Messages

1

Mitel MiVoice Business/MCD/3300 / Re: SIP REFER w/ Exchange UM

« on: April 27, 2012, 01:56:05 PM »

I tried play on phone again, and it does work with the full 10 digit DID and 10 digit external numbers.

It doesn't work with 7 digit DID's (As defined in System Speed Calls).

2

Mitel MiVoice Business/MCD/3300 / Re: SIP REFER w/ Exchange UM

« on: April 27, 2012, 11:27:25 AM »

I looked at the trunk attributes of the PRI Trunk and the Exchange SIP; Neither are absorbing or inserting any digits.

I have an extension assigned and tried referencing the fax server that way but exchange server complained and said:

Error:
sip:555@faxserver.domain.local:5060;transport=udp does not match the required format "sip:FQDN:port;transport=TCP/UDP/TLS". Example: "sip:fax.contoso.com:5060;transport=TCP"

On the COS of the Exchange peer profile and PRI I have:
Public Network Access via DPNSS
Public Network to Public Network Connection Allowed
Set to YES

My PRI COS is set as Public Trunk YES,
while the COS on Exchange is Public Trunk NO,

Are there other COS settings I am overlooking to allow for external transfers?
Maybe Call Forwarding (External Destination)?

Thanks

Nick

3

Mitel MiVoice Business/MCD/3300 / Re: SIP REFER w/ Exchange UM

« on: April 27, 2012, 09:57:47 AM »

Play on phone only works with extensions.

4

Mitel MiVoice Business/MCD/3300 / Re: SIP REFER w/ Exchange UM

« on: April 24, 2012, 03:19:21 PM »

I did; I assigned an extension to the SIP fax server and tried that. It didn't work.

When doing that I did see that from a phone if a dialed that extension it would place a call through to the FAX Server.

5

Mitel MiVoice Business/MCD/3300 / Re: SIP REFER w/ Exchange UM

« on: April 24, 2012, 02:34:41 PM »

Hex editors are useful. 

6

Mitel MiVoice Business/MCD/3300 / Re: SIP REFER w/ Exchange UM

« on: April 23, 2012, 03:55:22 PM »

Exchange 2010 UM doesn't have native fax capabilities.

Instead it relies on a partner fax server that it redirects faxes to.

I can't send it directly as the fax calls all need to originate from the 3300 and be forwarded to exchange w/ DIVERSION headers (just like voice mail). 

The SIP fax server does dot have a number assigned to it, only a SIP URI that exchange is configured to transfer the calls to.  It is only configured as a SIP peer on the 3300.

Do I need to configure a number/ARS Route for it and then perform URI/Number translation for it?

I can send faxes to analog ports directly from the 3300, but Exchange 2010 UM requires a SIP fax server to handle the calls.

The main issue is why the 3300 will not forward the call to the URI in the SIP REFER field.

7

Mitel MiVoice Business/MCD/3300 / SIP REFER w/ Exchange UM

« on: April 23, 2012, 01:57:55 PM »

Hello,

I am trying to forward a call using using SIP REFER but am getting a 603 error message.

I do this when an incoming fax is sent to our Exchange UM server.

The exchange server then sends back a SIP REFER message to the ICP3300 to send the call to our SIP Fax Server.

However the 3300 responds with a 603 Decline failure message.

I have attached a packet trace of the entire conversation.

Any help on getting the 3300 to forward this call would be appreciated.

Nick

8

Mitel MiVoice Business/MCD/3300 / UM Exchange integration problem

« on: April 29, 2011, 05:10:37 PM »

Hello,

I have Exchange 2010 configured as voicemail but am having a problem.

Currently running:
Release 4.0 SP2
Active load 10.0.2.8

The problem occurs when someone sets there phone to DND, or forwards it to voicemail, or to an external phone number.
When they do that and an outside caller calls the exchange autoattendant and dials thier extension the call is immediatly disconnected.

I have tried using the SIP Peer Profile in Mitel TechGuide "Configure MCD 4.0 UR3 for use with Microsoft Exchange 2010" (SIP CoE 10-4940-00117).
See Attachment for that profile.

I have also tried to use SIP Profile from brantn's post in thread "UM Exchange integration question"

Here is my current Profile:

SIP Peer Profile Label:    Exchange   
Network Element:    Exchange   
 
Local Account Information
  Registration User Name:       
  Address Type:    FQDN: fqdn.domain.local   
 
 
Call Routing and Administration Options
  Interconnect Restriction:    1   
  Maximum Simultaneous Calls:    8   
  Outbound Proxy Server:       
  SMDR Tag:    0   
  Trunk Service:    21   
  Zone:    1   
  Alternate Destination Domain Enabled:    No   
  Alternate Destination Domain FQDN or IP Address:       
  Enable Special Re-invite Collision Handling:    No   
  Private SIP Trunk:    No   
  Route Call Using To Header:    No   
 
 
Calling Line ID Options
  Default CPN:       
  CPN Restriction:    No   
  Public Calling Party Number Passthrough:    No   
  Use Diverting Party Number as Calling Party Number:    No   
 
 
Authentication Options
  User Name:       
  Password:    *******     
  Confirm Password:    *******     
  Authentication Option for Incoming Calls:    No Authentication   
 
 
SDP Options
  Allow Peer To Use Multiple Active M-Lines:    No   
  Enable Mitel Proprietary SDP:    Yes   
  Force sending SDP in initial Invite message:    No   
  Force sending SDP in initial Invite - Early Answer:    Yes   
  NAT Keepalive:    Yes   
  Prevent the Use of IP Address 0.0.0.0 in SDP Messages:    Yes   
  Renegotiate SDP To Enforce Symmetric Codec:    No   
  Repeat SDP Answer If Duplicate Offer Is Received:    No   
  RTP Packetization Rate Override:    No   
  RTP Packetization Rate:    20ms   
  Special handling of Offers in 2XX responses (INVITE):    No   
  Suppress Use of SDP Inactive Media Streams:    No   
 
 
Signaling and Header Manipulation Options
  Session Timer:    0   
  Build Contact Using Request URI Address:    No   
  Disable Reliable Provisional Responses:    Yes   
  Enable sending '+' for E.164 numbers:    No   
  Ignore Incoming Loose Routing Indication:    No   
  Use P-Asserted Identity Header:    No   
  Use P-Preferred Identity Header:    No   
  Use Restricted Character Set For Authentication:    No   
  Use To Address in From Header on Outgoing Calls:    No   
 

Can anyone help me?

Thanks
Nick