[asterisk-dev] Finding the original sip_pvt

[Torbjörn Abrahamsson]

> No. You can however code AST_CONTROL frames with payloads and transmit
> over the bridge.
> >
> > When you talk about the fax code, which part do you mean? In chan_sip
> or
> > res_fax(?) or something else? There are a bunch of files regarding
> fax. And
> > regarding p->owner, when I did some testing it seemed to me like p
> and
> > p->owner->tech_pvt were the same. Have I missed something?
> In chan_sip.
> >
> > The setup looks like:
> > SIP1 -> AST -> SIP2
> 
> SIP1 -> AST <bridge> AST2 -> SIP2
> Each pvt has an ast_channel. The bridge joins them. There are functions
> to find the bridged channel if you check the API.
> 

OK, thanks. I will try to find the relevant parts. Just to be sure, when you
speak of a bridged channel above, you do not mean what happens when to legs
of a call are connected, ie a bridged call? Because as this is before the
200 OK is sent the call is not bridged.. Just making sure I don't go looking
for the wrong things... :)

// T