[Asterisk-Users] Changing REINVITE status of the channel dynamically

[Álvaro Palma]

I've an Asterisk server running in my office, which forwards all
long distance calls to a third party SIP service using an extension rule:

exten => _1XX0.,1,Dial(SIP/{EXTEN:4}@external_sip_server.com)

(1XX0 is the international calls rule for Chile)

Also, in my sip.conf, I've defined canreinvite=yes to decrease the 
network load to the server caused by the RTP.

However, the external sip server seems to be buggy, because the 
REINVITE's against it only works for certain routes, and in others, it 
simply hang up the calls. Since I don't have control over that remote 
service (and I already inform them about this problem), I'd like to know 
if it's possible to set the REINVITE on or off dynamically, based on the 
extension being dialed. I don't like very much the option to completely 
disable the REINVITE's in my network (formed by a central, and a lot of 
offices connected to it by not too fast links, so the network usage is 
an issue)

Thanks a lot for your help.

-- 
Atly.
Alvaro Palma