[asterisk-users] Early rtp bridge and reINVITE in 1.4b2

[H Quintana]

Hi,

I just test the "reinvite" feature in this version and
I realized that the SDP is changed in the early stages
of the SIP call and thus * is not sending the
reINVITEs.

Is there any way to disable the early rtp bridge but
still having the reINVITEs? (may be some parameter in
sip.conf ).  Let's say the device is behind a NAT
router and knows its public IP, if the RTP port
choosen by the NAT-router is not the same port the
device set in the SDP then the early bridge is not
going to work.

I'm not asterisk savvy, but may I can comment
something out in chan_sip.c

Any ideas? 

Thanks for your help,

Humberto

__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com