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

Dan Austin Dan_Austin at Phoenix.com
Fri Dec 15 10:26:41 MST 2006


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


More information about the asterisk-users mailing list