Asterisk Cisco CallManager Integration using SIP trunk

Here is a Sip debug

– Executing [576@office:1] Dial(“SIP/518-b4444e98”, “SIP/1157@BandGTrunk1”) in new stack
Audio is at 70.42.10.xxx port 15190
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 66.182.xxx.xxx:5060:
INVITE sip:1157@66.182.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 70.42.10.xxx:5060;branch=z9hG4bK18a93a45;rport
From: “Don Fletcher” sip:518@70.42.10.xxx;tag=as5d5bf772
To: sip:1157@66.182.xxx.xxx
Contact: sip:518@70.42.10.xxx
Call-ID: 0cebf1f54c6aeec3453005a0179448f6@70.42.10.xxx
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: “Don Fletcher” sip:518@70.42.10.xxx;privacy=off;screen=no
Date: Mon, 28 Jul 2008 21:09:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 15166 15166 IN IP4 70.42.10.xxx
s=session
c=IN IP4 70.42.10.xxx
t=0 0
m=audio 15190 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv



-- Called 1157@BandGTrunk1

cp005*CLI>
<— SIP read from 66.182.xxx.xxx:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 70.42.10.xxx:5060;branch=z9hG4bK18a93a45;rport
From: “Don Fletcher” sip:518@70.42.10.xxx;tag=as5d5bf772
To: sip:1157@66.182.xxx.xxx
Date: Mon, 28 Jul 2008 21:09:41 GMT
Call-ID: 0cebf1f54c6aeec3453005a0179448f6@70.42.10.xxx
CSeq: 102 INVITE
Allow-Events: presence
Content-Length: 0

<------------->
— (9 headers 0 lines) —
cp005*CLI>
<— SIP read from 66.182.xxx.xxx:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 70.42.10.xxx:5060;branch=z9hG4bK18a93a45;rport
From: “Don Fletcher” sip:518@70.42.10.xxx;tag=as5d5bf772
To: sip:1157@66.182.xxx.xxx;tag=a2aa5f02-9414-45b1-8f6c-997a7e8886df-23686195
Date: Mon, 28 Jul 2008 21:09:41 GMT
Call-ID: 0cebf1f54c6aeec3453005a0179448f6@70.42.10.xxx
CSeq: 102 INVITE
Allow-Events: presence
Reason: Q.850;cause=1
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Transmitting (NAT) to 66.182.xxx.xxx:5060:
ACK sip:1157@66.182.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 70.42.10.xxx:5060;branch=z9hG4bK18a93a45;rport
From: “Don Fletcher” sip:518@70.42.10.xxx;tag=as5d5bf772
To: sip:1157@66.182.xxx.xxx;tag=a2aa5f02-9414-45b1-8f6c-997a7e8886df-23686195
Contact: sip:518@70.42.10.xxx
Call-ID: 0cebf1f54c6aeec3453005a0179448f6@70.42.10.xxx
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: “Don Fletcher” sip:518@70.42.10.xxx;privacy=off;screen=no
Content-Length: 0


-- SIP/BandGTrunk1-08354320 is circuit-busy

== Everyone is busy/congested at this time (1:0/1/0)
Really destroying SIP dialog '0cebf1f54c6aeec3453005a0179448f6@70.42.10.xxx’ Method: INVITE

The exten from CCM can call into asterisk, but when I try to call into CCM from asterisk, I get that trace.

THE CCM is behind a NAT.
it is CCM 5.1.3
Asterisk 1.4.15

Any ideas on trouble-shooting would be appreciated.

Addendum :

The sip trace from the CCM shows :
From: ““ContactPoint”<6027149884>” <sip:“ContactPoint”<6027149884>@70.42.10.xxx>;tag=as173073bd
To: sip:66.182.xxx.xxx
Contact: <sip:“ContactPoint”<6027149884>@70.42.10.185>
Call-ID: 71ab9b3c3720920c40e82349455b6fc1@70.42.10.185
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 25 Jul 2008 18:48:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

When I dial into it, but shows :
To: sip:1157@70.42.10.xxx;tag=as4e3384bd
Call-ID: f5e9b580-88a11060-1-151e010a@EXT_BGBCCMS
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

when trying to dial from the CCM to asterisk. (which completes)

I think the problem is in the NAT configuraiton, but I’m not sure what to change on the nat.