[asterisk-dev] Finding the original sip_pvt

[Olle E. Johansson]

16 maj 2011 kl. 13.06 skrev Torbjörn Abrahamsson:

>  
> 
> Now if I have missed the obvious (and correct) way to do this, please enlighten me.. Maybe there is somewhere the new pvt is created, and some of its variables gets populated?
> 
If you look into the fax part of the code, you can see that we peek into the bridged channel to the p->owner and check the pvt of that one - you need to cross over the bridge (hold on to the rails) and find the other channel. Don't assume that you always have one either.

In other cases, we use channel variables so that the incoming channel set a variable that begins with one or two underscores, these will be inherited to the new channel.

/O