[asterisk-users] Cisco Call Manager 4.0 to Asterisk, Anyone haveSIP Reinvite working?

[Dan Austin]

Pavel wrote:
> I think, callmanager needs media termination point (mtp) for 
> sip trunk, so rtp stream will always go through callmanager...
That is true for CCM 4.X, so SIP works with CCM 4.X, but is
far from ideal.  As of CCM 5.X  added RFC 2833 support to the
SCCP endpoints, so a MTP is not required and your not stuck with
just ULAW for the codec....

Now whether improved SIP support is enough to justify the big
jump to 5.x (Windows to Linux), is another issue...

Dan