[asterisk-dev] SIP DTMF INFO Singal-Update?

[Victor Toofic]

El Thu, Dec 06 de 2007 a las 09:53 +0000, Johansson Olle E comentaba:
> What does it mean? Any documentation anywhere on how to implement it?
> Is the previous duration of 2600 for signal 1 cut off to 305?

The only thing I've found is in this document:

  http://www.africamovies.co.uk/AgentForm/Agent.pdf

in "3.3.2.3.1.3 Alternatives to RFC 2833" in the section about Sonus "(c) Sonus".

Im not sure how that could be implemented. In the case of SIP-to-SIP my
first thought was that it could be simply relayed, but since I dont know
the Asterisk's internals I dont know if that is possible or correct.

--
Greetings..
Víctor Toofic