Lets start with your questions..This is happening because CUBE doesnt select dial-peers based on protocols, dial-peers are selected based on matching called or calling numbers. In this scenario you have two matching dial-peers with incoming called number .T, CUBE will always use the dial-peer with the highest tag in this case. In this case that is dial-peer 1.
If you want to have two dial-peers for h323 and sip, you will need to define more specific patterns to match on your incoming called number..
eg
dial-peer voice 2 voip
incoming called number 900T
session protocl sipv2
dtmf-relay rtp-nte
no vad
On the next question...
From your call flow, this is what your rtp connections should looke like
IPphone-----rtp--MTP(cucm)---rtp---CUBE-------rtp---ipphone(on ccme)
Your CUBE has two ip addresses...one is 142.1.64.254 and the other is 10.11.11.2
The interface defined for your sip trunk is 142.1.64.254 on cucm and the interface closes to ccme is 10.11.11.2. This is why you are seeing both addresses. I dont see a problem with this..
For the CCME, the reason why you dont see the ip address of the ip phone is down to DTMF megotiation and thats why the call is failing...
Lets look at the traces..
1.When CUCM sent invite to sip it advertised rtp-nte (rtpmap:101)
Received:
INVITE sip:900861062286001@142.1.64.254:5060 SIP/2.0
Date: Fri, 01 Mar 2002 01:16:28 GMT
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
From: "HQ Phone 1" <>;tag=0140a88e-20ea-4ea4-98ef-fc4a3f05c677-33234280
Allow-Events: presence, kpml
P-Asserted-Identity: "HQ Phone 1" <>
Supported: timer,resource-priority,replaces
Min-SE: 1800
Remote-Party-ID: "HQ Phone 1" <>;party=calling;screen=yes;privacy=off
Content-Length: 214
User-Agent: Cisco-CUCM7.0
To: <900861062286001>
Contact: <>
Expires: 180
Content-Type: application/sdp
Call-ID: f377c380-c7e1d66c-e-b40648e@142.100.64.11
Via: SIP/2.0/TCP 142.100.64.11:5060;branch=z9hG4bK1735b534a6
CSeq: 101 INVITE
Session-Expires: 1800
Max-Forwards: 70
v=0
o=CiscoSystemsCCM-SIP 2000 1 IN IP4 142.100.64.11
s=SIP Call
c=IN IP4 142.100.64.11
t=0 0
m=audio 24602 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
2. When CUBE sent out its INVITE to CCME, it also advertised rtp-nte
v=0
o=CiscoSystemsSIP-GW-UserAgent 5553 8018 IN IP4 10.11.11.2
s=SIP Call
c=IN IP4 10.11.11.2
t=0 0
m=audio 18168 RTP/AVP 0 101 19
c=IN IP4 10.11.11.2
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
3. However when we received 200 OK from CCME, we didnt get any dtmf advertised..Hence the reason why connection parameter C=10.11.11.1, points to the ip address of ccme not the phone, because it was attempting to invoke an MTP to address dtmf mismtach issue..
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.11.11.2:5060;branch=z9hG4bK142581
From: "HQ Phone 1" <>;tag=460334-7A8
To: <861062286001>;tag=45FB48-26F8
Date: Fri, 01 Mar 2002 01:16:28 GMT
Call-ID: CA7760AE-2BE811D6-8026EFAB-405440B5@10.11.11.2
Timestamp: 1014945388
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Supported: replaces
Allow-Events: telephone-event
Contact: <861062286001>
Content-Type: application/sdp
Content-Length: 208
v=0
o=CiscoSystemsSIP-GW-UserAgent 9854 850 IN IP4 10.11.11.1
s=SIP Call
c=IN IP4 10.11.11.1
t=0 0
m=audio 16462 RTP/AVP 0 19
c=IN IP4 10.11.11.1
a=rtpmap:0 PCMU/8000
a=rtpmap:19 CN/8000
a=ptime:20
To resolve this issue..please do the ff:
1. Ensure you define your inbound dial-peer correctly as suggested above. Then makse sure that dtmf-relay rtp-nte is advertised. You may remove dial-peer 1 for the sake of testing or use specific incoming called number I suggested
2. Your phones on CCME are SCCP phones and they dont support rtp-nte hence you will need to configure an MTP registered with your CCME or alternatively add the ff config to your ccme inbound dial-peer
dial-peer voice x voip
session protcol sipv2
dtmf-relay rtp-nte digit-drop sip-kpml
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