In addition to the current limitations the locations we are evaluating for future campuses don’t improve limitations on the list above.. in fact, they might be even more challenging.  Yet, being live is a huge desire from our leadership, so our quest continues.

Since Fiber isn’t an option at either of our campuses (but hopefully soon), we are limited to 50×10 cable modem in Peoria and a 16×2 cable modem in Galesburg.

our requirements for site to site streaming needs to:
-  provide 1080i display in the remote locations
-  not rely upon a private point to point connection
-  not require more than 10 mb upload from the sending location
-  be a solution that is easily reproduced for future sites in smaller towns with limited connectivity
-  be easily powered up and down by volunteers (not a 30 step process between multiple platforms).

We have demoed the Haivision Mako encoder/decoders and while the encoded video they produce is pretty amazing.. the pricetag is way to high to “not break the bank” not to mention  too high for for a “test environment” as we figure out what “live streaming” really means to our organization.  (However, you should at minimum demo their gear.. the Haivision gear gives a great benchmark for anything else you test.)

So we have been doing some testing with various other streaming solutions and thought we might share our mileage.

For our testing / phase 1 project we decided to try several pieces of gear:
-  Marshall VS-102 Encoder/Decoders
-  Wirecast and Wowza Streaming to a Roku
-  Marshall VS-102 and Zixi.com Hybrid

Our Media director thru the Church Technical Leaders and our peers at Willow Creek came across a encoder/decoder hardware (Marshall VS-102) made by the Display Monitor company Marshall Electronics (and Marshall USA).  We had heard that people were having good success using the VS-102 on the LAN but the device was capable of WAN streaming site to site.  The hardware is also able to additionally stream bi-directional Audio… (hmm maybe ClearCom in addition to the video’s audio?) This device across a LAN some pretty awesome results!  I came into the test expecting YouTube quality and was amazed.  If you are looking for a way to extend your HD/SDI video infrastructure this is a device you should checkout.  I don’t know of many hardware encoder/decoders in this price point … let alone something that can provide such a quality signal.

After a local LAN test, We quickly configured the boxes and streamed from site to site over our hardware VPN connections.  Remember we are using cable modems for our internet.. and the streaming at 1mb was solid.. but video quality was lacking… moving much above 2.5 mb we started to get a lot of jitter and audio drop outs.  If you have more than 10 mb upload.. I suspect you would have much better results, but those are just suspicions since we weren’t able to do such testing.

Next enter Chris Kehayias and his teaching us about Zixi.com.  Zixi is an internet based “private CDN” (Content Delivery Network), their strength is delivering HD video content over the public internet, including higher latency connections without the receiving end dropping frames or loosing quality or dropping audio.  The really awesome piece of the puzzle is the ability to stream from Zixi to a Netgear 550 Media Player.. (a endpoint and decoder for under $100 similar concept to roku).

So with all this new knowledge we started some field testing in Galesburg, so I thought I would share what we have tested and what our results were.

We first started streaming site to site with the two VS-102 units, with similar results to our pretesting, dropping frames and audio if we went above 3 mb.  Next we tested a roku streaming via the Amazon EC2 services but had stability issues even at 1 mb.  The quality of the video, when stable was pretty good, but couldn’t get it dialed in to keep a constant connection.  Next we configured the VS-102 encoder to stream a TS-Mpeg stream rather than the default streaming VS-102 to VS-102.  It streams to the Zixi “sending” application on a PC on the same network.  This PC is responsible for applying the Zixi goodness to the stream and sending it to their cloud.  Then at the remote campus we configured a Netgear 550 Media Player, pointed it to the stream and we have video.  We let the stream ‘chew’ for over 2 1/2 hrs, never dropped a frame or received the audio garbled. 

The only real test we couldn’t get working was to us the VS-102 decoder rather than the Netgear 550.  This was because we couldn’t get the VS-102 to receive what the Zixi receiver was pushing across the LAN.  We are working with Marshall support and expect to test this part soon.  Our motivation to getting the VS-102 to be the end point in Galesburg, 1 is the output of HD/SDI but also hopefully an improved video output beyond the Netgear Media Player.

After a fairly strong showing in Galesburg on Friday, we took the advice of Chris Kehayias, testing the stream during service to see the impact when our Wi-Fi is most populated and everything is buzzing… So during the Sunday Am services we tried streaming from the Peoria Campus to my house via Zixi, first 2 hrs total fail.. too much chewing thru our upload… and after smacking around a dropbox upload we were able to get a stable connection to Zixi and from there smooth sailing… even while the Sending Zixi software reporting having to recover over 50k dropped packets.  On the receiving end, you wouldn’t have known that Zixi was working so hard to keep the stream stable.

Overall I have been impressed by the flexibility of the VS-102, however their support has been limited.  The service of Zixi has been pretty amazing..  keeping a stream rock solid even with pretty poor ISP conditions.

10. You have a project you need to finish by May 1st and don’t have a clue how to do it… nothing like 100+ free consultants to help you.

9. BBQ (nuff said)

8. North Texas spring thunderstorms are awesome!

7. LAN Party (Throw back to an old school LAN party) Bring your own DEW.

6. In & Out Burger AND Chick-Fila AND Hard 8 BBQ in the same city!

5. Free Workshops on Networking/Vmware, VOIP, Wifi and Exchange.  Yes I said FREE!

4. Trying out some of the cheapest most promising site to site streaming gear.

3. Building relationships with some of the most committed partners (aka vendors) serving the Church IT market who are ready to make your ministry shine.

2.  Because everyone else is going.. and you’ll be sad if you aren’t.

1. Hundreds possibly Thousands of Dollars worth of training and peer learning for only the cost of admission $75!

Don’t wait Register NOW!

Continuing the series in our Lync Deployment.  As we are approaching the date that we will completely cut over all users to lync we wanted to build in some redundancy to our deployment. 

We have done this by licensing a second standard server and configuring it in the topology as a backup registrar.  This will allow us to have a fail over server to host all voice calls in the event of a failure to the primary standard edition server (PSE).  The Backup Standard Edition Server (BSE) will provide voice capabilities and limited IM capabilities in a production down situation of the PSE. 
Note: for calls to be made in a ‘failed over’ scenario backup calling routes will need to be configured for the BSE mediation role as discussed in a future post

So we have configured our backup in the topology (how to in a future post) and configured the failover routes so it is time to test the scenarios.  For our testing we want to confirm that the PSE can fail and we can still make calls to the PSTN and if the PSTN is not available make a call out the analog backup lines.

You will want to review the default setting in your topology to set it to the lowest value possible when testing otherwise this test could take 15-20 minutes depending upon your value selected to fail over to a backup registrar. 

Our test was to remove the NIC from the PSE, the Lync clients will disconnect, attempt to re-connect and after the specified time connect to the BSE as the fail over registrar and make calls via the PRI and Pots lines.

However after configuring a Backup Registrar Lync Clients wouldn’t login during a failed server.  The clients would drop the connection as expected but however, they wouldn’t login to the backup registrar with limited functionality as expected. 

Side note… Kudos to @DHannifin helping figure this one out…
check out our awesome buddy Dustin’s blog: http://www.technotesblog.com/ for lots of Uber good Lync goodness.

Even after changing the fail over time to just 30 seconds, the phone handset endpoints would login and calls could be made, but the Lync client would fail to login.   After some digging in the trace logs we found client that wouldn’t connect that we were getting an unauthorized error because the newly added BSE server wasn’t in the user certificate issued by the server to the client so the Lync client didn’t trust the backup registrar.

The Lync Client uses a certificate for communications with the front end server.  This certificate is not updated very often, in fact the default value to when it will update is 8760 HOURS that’s 365 DAYS!  (A little longer than we wanted to wait for our testing…)

You can use the PowerShell command: Get-CSWebServiceConfiguration
to review the current values of your setting for MaxValidityPeriodHours’

Since we didn’t have a year to wait, there are a couple solutions.
1. Change the default value by using the PowerShell command
Set-CSWebServiceConfiguration but this changes the cert settings for all clients and would require time for replication.
or
2. Delete the certificate on the machine that you are using for testing. This is a little more killing a fly with a sledge hammer, but for this testing appeared to be the best solution.

So in a testing scenario where you don’t want to change the re-issue certificate settings, on the machine you are using to test, simply launch an mmc window add the add-in for certificates and choose to manage users certificates.  Next browse to the personal certificates where you should find a certificate named the SIP URI of the user you are logged in as and it is issued by ‘Communications Server’. Delete the certificate and then restart your Lync Client (exit the application not just log off). 

Note: After deleting the cert, before you re-launch the Lync Client, you will need your primary front end server online so a new certificate can be issued to the client on the workstation.  Otherwise you still will not have valid certificate to connect and since the PSE is offline your client will try to connect to the BSE for which it still doesn’t have a valid cert.

After you re-connect to Lync to the PSE you can then power off the PSE (or remove the virtual nic from the virtual machine as we did.) You will notice the Lync client log off and after your Backup Registrar time out passes Lync will login to the Backup Registrar.  You will know this has happed when you see the Lync client display the red bar indicating limited functionality.

If you have correctly configured a backup call route to your gateway, all voice calling will route out the gateway as if your Lync topology was operating normally.

Note: In an actual failover after you have configured all backup routes a call in progress should stay active even while the Lync Client is going thru its log off/log on process to connect to the backup registrar.  If you are in an active call during this fail over, your call should stay connected, BUT it will disconnect if you hit cancel on the Lync client during the reconnection process.

I can’t believe that it’s that time of year again, but registration officially opens today for the National Church IT RoundTable (CITRT) April 18-20, 2012

The National CITRT  is one of those events that I plan to make every year… Thankfully 3.0’s release date is early March so I won’t feel TOOO guilty leaving my wife at home with a newborn and a 3^1/2 yr old.

The Church IT Round table gatherings are a great time to reconnect with old friends, meet some great new friends and learn from some of the great minds in Church IT.  I am already making my list of things I plan to learn while in Dallas.

Make plans now to join us at Watermark Church in Dallas.  This is going to be a can’t miss event!

JUST $75 includes most meals and the event registration for the 2 days (and optional training day).

This year we are adding a optional Pre-Event Training day that will include workshops on:
- Exchange
- VOIP
- VMWare & Networking
- and more

Also NEW this year, we’re making a focused effort to involve Church Web/ChMS developers and integrators, this is an important group of peers to our community, so this year we are building a track for web/dev.  We want our web/dev peers to benefit from an event like we have had over the past 6 years. The web/dev track will have breakout sessions to geek out about code, APIs, ChMS integrations, tools and best practices. These breakouts will be lead by web/dev heads David Drinnon (Second Baptist Houston) and Chris Kehayias (Calvary Chapel Melbourne). So go now and invite your web/dev peers on staff to join you this spring.

If you only go to one training/convention/workshop this year, CITRT2012 is the can’t miss event.. CITRT2012 will make you a better IT Pro, help you build relationships with others doing the same projects you are working on, and improve the ministry you support!

Check out the event info here: http://www.churchitnetwork.com/spring2012/

If you are a vendor with focus on serving the Church market and would like to partner with CITRT to help with this even or have space at the vendor expo check out the “Become A Partner” tab: http://www.churchitnetwork.com/spring2012/

We rewired over 60 data locations in just 3 Monday nights, and the work couldn’t have been done without our awesome volunteers.  I just wanted to say thanks again to Steve T, Wayne T, Gene S, Mark B, John S, and Bob P.  Guys your heart for kingdom ministry is awesome and with out your help we couldn’t do what we do!  Ceiling Tile Dust and carting around ladders is more fun with you guys around!

A couple crazy photos from our work nights:

This is a prime example of why we decided to re-wire! Scotch locks on Data wiring NOOOO!

The Growing pile of wire that has been pulled out

The new IDF wiring rack getting installed

Jeremie and the team formulating the action plan.

Steve has a history of wanting to drive… granted there have been a few accidents, but when it was time to clean up we had something special for him to drive that was fairly accident proof.

Thanks to our volunteers our VOIP migration has an end in site.

This post is a continuation of a series of posts about Lync Deployment. The documentation portion of this project has gotten the back burner, and I need to say that a blogger I am not.. but picking up the documentation of this process is important.

This can be used as a resource to configure an Avaya IPOffice (IPO) 412 (software version 5.0) as a Gateway for a Lync deployment calling the PSTN, with AsteriskNOW as a SIP proxy to resolve disconnected calls when placed on hold or transferred, your mileage may vary. Calls are routed over a SIP Trunk (Session Initiation Protocol) configured between the IPO and Asterisk and Asterisk and the Lync Front End server.

Once we deployed the calling from the PSTN via a PRI from the IPOffice to a SIP connection to the Lync Mediation server we were able to make and receive calls from Lync endpoints, however we quickly noticed that when calls were put on hold or needing to be transferred to another extension the call was simply dropped.  It doesn’t matter if the call was being transferred to a Lync extension or an Avaya extension the call would drop.  The only option to “hold” a call was to mute the call.  If Hold was used the call would disconnect.

After a few days of tracking this down we were able to identify this was an issue that happened every time.  It wasn’t specific to a user or extension.  In fact the Avaya white paper noted this as a known issue.

The issue is documented on the final page: https://devconnect.avaya.com/public/download/interop/OCSR2-IPO-PSTN.pdf

The document notes that calls cannot be placed on mute, nor does the PSTN caller ID pass thru to Lync, these notes however that was not our experience.  Mute and Caller ID worked fine on inbound calls.

We tried several different solutions to resolve this issue.  Our first attempt was routing all calls thru an inGate SIParator.  This is basically a SIP proxy device.  We happen to have one laying around from some testing with a SIP dial tone provider.  This device had worked well with the IPO connecting to SIP Trunks that required authentication with a different authentication handshake than the standard Avaya methods.  However the SIParator did allow to proxy the Avaya to Lync SIP trunk, but didn’t resolve the disconnects when holding or transferring calls.

Next we tried to use a SnomOne software PBX, this had some promise, after configuring the call to forward all calls to the Avaya or Lync (which was a hassle) we found that this resulted in calls connecting but the caller not hearing any of the conversation, or the call would just stop passing audio although it remained connected.  We also found that the SnomOne would keep terminated calls still active and you would have to reset the sessions manually.

Finally we landed on an asterisk installation installed on a virtual machine.  We installed Asterisk now (without the web interface) for simplicity.  Once you configure the two sip trunks (one for Avaya and one for Lync) and build the dial plan to forward all calls from Lync to Avaya and all calls from Avaya to Lync the configuration was basically complete.

Much Credit must go to my great Church IT RoundTable peer Dave Mast (@DaveMast) for his Asterisk Programming help! Kuddos to Dave!

Below are the steps to configure the Avaya and Lync to communicate via an Asterisk Proxy.

Install Asterisk on a machine, (in our case a new VM) and note the IP Address you give the server.  Next configure a new Avaya SIP Trunk and ARS Table. The same steps as noted here, except you need to enter the information of your Asterisk server in step 2 as the ITSP IP Field.

After completing steps 1,2,3 and 4. Complete Step 5 to prepare an incoming call route from Asterisk to the IPO.

Step 6 is basically the same and we repurposed the old ARS table that we created but changed the short codes and features a little. 

Note in step 9 if you have extensions on both IPO and Lync you can’t use variables in your short codes.  This remains true.

After step 10 things change a little so I will document that here.  The information may look very similar to the previous instructions with SIP for IPO and Lync with out a proxy but they are a little different.

Because of how you have to pass calls from Avaya to Asterisk you will need to configure you rARS table a little differently.  Step 10 walks you thru a extension with a DID, that in fact is no different.  But Step 11 has changed. I have quoted the information that hasn’t changed and added what needs to be adjusted for the dialing plan to work with Asterisk.

11. Configure routing for For Lync Extensions without DIDs (as documented here).

An ARS entry will have to be created for each Extension since the IPO cannot use variables in the E.164 formatting of the outbound call and Lync requires the call to come in in the +11235556500;ext=4175 format.

The Asterisk can’t pass the formatting with “;” so we will pass just the 4 digit extension from IPO to Asterisk, and our 4 digit dial plan dialing rule that translates calls TO those extensions from a lync endpoint into +11235556500;ext=4175 format will cause the call to route to the extension when it comes into Lync from Asterisk.

This example extensions 4150-4175 don’t have DIDs but were valid Lync extensions, in order for IPO extensions to call extensions 4150-4175 a short code would be required for 41xx Pointing to the the SIP-Lync ARS Table. (Assuming no other extensions in the 4100 range are homed on the IPO).
Then entries for each extension would need to be added to the ARS table.
Code: 41XX, Feature: Dial (if the IPO has any restricted calls to outside use Dial Emergency)
Telephone Number: +1235556500”ext=4150@192.168.1.100”
Telephone Number: 41N”@192.168.1.100” (the “”s are required to tell IPO that nothing contained in this part of the string is a variable. All extensions in this range can use this variable.

Next you will need to configure Lync to see the Asterisk as a gateway.

1. Configure Lync Call routing to use the Asterisk as a Gateway. This assumes you have enabled users for enterprise voice which is a fairly well documented process: http://technet.microsoft.com/en-us/library/gg413011.aspx
After users are enabled, go to the Topology builder and browse the Standard Server. Check the box for Enterprise Voice

Edit the properties and go to the Mediation Server. Enable Collocated Mediation Server. Define your Listening Ports and click new gateway enter the IP address of the Asterisk and the Port that it is listening for SIP traffic on.

Next associate the Gateway with the mediation server

Publish the Topology.

2. Configure Dial Plan and Trunk. Open Lync Control Panel and go to Voice Routing then Trunk configuration open the newly added Gateway and change the Encryption support level to Optional, Uncheck Media Bypass, Uncheck Centralized Media Processing and Uncheck Enable Refer Support.

3. Add a translation rule to call 4 digit extensions on the IPO via the Asterisk. This allows a normalized call from the Lync server to pass just 4 digits to the IPO so it correctly routes to the extension on the IPO.
Starting Digits: +12355565
Length: Exactly 12
Digits to remove: 8
This rule tells the Lync server to simply pass 65xx to the IPO.

You will also need to create a translation rule to pass all digits without the +
Starting Digits: +
Length: Exactly 12
Digits to remove: 0
This rule tells the Lync server to pass 11 digits to the Asterisk.

4. Create a Call Route. Select New Route and name it and add a description. Leave the Pattern to match the default “*” which matches all calls.

5. Scrolling down select Add for Associated Gateways and select the PSTN Gateway. Do not yet associate a PSTN Usage. But confirm the Gateway is added.

6. Create a Site Voice Policy Choose new and select the site you want to add a voice policy for. Add a Description and enable all appropriate features. Then New.

Associate the route just created in step 6 by hitting select

choose the route.

Go back to Routes and edit the Asterisk PSTN route and scroll to the bottom and Associate the PSTN Usage created.

Commit all Changes.

Configure the Asterisk Box

Finally you need to configure the Asterisk.

1.   First Configure the SIP Trunks
   Login as root to the asterisk server and enter: nano –w /etc/asterisk/sip.conf
   Your configuration should be as follows:
   [General]
   bindport=5060
   bindaddr=0.0.0.0
   tcpbindaddr=0.0.0.0
   tcpenable=yes

   [Lync_Trunk_Name]
   type=peer
   port=5068
   host=0.0.0.0 (where 0.0.0.0 is the ip address of your lync front end server)
   dtmfmode=rfc2833
   context=name-of-lync-context (use what ever name you want)
   qualify=yes
   transport=tcp

   [Avaya_Trunk_Name]
   type=peer
   host=0.0.0.0 (where 0.0.0.0 is the ip address of your ayava IPO)
   dtmfmode=rfc2833
   context=name-of-avaya-context (use what ever name you want)
   port=5060
   Transport=tcp
   Hit Ctrl-X and choose to save

   
   

2.   Next Define your Dial plan to forward all calls.
   enter nano –w /etc/asterisk/extensions.conf
   Your configuration should be as follows:
   [Name-of-lync-context]
   exten => _+1xxxxxxxxxx,1,Dial,(SIP/Avaya_Trunk_Name/${EXTEN},45)
   exten => _+12xx,1,Dial,(SIP/Avaya_Trunk_Name/${EXTEN},45)
   exten => _1xxxxxxxxxx,n,Hangup()

   NOTE:
   Line 1 passes PSTN calls from lync to the PSTN
   Line 2 passes 4 diget extensions dialed from the Lync to IPO

   
   [Lync_Trunk_Name]
   exten => _+1xxxxxxxxxx,1,Dial,(SIP/Lync_Trunk_Name/${EXTEN},30)
   exten => _+41xx,1,Dial,(SIP/Avaya_Trunk_Name/${EXTEN},30)
   exten => _1xxxxxxxxxx,n,Hangup()

   NOTE:
   Line 1 passes PSTN calls and all Lync Extensions WITH DID to Lync
   Line 2 passes 4 digit extensions dialed from the IPO that don’t have a DID.
   
   Exit and Save the configuration

   

   One item to note, the value of 45 is the seconds the phone rings before disconnecting the call.  We had to change the default of 30 to 45 because when someone would call a cell phone FROM Lync via the IPO PRI the call sometimes wasn’t getting to the cell phone voicemail before the 30 seconds and would drop the call before the Lync caller could leave a voicemail for the person they were calling.  After adjusting this value above 30 these dropped calls stopped happening.

3. Reload the Configurations
   Enter: asterisk –r
   Enter: reload

   After the config reloads enter: /sip Show peers
   your status for both SIP trunks should show “OK”

   You are new ready to make calls from lync to the PSTN and place calls on hold.

This post is a continuation of a series of posts about Lync Deployment.   This can be used as a resource to configure an Avaya IPOffice (IPO) 412 (software version 5.0) as a Gateway for a Lync deployment calling the PSTN, but your mileage may vary.  Calls are routed over a SIP Trunk (Session Initiation Protocol) configured between the IPO and the Lync Front End server.

An ISDN/PRI trunk provides inbound and outbound voice call access to the PSTN. Avaya IP
Office sends and receives SIP Invites to and from Lync Standard Server, Lync converts call signaling between standard SIP and Microsoft signaling protocol (MTLS).

The flow for an outbound call from an Enterprise Voice Lync User routes as the following: When an user dials a number,Lync normalizes the dialed number. If there is a match,
Lync checks that the number called is assigned to another Lync user. If so, Lync sends the call to the called user’s Lyc client. If not, Lync looks up a call routing table for a match of the
E.164-formatted called number. If there is a match, Lync routes the call to the Gateway for that route, which in this configuration is the IPO and then the IPO routes the call to the PSTN.

For inbound calls from the PSTN, Avaya IP Office receives the incoming call. Based on the
called party number,IPO looks up the corresponding Short Code (if the called number is a Lync Extension) and routes the call to the Lync server via SIP.

For this configuration an inbound call hits an IPO Inbound call route, matches the last 4 digits to a 4 digit short code which routes to an ARS table which matches the short code digits translates to E.164 format and routes the call over a SIP trunk to the Lync frontend server.

Configuration was modified from an OCSR2 & IPO document found here:
https://devconnect.avaya.com/public/download/interop/OCSR2-IPO-PSTN.pdf

Updated 2/5/2012

When configuring Lync and IPO directly as noted in the white paper above, hold may not function and disconnect the call.  Additionally calls originating from the the PRI on the IPO or an IPO homed extension when transferred to a lync extension cannot be placed on hold or transferred to any other extension (lync or avaya).  The work around used to resolve this issue is SIP proxy as noted here: http://jasonmlee.net/archives/447

Configuring Avaya IPOffice

1. Verify Avaya SIP Trunk license. Login to the IPO Manager application.   In the tree view navigate to Licensing and confirm that you have an active SIP Trunk Channel License.  If a valid license is not configured in the IPO calls will not route over the SIP Trunk.  You can purchase IPO 412 license keys from http://dpctelcom.com/

   

2. Create the SIP line for Lync Server. Select Line in the left panel. Right click
   and select New SIP Line. Enter the SIP Domain Name of local Domain in the ITSP
   Domain Name field. Enter the Lync Server IP Address in the ITSP IP
   Address field. Select Remote Party ID in the Send Caller ID field.
   Network Configuration is as follows:
   Layer 4 Protocol is TCP,
   Send Port is the Receive port on your Lync Server in Topology Builder Default is 5060
   Listen Port is the Send port in your Lync server Topology Builder Default is 5060
   Network Topology Info set to NONE

   

3. Configure SIP URI for known caller ID. Go to the URI Tab and and click add.  Create a primary SIP URI. Enter a unique number for the Incoming Group (Line Group 100) and Outgoing Group (Line Group 100) fields. Enter * for the Local URI, Contact and Display Name fields. Use defaults for all other field. Press the OK button.

   

4. Configure SIP URI for Unknown Caller ID. The documentation indicates a need for a SIP URI for calls received from the PSTN with withheld caller ID. However this appears not to be 100% necessary nor applicable, but was configured in our installation. Select the SIP URI tab and click on Add again. Enter another a unique number for the Incoming Group (Line Group 101) and Outgoing Group (Line Group 101) fields. Enter 000000000 for the Local URI, Contact and Display Name fields. Calls received with hidden caller ID from the PSTN will be shown as coming from this number on the Lync client. Use defaults for all other field. Press the OK button.

   

5. Create an Incoming call route for outbound calls from Lync incoming to the IPO over the SIP trunk. (This call can be both IPO extensions or out to the PSTN)  Select Incoming Call Routes in the Left Tree and right click and choose NEW.  Set the Incoming Group ID to the value you set in step 3 as your Incoming Group ID for the SIP URI (Line Group 100).

   

   In the Destinations Tab enter . in the Destination Field and select OK

   

6. Configure a Alternate Route Selection table (ARS) for calls going from PSTN or IPO Extensions to Lync.  The ARS is used to route the call to the SIP Trunk formatted in E.164 format for Lync to receive the calls correctly . Select ARS in the left panel. Right-click and select New. Enter a unique identifier for the route in the Route Name field (e.g. SIP-Lync) and use defaults for all other field on the ARS tab.

   

   Click on Add button and add short code.  Enter a code matching the 4 digits of the Lync Extension you are wanting to call. 
   Deployment of Lync Extensions with DIDs: In a deployment with DIDs of (123)555-65xx with 4 digit extensions in the 6500-6599 range and a Lync server ip address of 192.168.1.100 and the unique Line Group ID of the SIP trunk is 100 the following short code could be used. (note use of the xx and N variables to allow for creating just one short code for 100 DIDs or Extensions)
   For Deployments without DIDs see step 11 below.

   

7. Create a short code to route 4 digit extension calls from IPO to to Lync. 
   This short code allows for 4 digit dialing from the IPO to Lync extensions as well as will allow for inbound call routes to be configured for DIDs that are homed on Lync.
   Select Short Code in the left panel. Right-click and select New. Enter the first 2 digits of the extension range you are wanting to route to Lync followed by xx (example 65xx).  Select Dial for the Feature. Select the SIP-Lync ARS created previously from the Line Group Id drop down list. Enter “65N” for the Telephone Number field. Use default values for all other
   fields. Press the OK button.

   

8. Create a short code to route Lync calls to the PSTN.  This short code will be matched for any number if a Lync user calls the PSTN and the IPO has no extension match, the call will be routed to the PSTN, without the rule, the IPO doesn’t know what to do with digits dialed that aren’t extensions on the IPO.
   Select Short Code in the left panel. Right-click and select New. Enter “?” in the Code field. Select Dial for the Feature. Select the ISDN/PRI line Outgoing Group Id from the Line Group Id drop down list. Enter “.” for the Telephone Number field. Use default values for all other fields. Press the OK button.

   

9. Create a Short Code for each Lync 4 Digit Extension.  For the IPO to be able to route calls or allow Avaya Extensions to dial 4 digits to call a Lync user, each Lync Extension needs to have a IPO Short Code.  In Hybrid environment, you have to let IPO know that this 4 digit extension is not homed on the IPO but rather on Lync for each user.  Variables can’t be used in a hybrid environment because some extensions live on IPO and some on Lync.
   This example is for a Lync user extension 6500
   Select Short Code in the left panel. Right-click and select New. Enter “6500” in the Code field. Select Dial for the Feature. Enter “6500” for the telephone number and Select the SIP-Lync from the Line Group Id drop down list.  Use default values for all other fields. Press the OK button.

   

10.   Create incoming call route for Lync DIDs
    For an example DID (123) 555-6500 extension 6500
    Select Incoming Call Route in the left panel. Right-click and select New.  Select the PSTN’s incoming Group ID in the Line Group ID drop down box.  Enter “6500” in the Incoming Number to match the ICR last for digits.

    

    On the Destinations Tab enter “6500” to point to the short code created in step 7 above and the call will route via the ARS table to the SIP trunk to Lync formatted as +11235556500@192.168.1.100

    

11.   Configure routing for For Lync Extensions without DIDs (as documented here). An ARS entry will have to be created for each Extension since the IPO cannot use variables in the E.164 formatting of the outbound call and Lync requires the call to come in in the +11235556500;ext=4175 format.
    This example extensions 4150-4175 don’t have DIDs but were valid Lync extensions, in order for IPO extensions to call extensions 4150-4175 a short code would be required for 41xx Pointing to the the SIP-Lync ARS Table. (Assuming no other extensions in the 4100 range are homed on the IPO).

    

    Then entries for each extension would need to be added to the ARS table.
    Code: 4150, Feature: Dial
    Telephone Number: +1235556500”ext=4150@192.168.1.100” (the “”s are required to tell IPO that nothing contained in this part of the string is a variable.  Each subsequent extension would need a ARS entry.

    

12.   Configure Lync Call routing to use the IPO as a Gateway.  This assumes you have enabled users for enterprise voice which is a fairly well documented process: http://technet.microsoft.com/en-us/library/gg413011.aspx
    After users are enabled, go to the Topology builder and browse the Standard Server.   Check the box for Enterprise Voice

    

    Edit the properties and go to the Mediation Server.  Enable Collocated Mediation Server. Define your Listening Ports and click new gateway enter the IP address of the IPO and the Port that it is listening for SIP traffic on.  

    

    

    Publish the Topology.

13.   Configure Dial Plan and Trunk.  Open Lync Control Panel and go to Voice Routing then Trunk configuration open the newly added Gateway and change the Encryption support level to Optional, Uncheck Media Bypass, Uncheck Centralized Media Processing and Uncheck Enable Refer Support.

    
    

14.   Add a translation rule to call 4 digit extensions on the IPO.  This allows a normalized call from the Lync server to pass just 4 digits to the IPO so it correctly routes to the extension on the IPO.
    Starting Digits: +12355565
    Length:  Exactly 12
    Digits to remove: 8
    This rule tells the Lync server to simply pass 65xx to the IPO.

    

15.   Create a Call Route. Select New Route and name it and add a description.  Leave the Pattern to match the default “*” which matches all calls.

    

    Scrolling down select Add for Associated Gateways and select the PSTN Gateway.  Do not yet associate a PSTN Usage.  But confirm the Gateway is added.

    

16.   Create a Site Voice Policy  Choose new and select the site you want to add a voice policy for.  Add a Description and enable all appropriate features.  Then New.

    

    Associate the route just created in step 15 by hitting select

    

    choose the route.

    

    Go back to Routes and edit the AVAYA PSTN route and scroll to the bottom and Associate the PSTN Usage created.

    

    Commit all Changes.

After these steps you should be able to make calls via the IPO as a Lync Gateway.

This post isn’t in the planned sequence of documenting the Lync Deployment in this series, but I found the topic fairly frustrating and undocumented today so I decided to go ahead and post this now.  Our primary location has DIDs for each extension, but or second campus only has a few POTs (Plain Old Telephone) lines for service so there are not DIDs for each extension.

Lync Extensions without the use of DIDs (Direct Inward Dial)
When deploying Lync Enterprise Voice each user is configured with a SIP Address as well as a telephone Line URI.  In deployments where every extension has a DID the Tel URI can simply be the external DID number associated with that user.

When you make a 4 digit extension call internally, Lync uses your defined Dialing Rules and normalizes the number to the E.164 format.  When dialing extension 5555 Lync would normalize (because you configured this normalization rule already) it to: +112355555555 for a US telephone number of (123) 555-5555 and will route the calls internally to the appropriate user.  Since the call matches a Lync user the call isn’t routed to the PSTN (Public Switched Telephone Network).

When a user doesn’t have a DID, you can also enter a user’s Tel URI with the extension added  in the following format: +112355555555;ext=1234 where the main telephone number is (123)555-5555 and the extension is 1234.

Even though you have created the user with the main number and extension you won’t be able to make 4 digit calls without adding additional dialing rules so the call can be completed.

To make calls to 4 digit extensions that do not have DIDs go to Lync Server Control Pannel > Voice Routing and select the appropriate Dial Plan. Once you are viewing the appropriate dial plan choose new “Associated Normalization Rule”.  Give the new Rule a Name and Description. Then skip all the boxes for Starting Digits, Length, Digits to Remove and Digits to add and go to Pattern To match and select Edit.

This example will allow dialing for 4 digit extensions starting with 12## associated with the main number (123) 555-5555
(extensions 1200-1299)
The Value for “Match this Pattern” is: ^(12\d{2})$
The Translation Rule is: +11235555555;ext=$1

After you save and Commit the Rules and they replicate to your Lync Clients you will now be able to dial 4 digit extensions that don’t have a DID.

Since most hotels have Wi-Fi or you can usually find a fairly cost effective internet café. The quest for the ability to call any US landline or mobile phone from my mobile device when there is Wi-Fi available has begun.

An alternative is needed since Google voice simply re-routes your calls using GV still uses minutes on a mobile phone as well as requires phone service from your carrier. (Calling from GV redirects the call their phone number and then routes the call from GV to the person you are calling…)

I have found no direct SIP provider that offers free calling to the PSTN (Public Switched Telephone Network), but was able to find a SIP provider that allows free incoming calls… Enter GV Call Back, SipDroid, and SipGate and Google Voice… with those combined you have Free SIP calling anywhere you have a Wi-Fi connection.  Not to mention inbound calling from anyone who has your Google voice number.

Here is the basics:
Using an Android application called Google Voice Call Back you can initiate over an internet connection a Google voice call.  Google Voice then calls you back on your SIP line which then alerts your phone.  Once you answer the SIP call on your mobile device, Google Calls the person you want to talk to, and you are connected via your device on Wi-Fi to someone on their telephone (mobile or Landline).

Here’s how you set it up:

1. Download and install Google Voice Call Back
2. Download and install SIPDroid 
3. Setup a Free Sipgate One Account with SipGate (60 Free outbound minutes and unlimited incoming calls, but you won’t be using any of the outbound calling minutes so it really doesn’t matter)
4. Acquire a “local” US number from SIPGate by entering your zip code.  It doesn’t matter if this number isn’t a local number for you since you won’t be calling this number nor with anyone else.
5. Login to your Google voice account and go to Voice Settings. 
6. Add an additional number and enter your newly acquired SipGate telephone number. (you will be prompted to verify your new Google voice number, but a few more steps need completed first)
7. Back at your SipGate Dashboard, go to settings and then Click on “Voicemail, Call Forwarding &Hunting” and delete the forwarding settings.
   (this will allow for the Google voice call to ring your phone without SipGate voicemail picking up the call before you do on your mobile device)
   
8. Go to “Phone” in the settings of your SipGate Account, Mouse over your IP Phone and select “Sip Credentials” 
   
9. Note the registry, SIP-ID and SIP-Password as you need those in the next steps.
   
10. Launch SipDroid on your phone and press menu and Go to settings
    
11.   Select the first “SIP Account” (Line 1)
     
12. Enter your SIP-ID as the Authorization Username and enter your SIP-Password as the Password.
13.   Select server  or proxy and change from pbxes.org to sipgate.com (leave all other settings as the defaults)
    
14. Scroll down and select which networks SipDroid can use.

    

15. Launch the GV Call Back application and Set “When to use call back” to either use for all calls or ask for every call.
16. Enter your Google Voice username and password.
17. Set the Callback number to your sipgate number.
18. Select phone type as mobile.  Apply Settings.

    

19. You have now configured GV Call Back, SipDroid, and SipGate and Google Voice. 
20. Launch SipDroid and wait for the Yellow indicator to turn Green in the Status Bar.  After the indicator turns green you are able to answer SIPDroid Calls. 
21. Go Back to the Google Voice Settings page and initiate the test call to validate your SIPGate Number.  Your Android Device should begin ringing. Hit the keypad button and enter the code on the dial pad.
    
22. Once your number has been validated, you are ready to make Calls.  With the Google Voice Call back application enabled, and SIPDroid running, go to the phone dial pad and make a call.  GV Call back will indicate it is making a connection

    

23. A few seconds later you will notice the Green handset in the status bar and then the following screen will display.   The first number is your SIPGate Number from which you are receiving the inbound call, the second number is the caller ID of your Google Voice number. (in the case that someone is calling your Google voice number, this line will display the caller ID of the person calling your Google Voice Number.)

    

24.   During the call you will see a screen similar to the incoming call (with the addition of the dial pad icon to enter any touch tones during the call)

    

25.   Once the call is ended the following screen will display and you can resume normal usage of the device.

    

Apple documents the ability to turn off the .ds_store files here: http://support.apple.com/kb/ht1629 but isn’t totally complete in the instructions so I have documented the process here.

1.  Open Terminal.
2. Change Directory 
    cd ~/library/preferences
3. Write the plist file with the following command: 
    defaults write com.apple.desktopservices DSDontWriteNetworkStores true

4. Read the plist with the following command
   defaults read com.apple.desktopservices DSDontWriteNetworkStores
4. Either restart the computer or log out and back in to the user account.

This applies the setting to the current user but does not impact any other users.  To apply this to all future users who login to the machine copy the .plist file from the user directory to the user template with this command:

Sudo cp ~/library/preferences/com.apple.desktopservices.plist
  /system/library/UserTemplate/English.lproj/Library/Preferences

If you need to disable .ds_store files for an existing user use this command:

Sudo cp ~/library/preferences/com.apple.desktopservices.plist
/users/"username"/library/Preferences