Exchange UM 403 on REFER from Exchange->Brekeke

[dtosi]

1. Brekeke Product Name and version: Brekeke PBX, Version 2.2.6.2

2. Java version: Java 6 Update 10

3. OS type and the version: Windows 2003 Server R2

4. UA (phone), gateway or other hardware/software involved: Linksys SPA942, ITSPs Gafachi and QuantumVoice, Exchange 2007 UM

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html : Pattern 8

6. Your problem:
I am attempting to connect Brekeke to my Exchange 2007 server to use their swelt speech-enabled auto attendants as well as their unified messaging features. I followed the directions to connect to OCS from Brekeke (http://www.brekeke-sip.com/wiki/index.php?telepark.wiki=02416f48) and I can call into Exchange 2007 UM just great. I can even use the Outlook 'Play on Phone' feature which issues an INVITE command and that rings though to my phone just great. The problem comes when I try to transfer to an extension from the auto-attendant I get a 403 error. Here's the SIP command that Exchange 2007 UM issues:

[taitan]

Do you have a user ("NNNNNNNNN" in this case) in the Brekeke PBX?
I think you need to create a user in the user setting of the PBX.

[dtosi]

Thanks for the reply. Unfortunately NNNNNNNNN is the inbound phone number. In this case it's my cell phone. Any other ideas? I tried to create a rule that turned auth off but that didn't seem to do the trick.

Thanks-
David Tosi

[taitan]

Is 508 a PBX user? and is it registered in the SIP Server?

[dtosi]

Taitan-

Again, thanks for looking at this. I guess I should have defined the roles in the first post.

- 508 is a registered extension in Brekeke
- 400 is the Exchange UM Auto Attendant
- 192.168.1.42 is the IP Address of the Exchange Server
- 192.168.1.50 is the IP Address of the Brekeke Server
- NNNNNNNNN is the inbound phone number
- sip.gafachi.com is the ITSP provider I am testing with

Thanks again and I hope this helps!
-David Tosi

[james]

Is the Brekeke SIP Server working on the port 4832? (Is it the SIP SIP Server's listening port??)

Do you have the user "508" in your Brekeke PBX?

[dtosi]

Brekeke is configured for the default port (5060). User 508 is configured both in the PBX as a User and in the SIP server under User Authentication.

Thanks!
-David Tosi

[taitan]

dtosi,

Can you capture packets on the server PC?
Let me know who returned "403 Forbidden" response packet.

Can you make a call to "508" from the inbound phone without call transfer?

[dtosi]

Boy George I think I've figured it out. To fix the authentication problem I needed to add the Exchange Auto-Attendant (in my case 400) as an authenticated user. Also, I needed to add two more dialing rules to get everything working correctly.

Additional Rule #1: This is for forwarding to VM. You need to set each user's VoiceMail extension for 77{EXT} to work correctly. Note, that it adds the Diversion header needed for Exchange UM to correctly forward to VM. Also, for whatever reason Brekeke loses the Diversion header on the 3rd INVITE, so I had to add the extension to the To line. Note that 401 is the Exchange Pilot number.

Matching Patterns

Code: $port=15062 $localhost=true $registered=false $outbound=false $request=^INVITE $getUri( From )=!_ocs To=sip:77(.{3})

Deploy Patterns

Code: $transport=TCP $auth=false &net.sip.transport.follow.request=true To=sip:401@192.168.1.42;tag=%1 Diversion=<tel:%1>;reason=user-busy;screen=no;privacy=off

Additional Rule #2: This is a workaround for the problem with the lost Diversion header on the 3rd INVITE that Brekeke sends to Exchange. Anyways, it grabs the extension from the header's tag. Seems to work great.

Matching Patterns

Code: $request=^INVITE.+tag\=(.{3}) To=sip:401@

Deploy Patterns

Code: $transport=TCP $auth=false &net.sip.transport.follow.request=true Diversion=<tel:%1>;reason=user-busy;screen=no;privacy=off

I will have to test more but I think I am well on my way. Thanks for everybody help!

-David Tosi

[taitan]

Wow...