Currently only the first audiohook of a specific source found will be moved. We should add the capability to move multiple audiohooks from a single source as well.
Note that in future, we need to move to a model where we can differentiate further - e.g. between en_US & en_UK
SIP over TCP
SIP over TLS
Better support of forking
VIA branch tag transaction checking
Transaction support
need a better return code here
need a better return code here
XXX Maybe we could do something with packets, like dial "0" for operator or something XXX
XXX Ick: jumping into an else statement??? XXX
XXX overwrites data ?
XXX watch out, leading whitespace ?
Move app_getcpeid into this module
Create a core layer so that app_voicemail does not require res_adsi to load
If you unload this module, chan_gtalk/jingle will be dead. How do we handle that?
If you have TLS, you can't unload this module. See bug #9738. This needs to be fixed, but the bug is in the unmantained Iksemel library
Make sure we don't destroy the call if we can't handle the re-invite. Nothing should be changed until we have processed the SDP and know that we can handle it.
check if this is not set earlier when setting up the PVT. If not maybe it should move there.
Fix the transfer() dialplan function so that a transfer may fail
In theory, we should hang around and wait for a reply, before returning to the dial plan here. Don't know really how that would affect the transfer() app or the pbx, but, well, to make this useful we should have a STATUS code on transfer().