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Author Topic: UM Exchange integration problem  (Read 4891 times)

Offline nickp

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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
« Last Edit: April 29, 2011, 05:18:07 PM by nickp »
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Offline brantn

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Re: UM Exchange integration problem
« Reply #1 on: April 29, 2011, 06:45:48 PM »
Not more SIP problems  ;D

I would...

1.) Start with a trace of the call probably monitor the port of the exchange server and see what is happening.
2.) The DND issues that I have seen and is very similar to your issue can be resolved by a couple means...
     A.) Make sure DPNSS/QSIG Diversion Enabled is no in system options.
     B.) Make sure public network to public network is enabled in the trunks COS.
     C.) Most important of all set Suppress Simulated CCM after ISDN progress to no in the trunks COS.

This should fix it if not remember SIP = Stiill In Progress.
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Offline jwebb215

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Re: UM Exchange integration problem
« Reply #2 on: August 05, 2011, 12:02:06 PM »
I am experiencing the same issue where transfers to Exchange appear to be dropping the calls when DND is on (We've discovered that it's actually one way audio and not a dropped call though).   When we perform a Wireshark on one call with DND on and one with DND off the calls look quite different:

Without DND:

|Time     | 10.1.1.31                             | 10.1.1.33                             |
|         |                   | 10.1.1.86         |                   
|58.343   |         INVITE    |                   |                   |SIP From: "Outside Caller" <sip:1112223333@10.1.1.31 To:<sip:868687@10.1.1.86
|         |(2296)   ------------------>  (5065)   |                   |
|58.344   |         100 Trying|                   |                   |SIP Status
|         |(2296)   <------------------  (5065)   |                   |
|58.522   |         180 Ringing SDP (g723 g711U g711A REDRTPType-9...elephone-eventRTP) SDP (g723 g711U g711A REDRTPType-97 telephone-eventRTP)          |                   |SIP Status
|         |(2296)   <------------------  (5065)   |                   |
|58.522   |         180 OK    |                   |                   |SIP Status
|         |(2296)   <------------------  (5065)   |                   |
|58.744   |         ACK SDP (g711U telephone-eventRTPType-101)          |                   |SIP Request
|         |(2296)   ------------------>  (5065)   |                   |
|58.920   |                   |         RTP (g711U)                   |RTP Num packets:157  Duration:4.678s SSRC:0x1420E6
|         |                   |(63360)  <------------------  (50376)  |
|59.082   |                   |         RTP (g711U)                   |RTP Num packets:226  Duration:4.618s SSRC:0x311BA64F
|         |                   |(63360)  ------------------>  (50376)  |
|63.660   |         BYE       |                   |                   |SIP Request
|         |(2296)   ------------------>  (5065)   |                   |
|63.661   |         200 OK    |                   |                   |SIP Status
|         |(2296)   <------------------  (5065)   |                   |


And with DND on:

|Time     | 10.1.1.31                             | 10.1.1.33                             |
|         |                   | 10.1.1.86         |                   
|15.551   |         INVITE    |                   |                   |SIP From: "Inside Caller" <sip:2738@10.1.1.31 To:<sip:868687@10.1.1.86
|         |(2296)   ------------------>  (5065)   |                   |
|15.552   |         100 Trying|                   |                   |SIP Status
|         |(2296)   <------------------  (5065)   |                   |
|15.732   |         180 Ringing SDP (g723 g711U g711A REDRTPType-9...elephone-eventRTP) SDP (g723 g711U g711A REDRTPType-97 telephone-eventRTP)          |                   |SIP Status
|         |(2296)   <------------------  (5065)   |                   |
|15.732   |         180 OK    |                   |                   |SIP Status
|         |(2296)   <------------------  (5065)   |                   |
|15.848   |         ACK SDP (g711U telephone-eventRTPType-101)          |                   |SIP Request
|         |(2296)   ------------------>  (5065)   |                   |
|15.970   |                   |         RTP (g711U)                   |RTP Num packets:18  Duration:0.509s SSRC:0x22B351
|         |                   |(28800)  <------------------  (50364)  |
|16.222   |                   |         RTP (g711U)                   |RTP Num packets:338  Duration:6.852s SSRC:0x150F271B
|         |                   |(28800)  ------------------>  (50364)  |
|16.795   |         INVITE    |                   |                   |SIP From: "Inside Caller" <sip:2738@10.1.1.31 To:<sip:868687@10.1.1.86
|         |(2296)   ------------------>  (5065)   |                   |
|16.795   |         100 Trying|                   |                   |SIP Status
|         |(2296)   <------------------  (5065)   |                   |
|16.932   |         200 OK SDP (g723 g711U g711A REDRTPType-97 tel...one-eventRTP)          |                   |SIP Status
|         |(2296)   <------------------  (5065)   |                   |
|17.009   |         ACK SDP (g711U telephone-eventRTPType-101)          |                   |SIP Request
|         |(2296)   ------------------>  (5065)   |                   |
|17.030   |                   |         RTP (g711U)                   |RTP Num packets:200  Duration:5.968s SSRC:0x26D930
|         |                   |(28800)  <------------------  (50366)  |
|23.085   |         BYE       |                   |                   |SIP Request
|         |(2296)   ------------------>  (5065)   |                   |
|23.086   |         200 OK    |                   |                   |SIP Status
|         |(2296)   <------------------  (5065)   |                   |

We believe that this also happens in a busy condition on the users' extension.  Any ideas or pointers are appreciated!
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Offline brantn

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Re: UM Exchange integration problem
« Reply #3 on: August 05, 2011, 01:31:35 PM »
That is correct it is an issue with signalling check step 2 in my post it resolves the issue.
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