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**Solved** Gatekeeper - show voip rtp connections

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**EDIT**

I managed to solve the problem. i wasn't using CUBE on the HQ or PST routers. 

http://ieoc.com/forums/p/24910/212690.aspx#212690 

 

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I'm attempting some 'advanced gatekeeper' work. I've created a Gatekeeper on R1 at HQ. I have registered R3 at SC into the GK. Calls are working correctly. 

I've then attempted to direct calls to a third-party gatekeeper zone. I did this by setting up a H323 interface on the PSTN router and a VOIP dial-peer to match the incoming call: 

 

interface Loopback2

 description **Address for SIP/H323 Trunk

 ip address 167.26.1.253 255.255.255.255

 h323-gateway voip interface

 h323-gateway voip id R1 ipaddr 162.102.64.254 1719

 h323-gateway voip h323-id RGGK

 h323-gateway voip bind srcaddr 167.26.1.253

end

dial-peer voice 4020 voip

 incoming called-number 01144.

The incoming GK calls terminate on the PSTN's '911 line' 
Next I configured the HQ router: 
interface FastEthernet0/0.102
 encapsulation dot1Q 102
 ip address 162.102.64.254 255.255.255.0
 ip helper-address 162.1.64.254
 h323-gateway voip interface
 h323-gateway voip id R1 ipaddr 167.26.1.253 1719
 h323-gateway voip h323-id HQ
 h323-gateway voip bind srcaddr 162.102.64.254
end
I updated the GK config: 
gatekeeper
 zone local GK cisco.com 162.102.64.254
 zone local R1 cisco.com
 zone remote RGGK cisco.com 167.26.1.253 1719 outvia R1
 zone prefix RGGK 01144*
 no shutdown
I added a GK Route List and 901144! Route Pattern to CUCM. Calls are connecting correctly with two way audio. However, when I execute a 'show voip rtp connections command on R1, it says there are no active calls: 
R1#sh voip rtp conn
No active connections found
However, the PSTN router does detect an active call:
PSTN#sh voip rtp connections
VoIP RTP active connections :
No. CallId     dstCallId  LocalRTP RmtRTP LocalIP         RemoteIP
1   154        155        18916    22888  167.26.1.253    162.102.64.10 (HQ PH1 IP address)
Found 1 active RTP connections
Is this expected behaviour?
The guide I'm using states that R1 should have two RTP streams - both terminating on the voice vlan IP (162.102.64.254). 

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