[Asterisk-Dev] Lost in Translation (DTMF Relay)

[Conroy, Lawrence (SMTP)]

Hi Sales, Folks,
   I'm confused.
The first part talks about H.323, and then the rest talks about SIP.
The peer lines look very much like H.323 (et al).

Thus, if you really DO have two Cisco phones that are both talking SIP.
then I don't see what this H.323 stuff has to do with anything.

The statement "I am using only G.729" is wrong, one way or another.
If you are transferring DTMF, this is either in RFC2833 frames, OR
it will get hopelessly scrambled by the G.729 codec.

I happily transfer DTMF using RFC2833 mode via *. It works.
Now, I'm using G.711 but this is completely irrelevant, as the DTMF
goes to * inside RFC2833-coded RTP frames, NOT inline in the audio 
codec.

Assuming that this isn't yet another Cisco-ism, I wonder if the H.323 
mung
is the problem here - it's either SIP or H.323/H.245/dadada

So - which is it, Spitz or Swallows?

all the best,
   Lawrence

On 24 Apr 2004, at 5:24 pm, sales at minixel.com wrote:

> Now that Jeremy has started, but not finished, the huge
> endeavor of fixing H323, perhaps somebody else may
> attempt to fix the other real issue that keeps many
> people from using Asterisk in a more business-oriented
> model: The DTMF Relay is useless, using either protocol
> or mode. I did this experiment which failed and I had
> to bypass Asterisk: The calls come from A Cisco 5300
> using SIP, where the dialpeer has the usual lines
>
>  “dial-peer voice 400001 voip
>  destination-pattern 5619531599
>  translate-outgoing called 1
>  session protocol sipv2
>  session target ipv4:xx.xxx.xxx.x
>  dtmf-relay cisco-rtp rtp-nte h245-signal
> h245-alphanumeric
>  codec g729br8 bytes 40
>  fax rate 14400
>  fax protocol t38 ls-redundancy 0 hs-redundancy 0
>  no vad
> !
>
> The middle man is Asterisk, and the receiving end is
> also another Cisco 5300, configured similarly. Whatever
> I do, the dtmf gets lost using any codec except g711
> and Inband, which is not exactly DTMF relay at all. It
> fails with rfc2833, info, etc. The bridge kills the
> dtmf. There are many business models that require this
> functionality. In this case, I had to send the call
> directly from Cisco to Cisco, because I could not
> receive a single DTMF tone on the other end. My client
> really would benefit from having Asterisk in the
> middle, but to no avail. I use only G729.
>
> Any ideas?
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