least info but it seems I have it set for inband.

Is there any way to determine what methods a given SIP phone supports?

> I'm not familiar enough with the specifics of Asterisk's behavior to
> know whether that "just works" or if it needs some special setup.  Try
> putting SipDtmfMode(info) just before the dial command and see what
> happens.

Yeah, did that as above, but no joy.  But that could be due to my
sipphone->Asterisk connection.

Does anyone know if Asterisk will convert an inband DTMF from one sip
channel to an info or rfc2833 DTMF on another (i.e. bridged) SIP
channel?

b.

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