Author Topic: Problem Integrating Exchange 2013 UM with Mitel vMCD with MiCollab/MBG  (Read 4552 times)

Offline StanO

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Good Day:

I'm in the process of upgrading our Exchange environment to the 2013 version and have encountered problems getting Unified Messaging to work fully.  I believe the problem lies in the SIP configuration on the vMCD (v 7.1PR1) and or the MBG (MiCollab 6.0SP2 in Server Only configuration).  I believe the problem lies in the SIP configuration because Exchange UM (voicemail) functions properly when calls originate from within our environment.  When attempting to call the auto-attendant or allowing a call originating from outside to ring unanswered, however, I hear only silence.

We are using a SIP provider for our external telephone service and utilize the MBG as an external proxy for the vMCD.  Our Exchange 2007 environment is still running in co-existence with the new Exchange 2013 environment.  There are no issues with voicemail from external sources to exchange mailboxes still hosted on the Exchange 2007 servers.  The SIP Peer Profile for the Exchange 2013 server is configured per the recommendations in the "Configure the MCD 6.0 for use with Microsoft Exchange 2013 UM  (SIP CoE 13-4940-00274)" Technote.  The SIP Peer Profile and the COS definition for the Exchange 2007 and 2013 servers are similar, though slightly different due to adherence to the technote.  Turning up the verbosity on the Exchange UM logs appears to indicate the call is being received and answered by Exchange.

Any help that can be provided would be greatly appreciated - It's been about 2 weeks of head scratching to this point.  I can gather packet traces if that will be helpful, but first wanted to get this conversation started.

Thank you,

Stan

The SIP Peer Profile for one of the Exchange 2013 servers (two servers, both configured the same) is as follows.

The SIP Peer Profile for the
SIP Peer Profile Label:  Ex13UM-01
Network Element:  c38001e0-5665-11e5-8900-08000f217200
Registration User Name: 
Address Type:  IP Address: 172.16.0.10
Interconnect Restriction:  1
Maximum Simultaneous Calls:  4
Minimum Reserved Call Licenses:  4
Outbound Proxy Server: 
SMDR Tag:  9998
Trunk Service:  3
Zone:  2
User Name: 
Password: 
Confirm Password: 
Authentication Option for Incoming Calls:  No Authentication
Subscription User Name: 
Subscription Password: 
Subscription Confirm Password: 
Alternate Destination Domain Enabled:  No
Alternate Destination Domain FQDN or IP Address: 
Enable Special Re-invite Collision Handling:  No
Only Allow Outgoing Calls:  No
Private SIP Trunk:  No
Reject Incoming Anonymous Calls:  No
Route Call Using P-Called-Party-ID (if present):  Yes
Route Call Using To Header:  No
Default CPN: 
Default CPN Name: 
CPN Restriction:  No
Public Calling Party Number Passthrough:  No
Strip PNI:  No
Use Diverting Party Number as Calling Party Number:  No
Use Original Calling Party Number If Available:  No
Allow Peer To Use Multiple Active M-Lines:  Yes
Allow Using UPDATE For Early Media Renegotiation:  Yes
Avoid Signaling Hold to the Peer:  Yes
AVP Only Peer:  Yes
Enable Mitel Proprietary SDP:  No
Force sending SDP in initial Invite message:  No
Force sending SDP in initial Invite - Early Answer:  No
Ignore SDP Answers in Provisional Responses:  No
Limit to one Offer/Answer per INVITE:  No
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
Restrict Audio Codec:  No Restriction
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
Trunk Group Label: 
Allow Display Update:  No
Build Contact Using Request URI Address:  No
De-register Using Contact Address not *:  No
Disable Reliable Provisional Responses:  No
Disable Use of User-Agent and Server Headers:  No
E.164: Enable sending '+':  No
E.164: Add '+' if digit length > N digits:  0
E.164: Do not add '+' to Emergency Called Party:  No
E.164: Do not add '+' to Called Party:  No
Force Max-Forward: 70 on Outgoing Calls:  No
If TLS use 'sips:' Scheme:  No
Ignore Incoming Loose Routing Indication:  No
Multilingual Name Display:  No
Only use SDP to decide 180 or 183:  No
Prefer From Header for Caller ID:  No
Require Reliable Provisional Responses on Outgoing Calls:  No
Signal Privacy (if enabled) on Emergency Calls:  No
Suppress Redirection Headers:  No
Use Fixed Retry Time for 491:  No
Use Privacy: none:  No
Use P-Asserted Identity Header:  Yes
Use P-Asserted Identity for Billing:  No
Use P-Call-Leg-ID 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
Use user=phone:  No
Keep-Alive (OPTIONS) Period:  120
Registration Period:  3600
Registration Period Refresh (%):  50
Registration Maximum Timeout:  90
Session Timer:  90
Session Timer: Local as Refresher:  No
Subscription Period:  3600
Subscription Period Minimum:  300
Subscription Period Refresh (%):  80
Invite Ringing Response Timer:  0
Allow Inc Subscriptions for Local Digit Monitoring:  No
Allow Out Subscriptions for Remote Digit Monitoring:  No
Force Out Subscriptions for Remote Digit Monitoring:  No
Request Outbound Proxy to Handle Out Subscriptions:  No
KPML Transport:  default
KPML Port:  0
Creator: 
Date Created: 
Created with Version: 
Service Provider: 
Vendor Notes: 

The CoS Options associated with the trunk defined for routing calls to voicemail are:
Class Of Service Number:  7
Comment:  Voicemail
ACD Agent Behavior on No Answer:  Logout
ACD Agent No Answer Timer:  15
ACD Make Busy on Login:  No
ACD Silent Monitor Accept:  No
ACD Silent Monitor Accept Monitoring Non-Prime Lines:  No
ACD Silent Monitor Allowed:  No
ACD Silent Monitor Notification:  No
Follow 2nd Alternate Reroute for Recall to Busy ACD Agent:  No
Work Timer:  0
Call Announce Line:  No
Off-Hook Voice Announce Allowed:  No
Handsfree AnswerBack Allowed:  No
Busy Override Security:  Yes
Disable Executive Busy Override Tone:  No
Executive Busy Override:  No
Busy Tone Timer:  30
Dialing Conflict Timer:  3
First Digit Timer:  15
Inter Digit Timer:  10
Lockout Timer:  45
Call Duration:  10
Call Duration Forced Cleardown Timer:  0
Enable Call Duration Limit on External Calls:  No
Enable Call Duration Limit on Internal Calls:  No
Call Forward - Delay:  0
Call Forward No Answer Timer:  15
Call Forward Override:  No
Call Forwarding (External Destination):  No
Call Forwarding (Internal Destination):  Yes
Call Forwarding Accept:  Yes
Call Reroute after CFFM to Busy Destination:  No
Call Forwarding Reminder Ring (CFFM and CFIAH only):  No
Disable Call Reroute Chaining On Diversion:  No
Group Call Forward Follow Me Accept:  No
Group Call Forward Follow Me Allow:  No
Third Party Call Forward Follow Me Accept:  No
Third Party Call Forward Follow Me Allow:  No
Use Held Party Device for Call Re-routing:  Yes
Call Hold:  Yes
Call Hold - Retrieve with Hold Key:  No
Call Hold Remote Retrieve:  Yes
Call Hold Timer:  30
Local Music On Hold source:  No
Music on Hold on Transfer:  No
Use Called Party Call Hold Timer:  No
Call Park Timer:  180
Call Park-Allowed To Park:  No
Allow Directed Call Pickup Of Attendant Call:  No
Call Pickup Dialed Accept:  Yes
Call Pickup Directed Accept:  Yes
Call Privacy:  No
Calling Party Name Substitution:  No
Name Suppression on outgoing Trunk Call:  No
Privacy Released:  No
Public Network Identity Provided:  No
Call Waiting Swap:  No
ONS CLASS/CLIP: Visual Call Waiting:  Yes
Auto Campon Timer:  10
Campon Recall Timer:  10
Direct Voice Call - Accept:  No
Direct Voice Call - Allow:  No
Direct Voice Call - Maximize Volume:  No
After Answer Display Time: 
Calling Name Display - Internal - ONS:  Yes
Calling Number Display - Internal - ONS:  Yes
Display ANI/DNIS/ISDN Calling/Called Number:  No
Display ANI/ISDN Calling Number Only:  No
Display Caller ID on multicall/keylines:  No
Display Caller ID On Multicall/Keylines Timer:  5
Display Caller ID On Single Line Displays For Forwarded Calls:  No
Display Dialed Digits during Outgoing Calls:  No
Display DNIS/Called Number Before Digit Modification:  No
Display Held Call ID on Transfer:  No
Display Transfer Destination on Recall:  No
Hot Desk External User - Display Internal Calling ID:  No
Maintain Ringing Party During Recall:  No
Non-Prime Public Network Identity:  No
Originator's Display Update In Call Forwarding/Rerouting:  No
Suppress Delivery of Caller ID Display between Sets:  No
Suppress Delivery of Caller ID Display between Sets - Override:  No
Suppress Display Of Account Code Numbers:  No
Suppress Redial Display:  No
Campon Tone Security:  Yes
External Trunk Standard Ringback:  No
Fax Capable:  No
Return Disconnect Tone When Far End Party Clears:  No
HCI/CTI/TAPI Call Control Allowed:  No
HCI/CTI/TAPI Monitor Allowed:  No
Green BLF Lamp for Logged in Hotdesk User:  No
Hot Desk External User - Allow Mid-Call Features:  Yes
Hot Desk External User - Answer Confirmation:  Yes
Hot Desk External User - Dial Tone on Call Complete :  Yes
Hot Desk External User - Permanent Login:  Yes
Hot Desk External User - Remote MWI Enable Feature Access Code: 
Hot Desk External User - Remote MWI Disable Feature Access Code: 
Hot Desk Login Accept:  No
Hot Desk Remote Logout Enabled:  No
Backlighting - Enabled:  Yes
Clear All Features Remote:  No
Force Device Busy If Any Line In Use:  No
Handset Volume Adjustment Saved:  No
Head Set Switch Mute:  No
Multi-Color LED Support - Disable:  No
Phone Lock:  No
Reseize Timer:  180
Timed Reminder Allowed:  Yes
User Inactivity Timer:  0
Group Page Accept:  No
Group Page Allow:  No
Loudspeaker Pager Equivalent Zone Override Security:  No
Loudspeaker Pager Override:  Yes
Pager Access All Zones:  Yes
Pager Access Individual Zones:  No
PC Port On IP Device - Disable:  No
Answer Plus Delay To Message Timer:  20
Answer Plus Expected Off-hook Timer:  30
Answer Plus Message Length Timer:  10
Answer Plus System Reroute Timer:  0
Recorded Announcement Device:  No
Recorded Announcement Device - Advanced:  No
Delay Ring Timer:  10
No Answer Recall Timer:  17
Ringing Line Select:  No
Ringing Timer:  180
SMDR External:  No
SMDR Internal:  Yes
ANI/DNIS/ISDN Number Delivery Trunk:  No
DASS II OLI/TLI Provided:  No
Public Network Access via DPNSS:  Yes
Public Network To Public Network Connection Allowed:  Yes
Public Trunk:  No
R2 Call Progress Tone:  No
Suppress Simulated CCM after ISDN Progress:  Yes
Trunk Calling Party Identification:  Yes
Trunk Flash Allowed:  No
Two B-Channel Transfer Allowed:  No
COV/ONS/E&M Voice Mail Port:  No
ONS VMail-Delay Dial Tone Timer:  5
Account Code Length:  12
Account Code Verified:  No
Forced Non-Verified Account Code:  No
Forced Verified Account Code:  No
Non Verified Account Code:  Yes
Attendant Busy Out Timer:  10
SC1000 Attendant Basic Function Key:  No
Conference Call:  Yes
Disable Conference Join Tone:  No
Do Not Disturb:  Yes
Do Not Disturb - Access to Remote Phones:  Yes
Do Not Disturb Permanent:  No
Emergency Call - Audio Level for Set:  Ringer
Emergency Call Notification - Audio:  No
Emergency Call Notification - Visual:  No
Group Presence Control:  Yes
Group Presence Third Party Control:  Yes
Display VIP:  No
Hotel Room Monitor Setup Allowed:  No
Hotel Room Monitoring Allowed:  No
Hotel/Motel Room Personal Wakeup Call Allowed:  No
Hotel/Motel Room Remote Wakeup Call Allowed:  No
Message Waiting:  Yes
Message Waiting - Disable Ringing Lamp Notification:  No
Message Waiting Audible Tone Notification:  No
Message Waiting Deactivate On Off-Hook:  Yes
Message Waiting Inquire:  Yes
Message Waiting Ringing Start Time Hour:   
Message Waiting Ringing Start Time Minute:   
Message Waiting Ringing Stop Time Hour:   
Message Waiting Ringing Stop Time Minute:   
Multiline Set Voice Mail Callback Message Erasure Allowed:  No
ONS CLASS/CLIP: Message Waiting Activate/Deactivate:  No
Auto Answer Allowed:  Yes
Auto Release on Key Select:  No
Brokers Call:  No
Called Party Features Override:  No
Check COR after PSTN Dial Tone:  No
Dialled Night Service:  Yes
Disable Send Message:  No
Flexible Answer Point:  No
Individual Trunk Access:  Yes
Key A: 
Key B: 
Key C: 
Key D: 
Multiline Set Loop Test:  No
Multiline Set Message Center Remote Read Allowed:  No
Multiline Set Music:  No
Multiline Set On-hook Dialing:  Yes
Multiline Set Phonebook Allowed:  Yes
Non DID Extension:  No
ONS CLASS/CLIP: Set:  No
ONS/OPS Internal Ring Cadence for External Callers:  No
Override Interconnect Restriction on Transfer:  No
Recall If Transferred to Original Call Destination:  No
Redial Facilities:  Yes
Use Default Billable Number For Trunk Calls:  No
Voice Dial Preferred:  No
Voice Mail Softkey:  No
Phonebook Lookup - Default to User Location:  No
Phonebook Lookup - Display User Location:  No
Record-A-Call - Save Recording on Hang-up:  No
Record-A-Call - Start Automatic Incoming Call Recording:  No
Record-A-Call - Start Automatic Outgoing External Call Recording:  No
Record-A-Call Active:  No


Thanks again,

Stan


Offline StanO

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Re: Problem Integrating Exchange 2013 UM with Mitel vMCD with MiCollab/MBG
« Reply #1 on: September 21, 2015, 10:25:36 PM »
I'm still struggling with this and hoping somebody can point me in the right direction.

Running a packet capture while calling, via DID, an internal number that is still hosted on our Exchange 2007 server yields the following Graph Analysis:
|Time     | 64.28.213.10                          | 64.28.214.167                         |
|         |                   | 172.16.0.12       |                   
|94.005240|         INVITE SDP (g729 g71          |                   |SIP From: "Caller X" <sip:5555555211@64.28.214.167:5070 To:<sip:5555555621@64.28.114.164:5060;user=phone;tgrp=nfytk1234567
|         |(5060)   ------------------>  (5060)   |                   |
|94.005835|         100 Trying|                   |                   |SIP Status
|         |(5060)   <------------------  (5060)   |                   |
|94.053788|         180 Ringing                   |                   |SIP Status
|         |(5060)   <------------------  (5060)   |                   |
|94.186669|         PRACK     |                   |                   |SIP Request
|         |(5060)   ------------------>  (5060)   |                   |
|94.188806|         200 OK    |                   |                   |SIP Status
|         |(5060)   <------------------  (5060)   |                   |
|94.257148|                   |         RTP (g711U)                   |RTP Num packets:594  Duration:12.456s SSRC:0xFC2DBBFF
|         |                   |(25082)  ------------------>  (35580)  |
|94.397083|         200 OK SDP (g711U te          |                   |SIP Status
|         |(5060)   <------------------  (5060)   |                   |
|94.530074|         ACK       |                   |                   |SIP Request
|         |(5060)   ------------------>  (5060)   |                   |
|112.050889|         BYE       |                   |                   |SIP Request
|         |(5060)   ------------------>  (5060)   |                   |
|112.051357|         200 OK    |                   |                   |SIP Status
|         |(5060)   <------------------  (5060)   |                   |

The Graph Analysis for a call to an internal number with a mailbox residing on an Exchange 2013 server via its DID yields:
|Time     | 64.228.213.10                          |
|         |                   | 172.16.0.12       |                   
|70.764610|         INVITE SDP (g729 g71          |SIP From: "Caller X" <sip:5555555211@64.28.214.167:5070 To:<sip:5555555582@64.28.214.166:5060;user=phone;tgrp=nfytk1234567
|         |(5060)   ------------------>  (5060)   |
|70.765207|         100 Trying|                   |SIP Status
|         |(5060)   <------------------  (5060)   |
|71.279329|         180 Ringing SDP (g71          |SIP Status
|         |(5060)   <------------------  (5060)   |
|71.412567|         PRACK     |                   |SIP Request
|         |(5060)   ------------------>  (5060)   |
|71.414481|         200 OK    |                   |SIP Status
|         |(5060)   <------------------  (5060)   |
|71.415499|         200 OK    |                   |SIP Status
|         |(5060)   <------------------  (5060)   |
|71.548005|         ACK       |                   |SIP Request
|         |(5060)   ------------------>  (5060)   |
|78.993165|         BYE       |                   |SIP Request
|         |(5060)   ------------------>  (5060)   |
|78.993638|         200 OK    |                   |SIP Status
|         |(5060)   <------------------  (5060)   |

I notice in the case of the non-working Exchange 2013 call (DID 582), 180 response also contains SDP information, whereas this is not the case with the 180 response of the working call to the Exchange 2007 DID (621).  Subsequently, on the Exchange 2013 DID Graph Analysis, there is no RTP session established.  Pulling up the tug.log from the MBG for the call to the non-working 582 DID I see reports of bad RTP headers coming from the Exchange server.

Any clues here as to wherein the problem may lie?

Thank you,

Stan

Offline StanO

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Re: Problem Integrating Exchange 2013 UM with Mitel vMCD with MiCollab/MBG
« Reply #2 on: September 23, 2015, 11:52:12 PM »
This turned out to be an issue with inconsistent handling of SIP & RTP sessions by a Fortinet FortiGate firewall.

In spite of the default SIP Application Layer Gateway being set to used the session helper and the session helper having been removed, it continued to meddle with *some* SIP/RTP traffic.

http://help.fortinet.com/fos50hlp/52data/index.htm#FortiOS/fortigate-whats-new-52/other-new-features.htm#SIP_Traffic_is_Handled_by_the_SIP_ALG_by_Default%3FTocPath%3DChapter%25201%2520-%2520What%27s%2520New%2520for%2520FortiOS%25205.2.1|Other%2520New%2520Features|_____1

Quote
Previous versions of FortiOS used the SIP session helper for all SIP sessions. You had to remove the SIP session helper from the configuration for SIP traffic to use the SIP ALG.

In FortiOS 5.2, all SIP traffic is now processed by the SIP ALG by default. You can change the default setting using the following command:

config system settings
set default-voip-alg-mode {proxy-based | kernel-helper-based}
end

The default is proxy-based, which means the SIP ALG is used. If set to kernel-helper-based, the SIP session helper is used. If a SIP session is accepted by a firewall policy with a VoIP profile, the session is processed using the SIP ALG even if default-voip-alg-mode is set to kernel-helper-based.

If a SIP session is accepted by a firewall policy that does not include a VoIP profile:

    If default-voip-alg-mode is set to proxy-based, SIP traffic is processed by the SIP ALG using the default VoIP profile.
    If default-voip-alg-mode is set to kernel-helper-based, SIP traffic is processed by the SIP session helper. If the SIP session help has been removed, then no SIP processing takes place.

Tomorrow night I'll look for a balance that allows everything that worked previously to keep working reliably and add reliable Exchange 2013 voicemail as well.

Apologies for the post that ended up being off-topic.

Stan

Offline Gasmanz

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Re: Problem Integrating Exchange 2013 UM with Mitel vMCD with MiCollab/MBG
« Reply #3 on: February 21, 2016, 06:33:25 PM »
We had a customer with a Fortinet FortiGate firewall that engaged support from Fortinet to disable SIP ALG. The Fortinet engineer said that even after you have disabled the SIP ALG in the config you have to cold boot the FortiGate firewall for the new setting to take affect  ::) .

Ever since then when we have dealt with Fortinet FortiGate firewalls we have found this to be true and always get our customers to cold boot the firewall after disabling the SIP ALG.


 

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