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

[Eric Wieling]

I'm sure you know that if there is no matching [sipentry] for an
incoming call it will be allowed thru and will take it's settings from
[general] in sip.conf.  Try setting something like context=INVALID in
[general] and then set a context= line for each [sipentry].  That way if
a connection doesn't match a [sipentry] it will fail, rather than be
accepted with all the default settings.

On Sat, 2004-04-24 at 11:24, 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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-- 
          Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
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