Ports Required by Office Communications Server

Ports Required by Office Communications Server

The following table summarizes the ports and protocols used by Office Communications Server servers and clients.

Component (Server role or client)	Port	Protocol	Notes
Front End Servers	5060/5061	TCP MTLS	Used by Standard Edition Servers and Enterprise pools for all internal SIP communications between servers and between servers and Office Communicator
Front End Servers	443	HTTPS	Communication from front-end servers to the Web farm FQDNs (the URLs used by Web Components)
Front End Servers	444	HTTPS	Communication between the focus (Office Communications Server component that manages conference state) and the conferencing servers
Front End Servers	135	DCOM and RPC	Used when a load balancer is deployed, port 135 is used by the Front End Servers for WMI operations and moving users (a remote DCOM-based database operation)
Web Components	443	TCP	HTTPS traffic to the pool URLs
Web Conferencing Server	443	TLS	HTTPS communications to Web Components Servers
Web Conferencing Server	444	TLS	HTTPS between the Web Conferencing Server and the Front End Server
Web Conferencing Server	8057	TLS	Used to listen to direct PSOM connections from Live Meeting client
A/V Conferencing Server	5063	TCP	Used for incoming SIP listening requests
A/V Conferencing Server	49152 – 65535 media port range	UDP	Port range used for media requests sent.
Reverse Proxy	443	TCP	Used for SIP/TLS communications from external users on both the internal and external firewalls for external user access
Access Edge Server	5061	TCP	Used for SIP/MTLS communication for remote user access or federation.
Access Edge Server	443	TCP	Used for SIP/TLS communication for remote user access
Web Conferencing Edge Server	8057	TCP	Used to listen for PSOM/MTLS communications from the Web Conferencing Server on the internal interface of the Web Conferencing Edge Server
Web Conferencing Edge Server	443	TCP	Used for inbound communications for access of remote, anonymous and federated users to access internal Web conferences
A/V Edge Server	443	TCP	Used for STUN/TCP inbound and outbound media communications to allow external users to access media and A/V sessions
A/V Edge Server	5062	TCP	Used for SIP/MTLS authentication of A/V users. Communications flow outbound through the internal firewall.
A/V Edge Server	3478	UDP	Used for STUN/UDP inbound and outbound media communications
A/V Edge Server	50,000-59,999	RTP/TCP	Used for inbound and outbound media transfer through the external firewall.
Office Communicator	5060	TCP (SIP)	Used by Office Communicator for SIP communications internally
Office Communicator	5061	TCP (SIP)	Used by Office Communicator for SIP communications internally and for SIP/MTLS authentication of A/V users. Communications flow outbound through the internal firewall
Office Communicator	443	TCP (HTTP)	Used by Communicator clients connecting from outside the intranet for SIP communications
Office Communicator	1024-65535	UDP/TCP	Port range used for inbound and outbound media transfer through the external firewall.
Office Communicator	6891-6901	TCP	Port ranged used by Office Communicator for file transfer.
Live Meeting 2007 client	443	TCP	Used by Live Meeting 2007 clients connecting from outside the intranet for: SIP traffic sent to the Access Edge Server PSOM traffic sent to the Web Conferencing Edge Server
Live Meeting 2007 client	8057	TCP	Used for outgoing PSOM traffic sent to the Web Conferencing Server
Live Meeting 2007 client	5061	TCP	Used for SIP/TLS communication between Live Meeting and the Front End Servers or the Access Edge Server and for SIP/MTLS authentication of A/V users. Communications flow outbound through the internal firewall
Live Meeting 2007 client	1024-65535	UDP/TCP	Port range used for inbound and outbound media transfer through the external firewall
Live Meeting 2007 client	6891-6901	TCP	Port ranged used by Live Meeting for file transfer

Pasted from <http://technet.microsoft.com/en-us/library/bb870402.aspx>

Configuring Distribution Groups Outlook Voice Access

Configuring Distribution Groups Outlook Voice Access

Forward voicemail to a Distribution Group using the Outlook Voice Access - TechNet Forums http://forums.microsoft.com/TechNet/ShowPost.aspx?PostID=1943597&SiteID=17

C:\Program Files\Microsoft\Exchange Server\Bin>galgrammargenerator -l -x SpeechGrammarFilterList.xml -o GrammarLog3.txt

On the exchange server call distributionlist.cfg
This file has to be updated with the distribution groups that are in the AD.
There is a kicker
The distribution groups only get into this file if they are
Main Enabled Universal Distribution Groups
They also only get into this file if you run the following command:
C:\Program Files\Microsoft\Exchange Server\Bingalgrammargenerator -l -x speechgrammarfilterlist.xml –logfile1.txt(this creates a log file to see what was put in the List)
The other kicker
the Distribution Groups should be something a Human Can say IE Say ETC-MSFT-SME (the words not the letters)
so groups that do not contain normal speak-able words will most likely be impossible to ask for.

If you want to use this feature in your company you may need to readjust your distro lists to something we can say and not spell.

Hybrid Gateway

The Audiocodes M1K hybrid has a quark with it.

The M1k gateway does not come with any Trunk Cards in it or any FXS or FXO cards in it.

When you get a Card to put in to the gateway there is a sticker on it that says " do not install this card until the gateway has been upgraded to the supporting firmware" now there is no way to see the firmware on the gateway until you turn it on.

 
 

Symptoms: the Gateway will not hold its network connection the Link light on the 2 network ports will not stay linked. You will not be able to connect to the web page on the default IP address 10.1.10.10

 
 

Solution: Plug in the Trunk card into the gateway and it will fix the network interface problem and release the gateway from its recycling and seeking of cards

Communicator Requires Restart after Install

After installation of the Communicator Client requires a restart of the workstation which will not complete the install

It will constantly say to restart

 
 

The Resolution is a hot fix from Microsoft

Informational Link

http://support.microsoft.com/default.aspx/kb/943062/

 
 

Hot fix information and Download

http://support.microsoft.com/kb/941441/

Ini File for Audiocodes M1k

This is a Great file to upload into the Gateway to make a standard starting point for all installations. The Things in Bold and highlighted should be changed once the ini is uploaded into the gateway. They should be changed to information about your own environment

To use this file copy and paste the below code into notepad and save the file as an ini
It is always best to upload the file first and then modify it in the gateway.

Ini File Download
SysLog App Download

[SIPgw]
;------------------------------------
; General parameters
;------------------------------------
;To support M1K LAN port redundancy
MIIREDUNDANCYENABLE = 1
ExtBootPReqEnable = 1
;------------------------------------
; Channel parameters
;------------------------------------
; Is Silence Compression enabled (0 - no, 1 - yes) [default 0]
SCE = 0
; Is Echo Canceling enabled (0 - no, 1 - yes) [default 1]
ECE = 1
; [default 0] 0-2 (T38ProtectionMode = 0)
FaxRelayRedundancyDepth = 2
; Redundancy of T.38 control packets[default=1]
FaxRelayEnhancedRedundancyDepth = 2
; Voice gain control.Parameter range is -31 to +31 db. [default = 0 dB] VoiceVolume = 1
; [default 70] 0-150 msec Dynamic Jitter Buffer Minimum Delay.
DjBufMinDelay = 70
;------------------------------------
; DTMF parameters
;------------------------------------
; Use rfc2833 DTMF relay
RXDTMFOPTION = 3
TXDTMFOPTION = 4
RFC2833PayloadType = 101
; 0-1, When The DTMF is being detected (push Button or Release) [default = 1 = release]
MGCPDTMFDetectionPoint = 0
;------------------------------------
; Logger information
;------------------------------------
; When Syslog is enabled, the port must be 514 [Default = 1]
EnableSyslog = 0
; The IP address of the LogServer (when LogOutputType is SYSLOG)
;SysLogServerIP = 10.1.1.89
GWDebugLevel=5
DisableRS232 = 1
;------------------------------------
; Trunk Group Configuration Table
;------------------------------------
[MODULE 0]
TRUNKGROUP_1 = 0-3/1-24,1100
;Select next available channel for Trunk Group ID=1
TrunkGroupSettings = 1,1
;------------------------------------
; TrunkGroup Routing Table
;------------------------------------
PSTNPrefix = *,1
;------------------------------------
; Prefix Routing Table
;------------------------------------
Prefix = 123,10.2.10.1
Prefix = 4321,10.2.10.1
;------------------------------------
; Board Parameters
;------------------------------------
; Set to 0 when working with 10 Base-T hubs. (Default = 4, Auto-negotiation). EthernetPhyConfiguration = 4
; The Progress Tones filename.
CallProgressTonesFileName = 'M2K_usa_tones.dat'
SaveConfiguration = 1
;------------------------------------
; E1 / T1/ ISDN / CAS Parameters
;------------------------------------
; Sets the PSTN protocol to be used for this trunk.
ProtocolType = 10
; Selects the DS1 framing method
; 0 = Extended super frame with CRC6 (default)
; 1 = Super frame D4, F12 (12-Frame multiframe)
; A = F4 (4-Frame multiframe)
; C = Extended super frame without CRC6
FramingMethod = 0
;Use u-law for T1
PCMLawSelect = 3
; Selects the ISDN termination side. (NOT applicable for CAS protocols) (Default = 0).
TerminationSide = 0
; Selects the source of the clock (internal or recovered clock from E1/T1 line) (Default = 0)
ClockMaster = 0
TDMBusClockSource = 4
TDMBusLocalReference = 0
; Selects the line code method to be used for this trunk. (Default = 0). LineCode = 0
;------------------------------------
; Sip Parameters
;------------------------------------
; Applicable to FXO and CAS Mediant 2000. 1 = The Media Gateway disconnects calls when the busy/reorder tone is detected [default].
DisconnectOnBusyTone = 1
;For T1 CAS protocols, play reorder tone before disconnecting
TimeForReorderTone = 5
; The coder used.
CoderName = g711Alaw64k,20
CoderName = g7231,30
; If Proxy Server is used ?
IsProxyUsed = 0
; Proxy-server IP (if used)
ProxyIp = 10.2.1.2
;To enable the T.38 SIP fax relay
IsFaxUsed = 1
; Cnonce parameter for authentication
Cnonce = 0a123bcf
; Password parameter for authentication
Password = 787899
; When using a registering method, set 1. When not, set 0. Default = 0. IsRegisterNeeded = 0
EnableHold = 1
EnableTransfer=1
EnableForward = 1
; 0 = Don't use Early Media, 1 = Enable Early Media. If enabled, the IPmedia server will send 183 Session Progress response (instead of 180 ringing), allowing media session to be established prior to the call being answered.
EnableEarlyMedia = 1
; Registration expired timeout (sec). The value will be used in "Expires = " header. Typically, a value of 3600 will be used, for registration for one hour. The Media Gateway will resume registration before the timeout expires.
RegistrationTime = 3600
; Proxy server host name (if used). Only works if IsProxyUsed = 1. If it doesn't exist in the INI file, the a Proxy IP is used.
ProxyName = audiocodes.com
; Media Gateway host name (if used). Only works if IsProxyUsed = 1. If it doesn't exist in the INI file, the the board IP is used.
SipGatewayName = audiocodes.com
; The IP address of the primary DNS server, in IPv4 format: 'xxx.xxx.xxx.xxx' DNSPriServerIP = 10.2.1.2
; The IP address of the secondary DNS server, in IPv4 format: 'xxx.xxx.xxx.xxx' ;DNSSecServerIP = 10.2.1.3

Sip Test Phone for Sip Gateways

Here is a great way to a Sip Test Phone to make calls from inside the Audiocodes Gateway.

Here is the Situation you want to make some test calls through the Audiocodes/ Dialogic Gateway out to a phone or out to the PSTN.

Here is what i just used to make calls to test my Trunk Setting on an install

the only catch is that the Xlite Sip Phone calls using UDP not TCP

Do not download the demo version of Xlite download the full version

Click the drop down arrow on the top of the App
Go to sip accounts
Click add

Display name is the Caller ID
User name is the DID you want to show for the Caller ID it does not have to be a real number on the PBX at all or it could be a number that is already in use so you could impersonate a user
Domain is the IP address of the Gateway
Uncheck the Register with domain and receive incoming calls

http://www.counterpath.com/x-lite.html

Setup

Can't log into OWA

if you can't log into owa here are some things to check

• Certificates


• Permissions


• Log on challenge is using NTLM


• Has the Directory name of OWA been Changed?

Do not change the directory on OWA as it is integrated into the AD as well. If you need
to change the OWA director Please Read this Microsoft Article.

Cisco Call manager 5.x & 6.x

The Microsoft document That explains how to set up the call manager systems is missing one part if you want to have the traffic come through on TCP rather than UDP for Unified Communications. This is important to remember to change this setting in the call manager system. during the setup of the new Sip trunk other wise there will be a situation like the following

If you call into an Auto attendant or Subscriber access the UM role on that is a part of exchange will not pick up it will either just ring or it will ring and answer to "Dead Air"

To fix this problem you will have to set the Traffic To TCP rather then the Default UDP.