[asterisk-dev] Status of SIP call-limit in 1.2 branch

[Andrea Spadaccini]

Ciao Leif,

> This probably more of an asterisk-users question, but I use the GROUP 
> () and GROUP_COUNT() functions quite successfully. I don't like to  
> use things that are channel specific (even though I only use SIP at  
> the moment... but that could change in the future).

Thanks, I'll go and read the docs for those functions.
I posted it to -dev because I've been digging the source code and I found there
the indications that I reported in the original message.

Thanks again, have a nice day!

-- 
Dott. Andrea Spadaccini
Multimedia Technologies Institute s.r.l.