[asterisk-users] Regarding fmtp parameters.

[SiM]

Hello All,
               I'am doing a video call between two Video Phones, and i see
that Asterisk is stripping the fmtp parameters for the h263 video line in
SDP.
For example a line similar to the below is stripped,

                 a=fmtp:xx CIF=4;QCIF=2;F=1;K=1

Asterisk is configured NOT to be present in the Media path (My version :
Asterisk 1.4.19.1 ).
I have the following enabled in my sip.conf.

canreinvite=yes
directrtpsetup=yes