Hello Everyone,
I have some trouble configuring SIP trunk between CUCM 9.1 and Elastix with NAT. i don't have any config about Elastix, my part to config is CUCM.
SIP Trunk Security Profile
All "Accept" are checked, Outgoing Protocol UDP and port is 5060
SIP Profile
The first section all default (SIP Information)
The second section all default (Timers)
Third section Local RSVP unchecked, Early offer Support... Checked (The rest default)
SIP options Checked
Trunk
MRGL Selected from LAN (The Router route to elastix dont have any DSP)
MTP Checked, Retrive Video call as Audio Checked, MTP Codec g711ulaw, DTMF no preference
Nat Config
ip nat inside source static X.X.X.X(Local) X.X.X.X(Global Local) extendable
ip nat outside source static X.X.X.X(Gobal Local) X.X.X.X (Local) extendable
ip nat inside source static upd X.X.X.X(Local) 5060 X.X.X.X(Gobal local) 5060 extendable
i have a pool NAT address with not PAT, also permit all UDP and TCP to do NAT to all the traffic with
access-list 10 permit tcp any any
access-list 10 permit udp any any
The Problem
I see on the Phone that the call connected imediate but not sound (two-Way) (Only if i only put the nat with 5060), When i put the other two NAT command i heard the recording saying "the extension are busy please try again" but i dont now if the my CUCM or the Elastix saying the messages.
The phone can Reach the Router that do NAT also the CUCM and Vice-versa
I do the trace and receive a error 404 but i don't know what to do
