I am using asterisk v1.2.0.
I have a problem regarding the use of ‘canreinvite’ and ‘t’/‘T’ in the dial command option.
I wish to limit the data and signalling stream proxying by PBX such that internal call will not be proxied while external be proxied.
I found out that in order to NOT let the signalling and data streams be proxied by asterisk, I need to use canreinvite=yes in the sip extensions and NOT put any ‘t’/‘T’ in the dial command option.
I tried it and it works. The PBX will reinvite the videophones and keep the streams away from the PBX.
BUT I can’t seem to use the PBX features (e.g. call transfer, call parking etc) anymore.
My dtmfmode is rfc2833
Am I doing it wrongly?