Call from Cisco CUCM don't pass to PBX

Hi

my installation look like

Asterisk Server
VMWare server
OS: Ubuntu 9.10 full uptodate
FreePBX: 2.8.0.2 just updated from 2.7.0.5
extension: 35XX

Physical server
OS: Ubuntu 9.10 not uptodate
FreePBX: 2.7.0.1
extension: 15XX

Cisco environnement
CUCM: 7.1.3

First of all,
when I have past to 2.7.0.5 the extension_constum.conf has in problem. I’ve look in the log end I fund this problem. The _XXXX as not longer supported, when I have change all _XXXX to XXXX asterisk wase happy end do no error.

My problem is,
When a cisco phone call a extension 35XX the callmanager tell to me "your call cannot be forwarded.
In my console asterisk (sudo asterisk -R; core set verbose 9; core set debug9) I see

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5

In the /var/log/asterisk/full
I see

[Sep 15 09:51:57] DEBUG[1666] acl.c: Found IP address for this socket
[Sep 15 09:51:57] DEBUG[1666] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.54.20.98:5060
[Sep 15 09:51:57] VERBOSE[1666] netsock.c: == Using SIP RTP TOS bits 184
[Sep 15 09:51:57] VERBOSE[1666] netsock.c: == Using SIP RTP CoS mark 5
[Sep 15 09:51:57] DEBUG[1666] chan_sip.c: Setting NAT on RTP to Off
[Sep 15 09:51:57] DEBUG[1666] chan_sip.c: Allocating new SIP dialog for [email protected] - INVITE (With RTP)
[Sep 15 09:51:57] DEBUG[1666] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Sep 15 09:51:57] DEBUG[1666] chan_sip.c: Setting NAT on RTP to Off
[Sep 15 09:51:57] DEBUG[1666] chan_sip.c: Got unsupported a:fmtp:101 0-15 in SDP offer
[Sep 15 09:51:57] DEBUG[1666] chan_sip.c: We’re settling with these formats: 0x4 (ulaw)
[Sep 15 09:51:57] DEBUG[1666] chan_sip.c: Checking SIP call limits for device
[Sep 15 09:51:57] DEBUG[1666] chan_sip.c: Updating call counter for incoming call
[Sep 15 09:51:57] DEBUG[1666] chan_sip.c: Trying to put ‘SIP/2.0 404’ onto UDP socket destined for 10.74.20.5:5060
[Sep 15 09:51:57] NOTICE[1666] chan_sip.c: Call from ‘callmanager-sub’ to extension ‘3554’ rejected because extension not found.
[Sep 15 09:51:57] DEBUG[1666] chan_sip.c: Updating call counter for incoming call
[Sep 15 09:51:57] DEBUG[1666] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Sep 15 09:51:57] DEBUG[1666] chan_sip.c: Stopping retransmission on ‘[email protected]’ of Response 101: Match Found
[Sep 15 09:51:57] DEBUG[1666] chan_sip.c: Destroying SIP dialog [email protected]

If I call from a Aastra phone on my PBX server to a cisco phone, the call is forwarded.

What the diffence between the SIP Trunk processing to not work anymore.

But my other PBX server is working find.