Lync Monitoring Server - Attempted outbound call

Hi,
Im looking at the tables in the monitoring server as a means of monitoring outbound calls where the outgoing agent terminated the call prior to the call being picked up. Is this recorded in the monitoring server? I've checked the ErrorReport which is where
I thought this would land, as well as the sessiondetails table.
Does lync even report such transactions?
Cheers,
Dave.

Hi,
You can try to use Call Diagnostic Reports to check the attempted outbound call.
The Call Diagnostic Reports provide summary information and diagnostic data for failed peer-to-peer and conferencing sessions.
More details:
http://social.technet.microsoft.com/Forums/en-US/9d75e7c4-ba5d-4884-8ec8-ee3bd7195821/lync-monitoring-server-attempted-outbound-call?forum=ocsmonitoring
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support

Similar Messages

Problems with SNOM 7XX phones and presence of Lync Edge server

Hi to all,
we have this problem, this is the scenario (two Lync 2013 st ed. servers):
- lync 2013 FE server have internal IP address 172.21.212.XXX with internal gateway 172.21.212.254
- lync 2013 edge server have two network interface:
First INTERFACE: 3 IPs in 172.21.30.XXX (Access, web and A/V Edge) for external connection with 172.21.30.254 and internal gateway (IP NAT with public IP)
Second INTERFACE: IP 172.21.212.XXX for internal connection without gateway
- snom 7XX (50 phones) are connect to the lync server and all internal call works fine. All phones are in an internal dedicated network 172.21.218.XXX with default gateway 172.21.218.254
- when making external call with 7XX SNOM phones, the call was routed to Trunk COLT with Lync Mediation server and all works fine.
- when Lync Mediation server receive a call from our trunk COLT we have this situation:
All Lync 2013 clients work fine, audio is OK, (network 172.21.216.XXX)
Polycom CX3000 work fine audio is OK (network 172.21.218.XXX)
SNOM 710, 720, 760 FW 8.8.2.16 UC series, phones ring but NO SOUNDS from the phones and after a few seconds "Call failed due to network issues."
The only way to solve the problem is to disable the connection with Lync Edge server (remove gateway 172.21.30.254)
BUT this is not the solution because now we have no connection with INTERNET (skype, web conferencing doesen't work without edge gateway)
Why SNOM phones try to use the EDGE gateway to connect the call? Why doesn't use Lync Mediation server?
Can you help us to find a solution?
Thanks
Aurelio

Hi,
Thanks to all for yours support.
Today, we have done some test (no employee in office today
J) and we have solved the problems.
The old implementation have had this configuration:
- the phone numbers have had a no E.164 format compliant: for all users number, the phone number have had this format TEL:012345XYZW ; EXT=XYZW with the normalization
rules:
Starting digits: 01234567
Length: At least 8 digits
Digit to remove: 0
Digit to add: nothing
Pattern to match ^(01234567\d*)$
All worked fine with this previous configuration:
Lync 2010 std with only mediation server function + Lync 2013 std front-end with all the others functions and Lync 2013 std Edge server for external connection with
Lync client Skype world, BUT we have had disabled in SNOM phones ICE function because if ICE was enabled no voice can we hear from the phones.
After dismissed Lync 2010 with only a Lync 2013 infrastructure, this configuration don’t permit to use edge server because with ICE enabled or disabled no voice from
SNOM phones.
Today we have done this operation:
Setting in Lync 2013 control panel all number for all users, in E.164 format compliant:
The phone number now have this format TEL:+39012345XYZW ; EXT=XYZW and we have deleted the previous normalization roles.
We have added this role for the EXT numbers:
Name: Routing Interno
Starting Digits: XY
Length: Exactly 4 (i.e. XYZW)
Digit to remove 0
Digit to add: +39012345
Pattern to match: ^(XY\d(2))$
Translation rule: +39012345$1
Internal extension = checked
And now all work fine.
We have solved another problem:
Lync client 2013 can't find new users:
all new Lync users are not discovered from Lync 2013 client, probably because this setting is present with Lync 2010:
PS C:\> Get-CsAddressBookConfiguration
Identity
: Global
RunTimeOfDay
: 1:30 AM
KeepDuration
: 30
SynchronizePollingInterval : 00:00:30
MaxDeltaFileSizePercentage : 20
UseNormalizationRules
: True
IgnoreGenericRules
: False
EnableFileGeneration
: True
With only Lync 2013 servers we have changed
IgnoreGenericRules to True
To set UseNormalizationRules and IgnoreGenericRules to true for Lynk 2013 infrastructure.
http://technet.microsoft.com/en-us/library/jj205160.aspx
For us all the problems are SOLVED!
Aurelio

Outbound Calls stop working when Lync Edge server is offline

All,
We have had an issue inside our environment after one of our virtual hosts died and took out our sole Edge server, basically users could not dial out and were getting the error "Network is busy" on the client. Internal dialling worked perfectly to our Lync
users but the users could not dial out via our Cisco CME server.
Our configuration is a Lync Enterprise Pool with 2 servers, DB cluster and a single Lync edge server, the mediation servers are installed on our enterprise pool servers as a single server and the CME is looking at the mediation servers directly and has nothing
pointing at the edge server.
This issue affected internal users (as the external users were all kicked out due to the server being down), the strange thing was that you got errors in snooper about the server being unavailable when you dialled out, no idea why it did this. Even stranger
was that the call itself was sent to the CME as a debug SIP showed traffic being attempted between the user and the number which confused me even more as my mobile actually rang for a single ring as well.
Has anybody got any ideas as to why the Edge server would do this to internal users?
Thanks
James

Lync checks the bandwith policy against the Edge server. As the Edge is not responding Lync is unable to check the policy and the call fails.
For the time being you may want to remove the Edge from the topology, then Lync checks against the Front End server.
I hope you do understand, that this is not a great solution, but a drastic workaround suggestion to a hidden product defect! In hidden product defect I mean Microsoft Lync document team is cynically silent and trying to cover the tracks of this product defect.
What I would consider as a straight and honest retroactive action for the Lync document team, to add a big warning section to a) single server edge deployment page + b) the Call Admission Control caveats page on Technet:
"Warning: PRODUCT DEFECT / PRODUCT LIMITATION comes here
If you associate an edge server or pool to a FE pool, and enable Call Admission Control, your single / pool edge will become SINGLE POINT OF FAILURE for your entire enterprise telephony when doing outbound call attempts!
So if outbound calls is important in your company (hell, of course it is!) then deploy at least 2x Edge servers in the same pool before enable CAC!"
But I think that warning message is way too much to ask for, thatswhy is this 2,5 years old topic still open.

Lync 2013 Monitoring Server - Database reporting

Hi there,
Im a business intelligence professional rather than a Lync professional so excuse me if my question may show niavete or a clear lack of experience, both would probably be correct.
The basic question is, how does Lync record the relationship between a call being received and the call that is transferred.
Effectively I am looking to see how the monitoring server applies a unique record to a call. This looks like the SessionIdTime field. I also deduce that the 'ReplacesDialogueId' field is used to record when a call effectively replaces another one, and I
have formed a self join on the SessionDetailsView to track a calls progress.
Im struggling though when this relationship is not there. So for example, when a call comes in and is answered by a receptionist, we have the initial call entering the queue, with the call being answered by Reception having a replacedialogId that replaces
the call coming into the queue. Lets say that the receptionist then calls a director to see if they are available, and then transfers the call. There is a relationship between the call being made to the director, and the caller speaking to the director with
the dialogId and replacedialogid fields, though there is no relationship between the receptionist taking the call, and the receptionist calling the director and putting the call through.
Again, please forgive my noobie questions. In summary, how does Lync record the relationship between a call being received and a transferred call.

They are sparse, many of us have written custom reports but I forget what I've done as soon as I've done it. The Lync 2013 CDR Database Schema is available here:
http://technet.microsoft.com/en-us/library/gg398570.aspx
Typically what I do is just dig in and make test case calls, export the query results to something easy to work with like Excel and reverse engineer what I need for my reports. Watch the CorrelationID and anything else you can line up.
I'm not aware of any great guides that walk you though this unfortunately.
Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question please click "Mark As Answer".
SWC Unified Communications

Lync 2011 for Mac with iMessage on OSX inbound and outbound calls fail

Running Office 14.4.9 updates and many before that. If you activate or deactivate an iMessage account using the built-in Messages application, Lync can no longer make or receive calls. Calls received will automatically go to voicemail.
Outbound calls just ring and ring. The phone setting in Lync were all three set to Lync as well.
The only way I've found them to work is to following the first instructions at the following link:
https://support.microsoft.com/en-us/kb/kbview/2691870?wa=wsignin1.0
Log on to your computer by using administrative credentials.
Exit Lync if it's running.
Drag the Lync application to the Trash.
To remove your existing Lync preferences, delete the following files:
Users/username/Library/Preferences/com.microsoft.Lync.plist
Users/username/Library/Preferences/ByHost/MicrosoftLyncRegistrationDB.xxxx.plist
Users/username/Library/Logs/Microsoft-Lync-x.log
If you activate or deactivate an iMessage account steps 1-4 must be performed, otherwise, phone calls fail. One additional annoyance I've noticed is that Lync tends to hand with a red X on the icon in the dock when this happens as well. You must
use the terminal and do 'pkill Lync' in order to close it.

Hi MSGuest,
Can you install the latest update and then check again ?
https://www.microsoft.com/en-us/download/details.aspx?id=36517
Best regards,
Eric
Please remember to mark the replies as answers if they help, and unmark the answers if they provide no help. If you have feedback for TechNet Support, contact [email protected]

WMIProviderException: An error occurred when attempting to connect to the report server remote procedure call (RPC) end point

I am getting the following error message while trying to configure a database for a Reporting Service Server. Both SQL Server and Reporting server are 2012 named instances. I have 2014 instances as well on the same server.
Exception details...
Microsoft.ReportingServices.WmiProvider.WMIProviderException: An error occurred when attempting to connect to the report server remote procedure call (RPC) end point. Verify that the Report Server Windows service is running, and then retry the operation.
---> System.Runtime.InteropServices.COMException (0x800706B3): The RPC server is not listening. (Exception from HRESULT: 0x800706B3)
   --- End of inner exception stack trace ---
   at Microsoft.ReportingServices.WmiProvider.RSWmiAdmin.ThrowOnError(ManagementBaseObject mo)
   at Microsoft.ReportingServices.WmiProvider.RSWmiAdmin.GenerateDatabaseScript(String databaseName, Int32 lcid, Boolean isSharePointIntegrated, String& script)
   at ReportServicesConfigUI.WMIProvider.RSReportServerAdmin.GenerateDatabaseScript(String databaseName, Int32 lcid, Boolean isSharePointIntegrated, String& script)
Can someone help me on this please?

Hi Jaigi,
According to your description, you want to change a report server database for Reporting Services. But it fails on the last step and returns a error. Right?
In this scenario, it seems you are creating a report server database on another server via remote connection. Based on the error message, it has issues on executing the script for generating new database. Please check if you have permission to connect the
server and create a new database. Also please check if the Reporting Services Windows services is working properly.
If you have any question, please feel free to ask.
Best Regards,
Simon Hou

Lync Server 2010 Mobility. Server does not call mobile.

Hello all.
We have an infrastructure with Lync Server 2010 SE, Edge Server, ISA.
Everything works fine, Enterprise Voice is enabled and integrated with IP PBX.
Lync 2010 desktop client can call to PSTN without any problem.
But when user logs in to mobile client, iPhone Lync Mobility client for example, server tries to call to users' number but an error occurs, that user is not online, IM works just fine.
The reason is - call goes to trunk in the following format 98XXXXXXX with normalization rules, server tries to call in e.164 format +7XXXXXXXXXX.
Everybody knows how to normalize server calls?

Open Lync Server Control Panel, click Voice Routing and then click Test Voice Routing, enter the dialed number to test with the Dial Plan and Voice Policy assigned to the caller.
Lisa Zheng
TechNet Community Support

Outbound call failure in TelePresence Server

Setup has CUCM - Conductor - TelePresence Server (virtual). Plan is to use the same setup for scheduled conferences by including TMS. I have done all configuration as per the latest Conductor with TMS deployment guide.
While testing calls, I could see that the conference is getting created in the TelePresence server and the TelePresence server is trying to make a outbound call to the endpoint SIP address (extn@CUCMIP). But the calls are not getting completed.
If I configure TLS in the SIP settings of TS for outbound calls, then I am getting the below in the TS logs.
698
13:33:51.845
APP
Info
conference "Scheduled_Conference_zzzz": deleted via API (no participants)
697
13:29:41.040
APP
Info
call 14: tearing down call to "[email protected]" - destroy at far end request; networkError
696
13:29:41.040
CMGR
Info
call 14: disconnecting, "[email protected]" - network error
695
13:29:41.039
SIP
Error
call 14: Ending call due to network error during INVITE transaction
694
13:29:40.539
APP
Info
call 13: tearing down call to "[email protected]" - destroy at far end request; networkError
693
13:29:40.539
CMGR
Info
call 13: disconnecting, "[email protected]" - network error
692
13:29:40.539
SIP
Error
call 13: Ending call due to network error during INVITE transaction
691
13:29:08.544
APP
Info
call 14: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
690
13:29:08.437
APP
Info
call 13: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
689
13:29:07.765
APP
Info
conference "Scheduled_Conference_zzzz" created
If I use TCP in the SIP settings, I am getting the below in the TS logs.
688
13:03:51.822
APP
Info
conference "Scheduled_Conference_zzzz": deleted via API (no participants)
687
13:01:28.141
NTP
Info
time is Tue Apr 28 13:01:28 2015
686
13:00:32.121
APP
Info
call 12: tearing down call to "[email protected]" - destroy at far end request; unavailable
685
13:00:32.121
CMGR
Info
call 12: disconnecting, "[email protected]" - service unavailable
684
13:00:32.109
APP
Info
call 12: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
683
13:00:32.009
APP
Info
call 11: tearing down call to "[email protected]" - destroy at far end request; unavailable
682
13:00:32.009
CMGR
Info
call 11: disconnecting, "[email protected]" - service unavailable
681
13:00:31.996
APP
Info
call 11: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
680
13:00:01.955
APP
Info
call 10: tearing down call to "[email protected]" - destroy at far end request; unavailable
679
13:00:01.954
CMGR
Info
call 10: disconnecting, "[email protected]" - service unavailable
678
13:00:01.936
APP
Info
call 10: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
Some of the questions which are not answered in the guide are :
Is a new SIP trunk required from CUCM to Conductor. If yes, what is the destination IP for this trunk. Is this the primary conductor IP address. For adhoc & rendezvous conferences, there are seperate SIP trunks created and destination IP is the additional IP address configured.
Any other configuration required in any of the other applications.
Thanks.

Setup has CUCM - Conductor - TelePresence Server (virtual). Plan is to use the same setup for scheduled conferences by including TMS. I have done all configuration as per the latest Conductor with TMS deployment guide.
While testing calls, I could see that the conference is getting created in the TelePresence server and the TelePresence server is trying to make a outbound call to the endpoint SIP address (extn@CUCMIP). But the calls are not getting completed.
If I configure TLS in the SIP settings of TS for outbound calls, then I am getting the below in the TS logs.
698
13:33:51.845
APP
Info
conference "Scheduled_Conference_zzzz": deleted via API (no participants)
697
13:29:41.040
APP
Info
call 14: tearing down call to "[email protected]" - destroy at far end request; networkError
696
13:29:41.040
CMGR
Info
call 14: disconnecting, "[email protected]" - network error
695
13:29:41.039
SIP
Error
call 14: Ending call due to network error during INVITE transaction
694
13:29:40.539
APP
Info
call 13: tearing down call to "[email protected]" - destroy at far end request; networkError
693
13:29:40.539
CMGR
Info
call 13: disconnecting, "[email protected]" - network error
692
13:29:40.539
SIP
Error
call 13: Ending call due to network error during INVITE transaction
691
13:29:08.544
APP
Info
call 14: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
690
13:29:08.437
APP
Info
call 13: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
689
13:29:07.765
APP
Info
conference "Scheduled_Conference_zzzz" created
If I use TCP in the SIP settings, I am getting the below in the TS logs.
688
13:03:51.822
APP
Info
conference "Scheduled_Conference_zzzz": deleted via API (no participants)
687
13:01:28.141
NTP
Info
time is Tue Apr 28 13:01:28 2015
686
13:00:32.121
APP
Info
call 12: tearing down call to "[email protected]" - destroy at far end request; unavailable
685
13:00:32.121
CMGR
Info
call 12: disconnecting, "[email protected]" - service unavailable
684
13:00:32.109
APP
Info
call 12: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
683
13:00:32.009
APP
Info
call 11: tearing down call to "[email protected]" - destroy at far end request; unavailable
682
13:00:32.009
CMGR
Info
call 11: disconnecting, "[email protected]" - service unavailable
681
13:00:31.996
APP
Info
call 11: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
680
13:00:01.955
APP
Info
call 10: tearing down call to "[email protected]" - destroy at far end request; unavailable
679
13:00:01.954
CMGR
Info
call 10: disconnecting, "[email protected]" - service unavailable
678
13:00:01.936
APP
Info
call 10: new outgoing SIP call to "[email protected]" from conference "Scheduled_Conference_zzzz"
Some of the questions which are not answered in the guide are :
Is a new SIP trunk required from CUCM to Conductor. If yes, what is the destination IP for this trunk. Is this the primary conductor IP address. For adhoc & rendezvous conferences, there are seperate SIP trunks created and destination IP is the additional IP address configured.
Any other configuration required in any of the other applications.
Thanks.

Lync 2013 Server / Roles & Components

Lync
2013 Server / Roles & Components
Front End
User authentication and registration
Presence information and contact card exchange
Address book services and distribution list expansion
IM functionality, including multiparty IM conferences
Web conferencing, PSTN Dial-in conferencing and A/V conferencing (if deployed)
Application hosting, for both applications included with Lync Server (for example, Conferencing Attendant and Response Group application), and third-party applications
Primary store for user and conference data. Information about each user is replicated among Front End Servers in the pool
Optionally, Monitoring, to collect usage information in the form of call detail records (CDRs) and call error records (CERs). This information provides metrics about the quality of the media (audio and video) traversing your network for both Enterprise
Voice calls and A/V conferences.
Web components to supported web-based tasks such as web scheduler and join launcher.
One Front End pool runs the Central Management Server DB, which manages and deploys basic configuration data to all servers running Lync
Optionally, Archiving, to archive IM communications and meeting content for compliance reasons.
Optionally, if Persistent chat is enabled, Persistent Chat Web Services for Chat Room Management and Persistent Chat Web Services for File Upload/Download.
Back End
Database server running Microsoft SQL Server
Provide the DB services for the Front End pool
Acts as backup store for the pool’s user and conference data
Primary stores for other DB’s like Response Group
High Availability for the BE DB is provided via SQL Mirroring
Optional Witness to enable automatic failover for BE
SQL Sever 2008 R2 or higher required for SQL Mirroring
Edge Server
Enable users to communicate and collaborate with users outside the organization’s firewall
Comprises four separate server roles
Access Edge – Acts as a secure proxy for all remote Lync signaling traffic
Remote Access
Federation
Public IM Connectivity (PIC)
Web Conferencing Edge – Enable remote users to participate in Web conferences with internal or remote workers
A/V Edge – Responsible for secure relay of A/V media among internal, external, and federated contacts
XMPP Gateway – Allows IM/P with XMPP federated contacts
Reverse Proxy
Simple URL Publishing – Required for users to join Lync meetings
Web Conferencing Content – Users download meeting content (PowerPoint, Whiteboard, and Poll data) via Lync Web Services when in meeting
Address Book & Distribution List Expansion – Required for users to download Lync Address Book and perform DL expansion
User Certificates – Provides client certificate authentication via Lync Web Services
Device Updates – Provides software updates to Lync IP endpoints
Mobility – Provides connectivity for mobile clients via Lync Web Services
Mediation Server
Translates signaling and media between Lync Server and PSTN, IP-PBX, or SIP Trunk
Can be co-located on Front End or separated as stand-alone Server dependent on call volume
Role facilitates dial-in conferencing
Capacity
Co-located = 150 Concurrent Calls
Standalone = 1100 Concurrent Calls
Persistent Chat
Enable users to participate in multiparty, topic-based conversations that persist over time
Pchat Front End server role runs persistent chat service
Pchat Back End server stores chat content and compliance events
Geographic DR is provided via stretched pool and SQL log shipping to replicate DB info
150k provisioned users / 80k concurrent users
Archiving
Uses SQL Server 2008 R2 or SQL Server 2012 for DB
Capable of archiving the following:
Peer-to-peer IM
Multiparty IM
Web Conferences, including uploaded content and events
A/V for peer-to-peer IM and web conferences
Web conferencing annotations and polls
Monitoring
Agent that runs on each Front End Server that collects and manages information from the Front End and Mediation Servers
Stored on SQL Server DB
Leverages SQL Server Reporting Services for creation of reports related to call quality and metrics
Office Web Apps Server
External server leveraged for rendering PowerPoint slides within the Lync client and Lync Web App
Typically leveraged within SharePoint deployments to deliver browser-based versions of Microsoft Office applications
System Center Ops Mgr
Health configuration in Lync Server 2013 is built around System Center Operations Manager and the use of Lync Server Management Packs. These Management Packs include a number of new features and enhancements, including:
Feature
Description
Synthetic Transactions
Windows PowerShell cmdlets that can be run from various locations to ensure that end user scenarios such as sign-in, presence, IM, and conferencing are readily available to end users.
Call Reliability Alerts
Database queries for Call Detail Records (CDR). These records are written by Front End Servers to reflect whether end users were able to connect to a call or why a call was terminated. These queries result in alerts that indicate when a wide range of end
users are experiencing connectivity issues for peer-to-peer calls or basic conferencing functionality.
Media Quality Alerts
Database queries that look at Quality of Experience (QoE) reports published by clients at the end of each call. These queries result in alerts that pinpoint scenarios where users are likely to be experiencing poor media quality during calls and conferences.
The data is built upon key metrics such as packet latency and loss, metrics that are known to directly contribute to call quality.
Component Health
Individual server components raise alerts by using event logs and performance counters. These alerts indicate failure conditions that can severely impact one or more end user scenarios. These alerts can also indicate a variety of other failure conditions,
including services not running, high failure rates, high message latency, or connectivity issues.
Dependency Health
Failures can occur for a variety of external reasons. The management packs now monitor and collect data for some of the critical external dependencies that might indicate severe issues, including IIS availability, CPU and memory usage of servers and processes,
and disk metrics.
Exchange UM
http://www.contactcenterarchitects.com/lync-2013-server-roles-components/

Hi,
Thank you for sharing the information. It is useful for others who not understand Lync Server Roles and Components. You time and effort are appreciated.
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support

Outbound Call in Cisco Finesse

Anybody please help what all configuration to be done in Finesse to do the outbound calls.
I am using Cisco UCCX 10.0, in which i have both the option for Cisco CAD and Finesse. When i run a outbound campaign its coming as an incoming call in Finesse.
When i answer the call, its showing the error "Unable to communicate with Enterprise Server. Outbound option not available".
Please help how to enable the outbound controls in Finesse

This is ineed the case, from RN (somewhat cryptic :-) ):
Cisco Finesse
Cisco Finesse is the next generation browser-based agent and supervisor desktop for Unified CCX. Finesse is an alternative to Cisco Agent Desktop, Cisco Supervisor Desktop, and Cisco Desktop Administrator. Finesse is available with Enhanced and Premium license packages and provides typical inbound voice contact center functionality. It supports Unified Communications Manager-based silent monitoring and workflow-based recording with MediaSense and Work Force Optimization (WFO).
Chris

Response Group to PSTN Outbound Caller ID

Hi Guys,
I'm trying to implement an after hours paging service on our lync system using a response group (with overflow set to 0), but the problem I'm currently facing is that the outbound calling number ID is set to the number of the caller, which I'm not legally
allowed to present here in Australia. I could in theory set the number to be private, but that will cause people to screen the calls.
Is there any way to get the RGS application to set the call history in the same way that simultaneous ringing does, even perhaps on some server side-scripting?
This is the SIP info I get in a simultaneous ring, where xxx is the number they called, and yyy being the number they called from.
HISTORY-INFO: <sip:+xxx@frontend;user=phone>;index=1,<sip:+yyy@frontend;user=phone>;index=1.1
Any input would be very appreciated :)

I'm curious what happens if you try this:
You'd create a new voice policy, PSTN usage and route. On the route, you mask the caller ID to be your business's primary number.
Then assign that policy to a user account that you've created just for this purpose. That user account should forward it's calls to the number the response group should call.
Then, with the overflow still set to 0, instead of a calling the number, you call that forwarding user account with the caller id mask.
Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question please click "Mark As Answer".
SWC Unified Communications
This forum post is based upon my personal experience and does not necessarily reflect the opinion or view of Microsoft, its employees, or other MVPs.

485 Ambiguous - Outbound Calls Only

I'm having some issues with the 485 Ambigous Error, but on Outbound calls only. I've read several blog posts and was able to solve this issue for incoming calls, but have yet to find a solution for outbound calls.
I have two phone numbers: XXX-XXX-5232 and XXX-XXX-3081. All of my users are configured in Lync with +1XXXXXX5232;ext=XXXXX
XXX-XXX-5232 is set to normalize to +1XXXXXX5232;ext=53999 which is a Lync User that is setup with Team Call to ring multiple other extensions. I'm not aware of any AutoAttendant or Response group that is configured with just +1XXXXXX5232.
Again, inbound calling is working just fine and outbound calls work as expected, they just report the 485 Ambiguous for each call.
Thanks in advance for any help you can provide.

Yes, the call is routed to the Gateway. In fact, the call completes successfully. Here is the trace. Hopefully it will be readable:
12:17:03.513 : 10.188.0.19 : NOTICE : (      lgr_flow)(2288602   ) ---- Incoming SIP Message from 10.188.0.18:61275 to SIPInterface #0 ---- [Time: 11:17:03]
12:17:03.543 : 10.188.0.19 : NOTICE : INVITE sip:[email protected];user=phone SIP/2.0
FROM: "VRRL - Kitchen"<sip:+1XXXXXX5232;[email protected];user=phone>;epid=621B6C2CF5;tag=aeb33cd2cc
TO: <sip:[email protected];user=phone>
CSEQ: 32959 INVITE
CALL-ID: 14fe8790-50ef-476a-80ac-e57061c0a2af
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 10.188.0.18:61275;branch=z9hG4bK7f7f1664
CONTACT: <sip:VRRL-SBA.cdol.int:5067;transport=Tls;ms-opaque=86db4f0fd1133b15>
CONTENT-LENGTH: 552
SUPPORTED: 100rel
USER-AGENT: RTCC/4.0.0.0 MediationServer
CONTENT-TYPE: application/sdp
ALLOW: ACK
Allow: CANCEL,BYE,INVITE,PRACK,UPDATE
v=0
o=- 885 1 IN IP4 10.188.0.18
s=session
c=IN IP4 10.188.0.18
b=CT:1000
t=0 0
m=audio 53854 RTP/AVP 97 101 13 0 8
c=IN IP4 10.188.0.18
a=tcap:1 RTP/SAVP
a=pcfg:1 t=1
a=rtcp:53855
a=label:Audio
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ssoH69QQto9/wyQyDEbtGezAe4zuH4ulyHNtUfRT|2^31|1:1
a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:Eas2Y5diRZ5HKgxFHpLLTdr8EWMmERj6ZGLjf8LO|2^31
a=sendrecv
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
[Time: 11:17:03]
12:17:03.573 : 10.188.0.19 : NOTICE : (     sip_stack)(2288604   ) new AcSIPCallAPI created - #276 [Time: 11:17:03]
12:17:03.593 : 10.188.0.19 : NOTICE : (      lgr_flow)(2288605   ) |       | new GetNewSIPCall created - #517 [Time: 11:17:03]
12:17:03.603 : 10.188.0.19 : NOTICE : ( lgr_stk_mngr)(2288606   ) Resource StackSession <#276> Allocated [Time: 11:17:03]
12:17:03.613 : 10.188.0.19 : NOTICE : (     sip_stack)(2288607   ) TlsTransportObject#57::CheckForConnectionPersistent - Opening persistent connection with proxy: 10.188.0.18:61275 [Time: 11:17:03]
12:17:03.613 : 10.188.0.19 : NOTICE : (      lgr_flow)(2288608   ) |       |(SIPTU#517)INVITE State:Idle() [Time: 11:17:03]
12:17:03.623 : 10.188.0.19 : NOTICE : (      lgr_flow)(2288609   ) DNSResolver::HandleARecordQuery - Host:VRRL-SBA.cdol.int resolved in external table [Time: 11:17:03]
12:17:03.633 : 10.188.0.19 : NOTICE : (     sip_stack)(2288610   ) (SIPTU#517) HandleResolutionSuccessEV: Domain name VRRL-SBA.cdol.int was successfully resolved to IP: 10.188.0.18 [Time: 11:17:03]
12:17:03.643 : 10.188.0.19 : NOTICE : (     sip_stack)(2288611   ) SIPCall(#517) changes state from Idle to Invited [Time: 11:17:03]
12:17:03.653 : 10.188.0.19 : NOTICE : (      lgr_flow)(2288612   ) |       |       |       #276:SIP_DNS_RESOLVED_EV(14fe8790-50ef-476a-80ac-e57061c0a2af)
[Time: 11:17:03]
12:17:03.663 : 10.188.0.19 : NOTICE : (      lgr_flow)(2288613   ) |       |       |       #276:SIP_SETUP_EV(14fe8790-50ef-476a-80ac-e57061c0a2af)
[Time: 11:17:03]
12:17:03.673 : 10.188.0.19 : NOTICE : (      lgr_call)(2288614   ) (#276) Call Allocated. [Time: 11:17:03]
12:17:03.673 : 10.188.0.19 : NOTICE : (   lgr_stk_ses)(2288615   ) SIPStackSession::HandleStackSetupEV - NEWCALL: SrcPN=0 [Time: 11:17:03]
12:17:03.683 : 10.188.0.19 : NOTICE : (   lgr_stk_ses)(2288616   ) <SESSION #276> SendToCall - event: NEW_CALL_EV m_Call#276 [Time: 11:17:03]
12:17:03.693 : 10.188.0.19 : NOTICE : (      lgr_flow)(2288617   ) |       |       #276:NEW_CALL_EV:(14fe8790-50ef-476a-80ac-e57061c0a2af) [Time: 11:17:03]
12:17:03.703 : 10.188.0.19 : NOTICE : (      lgr_flow)(2288618   ) |       |       #276:Call changing states from:IdleState to:NewCallState_IP2Tel [Time:
11:17:03]
12:17:03.713 : 10.188.0.19 : NOTICE : (      lgr_flow)(2288619   ) ServicesMngr::GetEndPoint PhoneNum = 402XXX0899
[Time: 11:17:03]
12:17:03.713 : 10.188.0.19 : NOTICE : (   lgr_psbrdif)(2288620   ) GetTrunkGroupId- TrunkGroup:1 found DstNum:402XXX0899 DstPfx:* SrcNum:+1XXXXXX5232 SrcPfx:* SrcIp:abc0012 SrcIpPfx:10.188.0.18 [Time: 11:17:03]
12:17:03.723 : 10.188.0.19 : NOTICE : (   lgr_psbrdif)(2288621   ) QueryOnHookPortStatus (ChannelNum=0), status = 1 Polarity = 0 [Time: 11:17:03]
12:17:03.733 : 10.188.0.19 : NOTICE : (   lgr_psbrdif)(2288622   ) Current trunks status: [Time: 11:17:03]
12:17:03.743 : 10.188.0.19 : NOTICE : (       lgr_num)(2288623   ) PhoneNumber::RemovePrefix - Number change from +1XXXXXX5232 to 1XXXXXX5232 [Time: 11:17:03]
12:17:03.753 : 10.188.0.19 : NOTICE : (      lgr_call)(2288624   ) Call::SetCoderListForCall #276 Found 2 Common Coders For Call [Time: 11:17:03]
12:17:03.763 : 10.188.0.19 : NOTICE : (      lgr_call)(2288625   ) <Call #276> Coder g711Ulaw64k20 : 20 [Time: 11:17:03]
12:17:03.763 : 10.188.0.19 : NOTICE : (      lgr_call)(2288626   ) <Call #276> Coder g711Alaw64k20 : 20 [Time: 11:17:03]
12:17:03.773 : 10.188.0.19 : NOTICE : ( lgr_profiling)(2288627   ) <Call 276> Profiled<Tel=0,Ip=0>: JBMinDel=10 JBOptF=10 EEarlyM=1 FaxTM=1 IPDS=46 IsFaxU=2 PI2IP=-1 SigIPDF=40 CNGMode=0 DTMFUsed=0 NSEMode=0 PlayRBTone2IP=1
RBUdpPort=0 RTPRD=0 SCE=0 VxxTT=2 Dst2Rdrt=0 DTMFVol=20 ECE=1 ECurDis=0 EDigDel=0 ERevP=0 FHPer=700 InG=32 MWIA=0 MWID=0 VVol=32 ReorderTime=255 DIDWink=0 2StageDial=0 DiscOnBusyT=1 DiscOnBrok=1 DPInd=255 AGC=0 NLP=0 [Time: 11:17:03]
12:17:03.783 : 10.188.0.19 : NOTICE : (      lgr_call)(2288628   ) |       |       #276GetNextUI:GlobalUI=442334516, mACAddrLsb=3257879 [Time: 11:17:03]
12:17:03.793 : 10.188.0.19 : NOTICE : (      lgr_call)(2288629   ) |       |       #276GetNextUI:GlobalUI=442334517 [Time: 11:17:03]
12:17:03.803 : 10.188.0.19 : NOTICE : (      lgr_flow)(2288630   ) |       #0:NEW_CALL_EV   : (14fe8790-50ef-476a-80ac-e57061c0a2af) [Time: 11:17:03]
12:17:03.813 : 10.188.0.19 : NOTICE : (      lgr_flow)(2288631   ) EndPoint::MediaResourceList::AllocateMediaIpPortsByMediaRealmID Perform NEW allocation of Media ports for RealmIndex(0) port(6220) current allocations
are:(1) [Time: 11:17:03]
12:17:03.813 : 10.188.0.19 : NOTICE : (     sip_stack)(2288632   ) SIPSDPSession#276 - Changing state from SIP_MEDIA_IDLE to SIP_MEDIA_OFFERED [Time: 11:17:03]
12:17:03.823 : 10.188.0.19 : NOTICE : (     sip_stack)(2288633   ) <BaseSIP SDPSESSION #276> UpdateChosenMediaByCN - CN as Remote 1 [Time: 11:17:03]
12:17:03.833 : 10.188.0.19 : NOTICE : (     sip_stack)(2288634   ) <BaseSIP SDPSESSION #276> UpdateChosenMediaByCN - Force silence suppression on chosen coder, because remote & local support CN [Time: 11:17:03]
12:17:03.843 : 10.188.0.19 : NOTICE : (      lgr_flow)(2288635   ) |       |(SIPTU#517)TRYING_REQ State:Invited(14fe8790-50ef-476a-80ac-e57061c0a2af) [Time: 11:17:03]
12:17:03.853 : 10.188.0.19 : NOTICE : (     sip_stack)(2288636   ) New SIPMessage created - #58 [Time: 11:17:03]
12:17:03.863 : 10.188.0.19 : NOTICE : (      lgr_flow)(2288637   ) ---- Outgoing SIP Message to 10.188.0.18:61275 from SIPInterface #0 ---- [Time: 11:17:03]
12:17:03.873 : 10.188.0.19 : NOTICE : SIP/2.0 100 Trying
Via: SIP/2.0/TLS 10.188.0.18:61275;branch=z9hG4bK7f7f1664
From: "VRRL - Kitchen"<sip:+1XXXXXX5232;[email protected];user=phone>;epid=621B6C2CF5;tag=aeb33cd2cc
To: <sip:[email protected];user=phone>;tag=1c274616087
Call-ID: 14fe8790-50ef-476a-80ac-e57061c0a2af
CSeq: 32959 INVITE
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000 - MSBG/v.6.20A.045.006
Content-Length: 0
[Time: 11:17:03]
12:17:03.883 : 10.188.0.19 : NOTICE : (     sip_stack)(2288639   ) Resource SIPMessage deleted - #58 [Time: 11:17:03]
12:17:03.883 : 10.188.0.19 : NOTICE : (   lgr_stk_ses)(2288640   ) SIPStackSession::HandleStackSetupEV - SETUP: SrcPN=0 [Time: 11:17:03]
12:17:03.893 : 10.188.0.19 : NOTICE : (   lgr_stk_ses)(2288641   ) <SESSION #276> SendToCall - event: SETUP_EV m_Call#276 [Time: 11:17:03]
12:17:03.903 : 10.188.0.19 : NOTICE : (      lgr_flow)(2288642   ) |       |       #276:SETUP (TO:402XXX0899, FROM:+1XXXXXX5232):(14fe8790-50ef-476a-80ac-e57061c0a2af)
[Time: 11:17:03]
12:17:03.913 : 10.188.0.19 : NOTICE : (      lgr_flow)(2288643   ) |       |       #276:Call changing states from:NewCallState_IP2Tel to:InitiatedState_IP2Tel
[Time: 11:17:03]
12:17:03.923 : 10.188.0.19 : NOTICE : (      lgr_flow)(2288644   ) |       #0:SETUP_EV   : (14fe8790-50ef-476a-80ac-e57061c0a2af) [Time: 11:17:03]
12:17:03.933 : 10.188.0.19 : NOTICE : (   lgr_psbrdif)(2288645   ) UpdateChannelParams, Channel 0
[Time: 11:17:03]
12:17:03.943 : 10.188.0.19 : NOTICE : (   lgr_psbrdif)(2288646   ) #0:PSOSBoardInterface::ConfigureFaxModemChannelParams FAXTransportType=3 Modem configuration VxxTransportType=2 not allowed, forced to 3
[Time: 11:17:03]
12:17:03.953 : 10.188.0.19 : NOTICE : (   lgr_psbrdif)(2288647   ) #0:ConfigFaxModemChannelParams NSEMode=0, CNGDetMode=0, FAXTranType=3, VxxTranType=3, VoiceVol= 0, DTMFVol=-11, InGain=0, RTPRedDepth=0, ECE=1, SCE=3, ECNlpMode=0,
DJBufMinDelay=10, DJBufOptFac=10, Result=1) [Time: 11:17:03]
12:17:03.963 : 10.188.0.19 : NOTICE : (   lgr_psbrdif)(2288648   ) Turn ringer ON for channel 0 [Time: 11:17:03]
12:17:03.973 : 10.188.0.19 : NOTICE : (      lgr_flow)(2288649   ) |       #0:FXO Seize Line [Time: 11:17:03]
12:17:03.973 : 10.188.0.19 : NOTICE : (      lgr_flow)(2288650   ) |       #0:ALERT_EV (send) : (14fe8790-50ef-476a-80ac-e57061c0a2af) [Time: 11:17:03]
12:17:03.983 : 10.188.0.19 : NOTICE : (      lgr_flow)(2288651   ) |       |       #276:ALERT_EV:(14fe8790-50ef-476a-80ac-e57061c0a2af) [Time: 11:17:03]
12:17:03.993 : 10.188.0.19 : NOTICE : (      lgr_flow)(2288652   ) |       |       #276:Call changing states from:InitiatedState_IP2Tel to:AlertingState_IP2Tel
[Time: 11:17:03]
12:17:04.003 : 10.188.0.19 : NOTICE : (      lgr_flow)(2288653   ) |       |       |       #276:ALERT_EV(14fe8790-50ef-476a-80ac-e57061c0a2af)
[Time: 11:17:03]
12:17:04.013 : 10.188.0.19 : NOTICE : (     sip_stack)(2288654   ) New SIPMessage created - #93 [Time: 11:17:03]
12:17:04.013 : 10.188.0.19 : NOTICE : (     sip_stack)(2288655   ) SIPSDPSession#276 - Changing state from SIP_MEDIA_OFFERED to SIP_MEDIA_COMPLETED [Time: 11:17:03]
12:17:04.023 : 10.188.0.19 : NOTICE : (   lgr_stk_ses)(2288656   ) DtmfCapNegotiationAlgorithm :: TxDtmfMethod = DTMF_RFC2833_SUPPORTED [Time: 11:17:03]
12:17:04.033 : 10.188.0.19 : NOTICE : (   lgr_stk_ses)(2288657   ) DtmfCapNegotiationAlgorithm :: TxRtpRfc2833Payload = 101 [Time: 11:17:03]
12:17:04.043 : 10.188.0.19 : NOTICE : (   lgr_stk_ses)(2288658   ) <SESSION #276> SendToCall - event: DTMF_CONTROL_EV m_Call#276 [Time: 11:17:03]
12:17:04.053 : 10.188.0.19 : NOTICE : (      lgr_flow)(2288659   ) |       |       #276:DTMF_CONTROL_EV:(14fe8790-50ef-476a-80ac-e57061c0a2af) [Time:
11:17:03]
12:17:04.063 : 10.188.0.19 : NOTICE : SIP/2.0 183 Session Progress
Via: SIP/2.0/TLS 10.188.0.18:61275;branch=z9hG4bK7f7f1664
From: "VRRL - Kitchen"<sip:+1XXXXXX5232;[email protected];user=phone>;epid=621B6C2CF5;tag=aeb33cd2cc
To: <sip:[email protected];user=phone>;tag=1c274616087
Call-ID: 14fe8790-50ef-476a-80ac-e57061c0a2af
CSeq: 32959 INVITE
Contact: <sip:[email protected]:5067;transport=tls>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Require: 100rel
RSeq: 1
Server: Audiocodes-Sip-Gateway-Mediant 1000 - MSBG/v.6.20A.045.006
Content-Type: application/sdp
Content-Length: 254
v=0

UCCX Finesse silent monitoring for direct dial calls

I completely understand that Finesse is geared towards the inbound contact center, at least for right now. Although I can't say the marketing materials really stress that. But I cannot understand the lack of silent monitoring for all call types. Currently silent monitoring only works for inbound calls from UCCX, and does not function for direct inbound calls, or direct dialed outbound calls.
From a system view, Finesse is using UCM's built in silent monitoring, via a very basic JTAPI command. There is very little engineering required to enable this functionality, and I truly thing is is a major limiting issue. Many customers in fact only want the ability to do silent monitoring and basic reporting, and Finesse would be a perfect product for this.
I am considering looking at putting together some custom scripts/web apps to engage the required JTAPI in CUCM via a Finesse Gadget, to replace the built in Finesse silent monitoring function. I'm curious if anyone has some input on how involved that might be, seems like I just really need to pass the monitoring DN and monitored DN through a JTAPI command.
More importantly, is there any hope for Finesse to support this natively? Or other outbound features? If it's a feature "in the works" we have a customer who might be able to "get by" using the Cisco Agent Desktop silent monitoring, but only if that is the case.
Good link on silent monitoring capability in Cisco Finesse: http://www.cisco.com/c/en/us/products/collateral/customer-collaboration/unified-contact-center-express/white-paper-c11-730883.html

We've got to delimit the problem.
When we use 3rd party recording solution there are 4 RTP flows: 1 incoming to monitored/recorded phone and 3 flows outgoing.
1 incoming RTP flows
- voice of remote phone
3 outgoing RTP flows
- voice of monitored/recorded phone
- voice of monitored/recorded phone sent to 3d party recording server
- voice of remote phone sent to 3rd party recording server
When we use a network analyzer we see that the agent running in the PC (UCCX 5.0) establishes two RTP flows for the monitoring session. But we can see that one of these flows sends 3 packet RTP per 1 packet RTP sent in the other RTP flow (in the same time).
We think agent sends to monitoring device all the outoing RTP packets of the phone (that belong to three different RTP flows), instead of sending only the RTP packets belonging to voice of monitored/recorded phone.
The other flow, voice of remote phone, sounds fine.
Is there a solution to avoid this problem when we monitor with UCCX 5.0 and record with a 3rd party software at the same time?
Thanks,
Christian

Cisco Jabber for Windows in Extend and Connect mode and making outbound calls

Hi guys,
I've set up Cisco Jabber for Windows to use Extend and Connect to control a remote PBX endpoint. I've configured the required CTI-RD device, remote destinations, associated the users to the line and added the devices to end-user controlled device. The extend and connect part is working flawlessly without any issues. I'm able to receive inbound calls on the remote PBX endpoint and control the call (hold, resume, transfer etc.) using the Jabber call window that pops up.
However, I'm unable to make any outbound calls via the Jabber client when in extend and Connect mode. Reading the Extend and Connect guide, I need to configure Dial Via Office (DVO) Reverse. So when the user initiates a Dial-Via-Office reverse call, CUCM calls and connect to the Extend and Connect device (CTI-RD). CUCM then calls and connects to the number the user dialled and finally connects the two call legs.
After attempting to configure DVO-R for Jabber for Windows in Extend and Connect mode following the CUCM feature services guide, i'm unable to get any outbound calls working. From RTMT, i am receiving the following Termination Cause Code: (27) Destination out of order. What i also notice is that there is no calling number for that trace either. I would've thought that the calling party would've been the Enterprise Feature Access (EFA) number.
Has anyone got this working or can provide some guidance?
Thanks.

Hi guys,
I've set up Cisco Jabber for Windows to use Extend and Connect to control a remote PBX endpoint. I've configured the required CTI-RD device, remote destinations, associated the users to the line and added the devices to end-user controlled device. The extend and connect part is working flawlessly without any issues. I'm able to receive inbound calls on the remote PBX endpoint and control the call (hold, resume, transfer etc.) using the Jabber call window that pops up.
However, I'm unable to make any outbound calls via the Jabber client when in extend and Connect mode. Reading the Extend and Connect guide, I need to configure Dial Via Office (DVO) Reverse. So when the user initiates a Dial-Via-Office reverse call, CUCM calls and connect to the Extend and Connect device (CTI-RD). CUCM then calls and connects to the number the user dialled and finally connects the two call legs.
After attempting to configure DVO-R for Jabber for Windows in Extend and Connect mode following the CUCM feature services guide, i'm unable to get any outbound calls working. From RTMT, i am receiving the following Termination Cause Code: (27) Destination out of order. What i also notice is that there is no calling number for that trace either. I would've thought that the calling party would've been the Enterprise Feature Access (EFA) number.
Has anyone got this working or can provide some guidance?
Thanks.

Lync monitoring Details reports are not getting opened, Getting SQL error like "Cannot impersonate user for data source 'CDRDB"

Hi,
Two month before, We have deployed Lync 13 set up with lync monitoring and archiving service enabled. After configuring we have the Lync monitoring and archiving services are working fine... few weeks back I have changed the Admin Password
of the Lync Setup. After that Lync Monitoring and Archiving services are not working.
when I try open Lync monitoring service from web UI,
http://localhost/ReportServer_LyncMonitoring/
Web UI is opening but Reports home page is not Accessible , Getting error Like
"An error has occurred during report processing. (rsProcessingAborted)
Cannot impersonate user for data source 'CDRDB'. (rsErrorImpersonatingUser)
Log on failed. Ensure the user name and password are correct. (rsLogonFailed)
The user name or password is incorrect"
Please help me to trouble shoot this problem.
Thanks,
Rajarajan.D

Hi Rajarajan.D
You probably need to update the password against the CDRDB datasource in SSRS Report manager, take a look at this article: http://lyncme.co.uk/microsoft-lync-server-2013/cannot-impersonate-user-for-data-source-cdrdb-rserrorimpersonatinguser-error/
If this helped you please click "Vote As Helpful" if it answered your question please click "Mark As Answer"
Georg Thomas | Lync MVP
Blog www.lynced.com.au | Twitter
@georgathomas
Lync Edge Port Check (Beta)
This forum post is my own opinion and does not necessarily reflect the opinion or view of Microsoft, its employees, or other MVPs.

Lync Monitoring Server - Attempted outbound call

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