Dan Austin
2006-Dec-15 10:26 UTC
[asterisk-users] Cisco Call Manager 4.0 to Asterisk, Anyone haveSIP Reinvite working?
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