Asterisk 1.6 registration

Hi Guys,
I am not sure if the problem I have is related to asterisk 1.6.0.22 or trixbox 2.8.0.2. I have posted on trixbox’s forum but no one has responded for 2+ days. I am hoping someone might help …

I am trying to connect from my system to another asterisk system running on non-standard port 8891 instead of 5060. With asterisk 1.4 my registration string was
register=>username:password@TrunkName/extension
and the TrunkName definition had port=8891 and from-domain=host:8891 and it worked fine until I upgraded tp asterisk 1.6. Now I am forced to put the port number in the registration string as follows,
register=>username:password@TrunkName:8891/extension
This way I am able to register but the last optional parameter, the extension is not being passed to the remote and as a result the incoming call does not come to the defined context. I appears as anonymous call and gets dropped. On the remote side sip show peers shows extension logged in as extension/s rather than extension/extension. Here is some output,

<------------->
— (11 headers 0 lines) —
Responding to challenge, registration to domain/host name 10.100.1.95
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 10.100.1.95:8891:
REGISTER sip:10.100.1.95:8891 SIP/2.0
Via: SIP/2.0/UDP 10.100.1.50:5060;branch=z9hG4bK3930ed8e;rport
Max-Forwards: 70
From: sip:203@10.100.1.95;tag=as7e9dd901
To: sip:203@10.100.1.95
Call-ID: 1645eadd30db855942a848d135023b07@127.0.0.1
CSeq: 5303 REGISTER
User-Agent: Asterisk PBX 1.6.0.22-samy-r60
Authorization: Digest username=“203”, realm=“asterisk”, algorithm=MD5, uri=“sip:10.100.1.95:8891”, nonce=“4a7234a3”, response="b603c701d4b8d96cda3511211aaa35c3"
Expires: 120
Contact: sip:s@10.100.1.50
Event: registration
Content-Length: 0

<— SIP read from UDP://10.100.1.95:8891 —>
OPTIONS sip:s@10.100.1.50 SIP/2.0
Via: SIP/2.0/UDP 10.100.1.95:8891;branch=z9hG4bK7f428fa0;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@10.100.1.95:8891;tag=as543f0e79
To: sip:s@10.100.1.50
Contact: sip:Unknown@10.100.1.95:8891
Call-ID: 37b24a8550681ca27addd9e07355f231@10.100.1.95
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Date: Thu, 25 Feb 2010 01:53:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Looking for s in from-sip-external (domain 10.100.1.50)

<— Transmitting (no NAT) to 10.100.1.95:8891 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.100.1.95:8891;branch=z9hG4bK7f428fa0;received=10.100.1.95;rport=8891
From: “Unknown” sip:Unknown@10.100.1.95:8891;tag=as543f0e79
To: sip:s@10.100.1.50;tag=as4b45d521
Call-ID: 37b24a8550681ca27addd9e07355f231@10.100.1.95
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.22-samy-r60
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:10.100.1.50
Accept: application/sdp
Content-Length: 0

Local IP is 10.100.1.50 and remote is 10.100.1.95. Notice the To: sip:s@10.100.1.50 and not 203.

I have an extension on another asterisk server which is running on standard port 5060 where I do not have any issues. Do you think it is bug is asterisk or in trixbox? Any help will be appreciated.

Regards,

Sohail.