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

[Johansson Olle E]

6 dec 2007 kl. 15.29 skrev 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.

Asterisk is a multiprotocol PBX, not a SIP proxy. We never just relay
something :-)

/O