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Step-by-step Microsoft Lync 2010, Asterisk and Skype installation/integration guideOctober 9th, 2010 | Tags: Asterisk, Lync 2010, Lync Server 2010, Skype, Skype for Asterisk, Skype for SIP

Once a year I give my “blessing” to the wife to go away on a long weekend with the girls and usually Itry to call in a few child minding favours from my parents/in-laws and this weekend, thank goodness, isno exception to the rule!

Last time I was given these days of peace I wrote a Trixbox/Exchange 2010 integration guide, theemphasis was on this becoming the first in a series of how-to’s – however this never really came tofruition, the reason? Asterisk + friendly UI = Bad bad bad…so from here on in I have chosen to move toAsteriskNOW.

Trixbox is a great distribution of Asterisk, however it does break certain Asterisk standards and youcan’t beat a good ol’ command line – yes in Asterisk’s case the command line is easier than a webinterface.

So why not plain old Asterisk? AsteriskNOW makes light work of the install and I’m by no means a Linuxguru! You can still opt for the FreePBX front end – but we will choose to not go down this dark path –trust me on this!

So let’s talk objectives…

1. Setup AsteriskNOW, configuring a SIP extension and corresponding dial-plan2. Install and configure Skype for Asterisk (SFA), ensuring the SIP extension above can route in/out

(SkypeOut)3. Take the Lync 2010 Server install performed here and integrate it with AsteriskNOW

Make calls to and from the Asterisk SIP extension (Lync & SFA)Make calls to and from the Lync client (SIP & SFA)

So here is an idea of how this will all piece together:

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Sounds like a tall order right? Wrong. With AsteriskNOW and Lync Server 2010, it is reasonably straightforward and I will endeavour to document the end-to-end setup process.

Before I begin let me talk about SFA…

Skype as you may or may not be aware offers two SME level VoIP integrations:-

1. Skype for SIP (now re-branded as Skype Connect) – essentially a way of integrating Skype’s cloud ofPSTN in/out connectivity including the capability to call Skype users (22.5k online as I type) to an SIPenabled IP PBX – for supported vendors see here.

2. Skype for Asterisk (SFA) – an add-on Asterisk channel driver which allows for Skype-to-Skype callsand access to Skype’s uber cheap calling rates via your Asterisk end-point.

If you are already running an Asterisk based PBX you will probably want to know the difference. From ahigh level it comes down to the following:-

1. Cost – Skype Connect is subscription-based, you pay $6.95 per channel plus calling costs – not cheapfor those who want to use this for a lab sized implementation.

2. Functionality – SFA is not channel-based, it is user-based, for a one off charge of $66 you get a singleuser license – sounds a bit more digestible, right? A single license would give you one channel. In thisguide we will enable a single license be configured to route out from either SIP or Lync end points.From an inbound perspective you could create a Lync response group or Asterisk call group tobroadcast inbound calls to multiple users.

One (or should i say three?) last caveat before we get on with the good stuff:-

1. Lync is currently in release candidate, it is unlikely to change on a grand scale, but be aware it is notsupported by Microsoft

2. Lync (or OCS) + Asterisk integrations are not supported by Microsoft3. This is a “just for fun” guide or lab setup only

Okay, with that over with let’s look at requirements

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Okay, with that over with let’s look at requirements

1. I’ll be using Windows Server 2008 R2 with Hyper-V to run Lync Server 2010 RC & AsteriskNOW2. I have assigned 2gb of memory to Lync Server 2010 RC and 512mb to AsteriskNOW (I know this

seems minimal but it is enough for this small test setup)3. You’ll need to setup a Skype business account as SFA will not work with regular consumer accounts

(you can route Skype-to-Skype calls between business and consumer accounts)4. Once you have setup a free Skype business account you’ll need credit as without credit it won’t route

out to PSTN. I suggest you test the account by adding it to a Skype software client first (if you hit anyroadblocks further down the line you’ll be pleased to have ruled this potential issue out)

5. Buy an SFA single channel license which can be purchased directly from Digium, the makers ofAsterisk, via their online store (currently at $66) – you’ll get a licence key that we will activate later…

Let’s begin…

Download a copy of AsteriskNOW, I have opted for the 64-bit version here, whilst this is downloading (itis approximately 600mb), let’s setup our VM.

Create a name:

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Set memory:

Don’t connect it to your virtual network, we’ll need to create a legacy network adaptor as we are usingLinux

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I’m going to accept the default options when configuring my virtual disk (this isn’t usually recommendedfor performance, but for AsteriskNOW it’ll be sufficient)

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Lync Software Updates Center

Updates for Lync Server 2013: January

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Once your bootable AsteriskNOW ISO is downloaded, select this as the operating system to boot fromwithin Hyper-V Manager

Updates for Lync Server 2013: January2014

Cumulative updates for Lync Server2010: January 2014

Cumulative update for Lync 2010:January 2014

Cumulative updates for Lync PhoneEdition (for Aastra 6721ip and Aastra6725ip): January 2014

Cumulative updates for Lync PhoneEdition (for HP 4110 and HP 4120):January 2014

Cumulative updates for Lync PhoneEdition (for Polycom CX500, PolycomCX600, and Polycom CX3000): January2014

Cumulative updates for Lync PhoneEdition (for Polycom CX700 and LG-Nortel IP Phone 8540): January 2014

Security Update for Lync 2013:December 2013

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Our summary, click Finish

December 2013

Cumulative updates for Lync for Mac2011: December 2013

Security update for Lync 2010 Attendee(Administrator level installation):December 2013

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Classifications Exchange2007 Exchange2010 Google Hyper-V

iPhone iPod Laptop Hunters

Lync 2010 Lync

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Before we kick off the install, you’ll need to go to your settings and add one hardware component, thelegacy network adaptor mentioned earlier – and make sure this is connected to your virtualnetwork

Lync 2010 LyncServer 2010 Microhoo!

Microsoft Microsoft Retail MUCUGL

OCS 2007 OCS2007 R2 Office 365 Office

2007 Office 2010 Outlook 2007

Outlook 2010 PlayStation3 Polycom

Security Skype Skype for Asterisk

Speaker TechEd UC Expo UnifiedCommunicationsUser Group Virtualisation VoIP

Windows 7 Windows

Server Windows Server 2008

Windows Server 2008 R2Windows Vista

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Let’s start our VM! Fingers-crossed the AsteriskNOW ISO will boot and the install commences, selectoption 5 – Asterisk 1.6 only (we need Asterisk 1.6 for TCP support, a SIP trunk requirement for OCS andLync)

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Select yes, to accept the creation of partitions and wiping of data

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The default partition scheme is fine, select next. Set your region, select next and create a root (orAdministrator) password then click next.

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The install will now commence, in my case I’m going to grab some breakfast…!

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I’m back and the install is complete, eject your ISO via the Hyper-V toolbar, Select Media -> DVDDrive -> Eject. Then click reboot

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During the boot sequence you’ll see a ton of text, which will all (hopefully) end with [ OK ]. Uponcompletion you will see a Setup Agent, quit this and you will be presented with the screen below.

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Login with your root account and start the network configuration utility, type system-config-network. You will then be presented with the screen below

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Edit your network device “eth0 – Digital Equipment Corporation DECchip 21140 [FasterNet]“, removethe DHCP option and set a static address, in my case 192.168.10.30

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Save and now edit your DNS configuration. In my case I have a local DNS server (192.168.10.253) butset this to suit your needs. I have also set my host name as ast.jacobs.local (jacobs.local is my localdomain name) and set the search to my local domain. Then Save & Quit.

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Once you are back to the command line type: shutdown –r now this will result in a reboot of thesystem. Once the system has completed a restart, login again as root. You should now be able to pingfrom this system to another address on your local network including the Internet (to check this, pingwww.bing.com to ensure internet connectivity is working) – if you can’t then something has gonewrong! In some cases I have found that you need to head back into the network settings and re-inputthe DNS, this issue shouldn’t re-occur.

Now you are back to your Asterisk command line, type yum install register, you’ll be prompted todownload the package, accept this by inputting Y and hitting enter. YUM is an abbreviated word forYellow dog Updater Modified, it is a command line package management tool. The Register package isused to activate your SFA license.

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Next we’ll update our version of AsteriskNOW, type yum update asterisk16, accept the upgradedpackages (as detailed within our previous step)

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Once completed, we’ll install SFA, type yum install asterisk16-skypeforasterisk, accept the packagedownload. Once installed reboot, type shutdown –r now.

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Once the reboot has completed login and we’ll register the SFA module (using the license key receivedfrom Digium). Login as root and type, register. Select option 1 (Digium Products), then option 7 Skypefor Asterisk. Next you will be prompted to enter you SFA key, enter the key and register now. Click thespace bar to run through the license agreement (does anyone read these?) and accept theagreement. Complete your personal details and your license should be written to/var/lib/asterisk/licenses/ (you should back this up – I’ll explain this next). But one last reboot first, typeshutdown –r now, our Asterisk install is now complete, next we’ll need to configure it!

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To configure our Asterisk PBX we need to edit a number of text files, there are Linux-based editors, butin my experience they are not that user friendly and I like to perform this remotely using my friendlyWindows PC, to do this you need an SSH client – I use WinSCP. Download a free copy via theirwebsite here. Once installed add a new site (see below)

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Then change the default remote directory (as per the illustration below) and save the site.

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Once saved click login. The first time you connect you will be prompted to save your Asterisk serverkey, click yes to add the host key to your cache. Finally enter your password, once connected anexplorer type view of your Asterisk’s file system will be displayed – I have changed to a detailed view(choose view -> details)

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Back-up your SFA license by copying the file from /var/lib/asterisk/licenses – there should only beone .lic file in there, just right click and copy to your desktop to save locally. Now head back to thefolder /etc/asterisk – this is where our Asterisk config files are located.

We will be editing three files: (right click and “edit” within WinSCP)

1. Sip.conf – for main Asterisk settings (trunks/extensions)2. Extensions.conf – for dial plans3. Chan_skype.conf – SFA settings

First sip.conf, replace the content of your file with the following settings (you should probably backupyour original conf files before)

[general]context=default ; Default context for incoming callsallowoverlap=no ; Disable overlap dialing support. (Default is yes)udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)bindport=5060

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bindaddr=0.0.0.0tcpenable=yes ; Enable server for incoming TCP connections (default is no)tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)srvlookup=yes ; Enable DNS SRV lookups on outbound callsnotifyhold = yes ; Notify subscriptions on HOLD state (default: no)

[1001] ; A locally attached SIP extensiontype=friendcallerid=1001canreinvite=nodtmfmode=rfc2833mailbox=1001disallow=allallow=ulawtransport=udpsecret=passwordhost=dynamiccontext=default

[Lync_Trunk] ; Our Lync trunktype=friendport=5068 ; This is the default Lync Server TCP listening porthost=192.168.10.29 ; This should be the IP address of your Lync Serverdtmfmode=rfc2833context=from-lyncqualify=yestransport=tcp,udp

Next extensions.conf, replace the content of your file with the following settings

[general]static=yeswriteprotect=no

[globals]

[default]

;dialling other extensions starting with 1 followed by three digitsexten=>_1XXX,1,Dial(SIP/${EXTEN},20)

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exten=>_1XXX,n,hangup();send every digit after 9 to Skype for Asteriskexten=>_9.,1,Dial(Skype/${EXTEN:1},20)exten=>_9.,n,hangup()

;dialling other extensions starting with 2 followed by three digitsexten=>_2XXX,1,Dial(SIP/Lync_Trunk/${EXTEN},20)exten=>_2XXX,n,hangup()

[from-lync]

;dialling other extensions starting with 1 followed by three digitsexten=>_1XXX,1,Dial(SIP/${EXTEN},20)exten=>_1XXX,n,hangup();send other calls to Skype for Asteriskexten=>_.,1,Dial(Skype/${EXTEN},20)exten=>_.,n,hangup()

This dial plan will enable call routing between Lync , Asterisk & SFA. Finally configure yourchan_skype.conf

You will need to change the default_user to represent your Skype ID, in my case:

;default_user=james_bond changed to (general section)

default_user=imapcblog

update the password

;secret=goldeneye changed to (user section)

secret=myskypepassword

change the default context to match your dial plan

;context=demo changed to (user section)

context=default

route the inbound calls to your Lync extension, in my case 2001

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;exten=s changed to (user section)

exten=2001

Once you have configured all three config files reboot Asterisk, type shutdown –r now. Congratulation,your Asterisk configuration is complete! Next we need to configure Lync and pre-supposing you followedmy previous Lync install guide here, you will need to head back into the Lync Topology Builder – wedidn’t add a PSTN gateway previously. Download your Topology from the existing deployment andsave the file locally, then add a PSTN gateway (see both steps below)

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Once you have added the PSTN gateway you will have to re-publish the Topology, this will update theexisting Topology with the new configuration settings, click finish

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To permit call flow between both Lync and Asterisk worlds we need to define our Voice Routing withinLync Server 2010. Open the Lync Server Control Panel and access the Voice Routing options, we’llneed to configure our Dial Plan, Voice Policy, Route and PSTN Usage. I won’t walk you through thisconfiguration (some is based upon location preferences, in my case UK), but I will show you theresulting configuration within the Lync Control Panel.

First the dial plan summary (pay close attention to my normalisation rules) – this will route 1xxx toAsterisk, +44xxxxxxxxx to Asterisk and 2xxx internally (treated as local extension)

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Next the Voice Policy, the default Global Policy should have the following PSTN usage records (theserules will handle our calls destined for Asterisk)

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and the routes, as follows

Finally the PSTN usage, as follows

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In my previous guide we created a test user, that user should now have his or her telephone detailsset, in our scenario tel:2001 (this is not best practice in a production deployment of Lync/OCS, but as alab setup with a single PSTN number we should be given a free pass!) typically in production we wouldassign individual PSTN number and normalise to an internal DDI range i.e. +44208 555 2001 wouldnormalise to 2001 when dialled.

With these settings committed successfully your setup should be complete, calls can now be madebetween both Asterisk and Lync, to setup a SIP based client I recommend X-Lite (for download andsetup instruction follow my previous guide here)

Here we can see X-Lite to Lync

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And the reverse Lync to X-Lite

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Finally, let’s SkypeOut, first from Lync! (this illustration won’t mean a lot, but you will have to trust me –it works!)

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Profile

That’s it, whilst I’m sure there are better ways of achieving PSTN breakout from OCS or Lync (withhardware/gateways), there is a certain amount of self-gratification from 100% software based VoIP.

I’m sure my configuration could have been applied in a number of different ways and you’ll probablynotice that whilst X-Lite will permit Asterisk-to-Skype calls (dial 9 + Skype name), at this time Lync willnot – only numbers can be passed (I’m happy to take suggestions on this). Otherwise let me know ifyou have spotted any errors or need guidance on issues (I’ll be moderating the comments below) andabove all good luck and have some fun!

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[...] This post was mentioned on Twitter by Tom Arbuthnot, Jan S. Andreassenand Mike Pfeiffer, Adam Jacobs. Adam Jacobs said: I'm a UC Blog: Step-by-stepMicrosoft Lync 2010, Asterisk and Skype installation/integration guidehttp://retwt.me/1P8gQ #lync #asterisk [...]

Great and inspiring article.Just at this week I have talk with client. He wanted to integrate Skype with OCS(Lync).Now I can see the way to do it.

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October 9, 2010

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Thank you, great article!

[...] Step-by-step Microsoft Lync 2010, Asterisk and Skype installation/integrationguide | I’m a UC Blog Posted on October 11, 2010 by johnacookhttp://imaucblog.com/archive/2010/10/09/step-by-step-microsoft-lync-2010-aste… [...]

Nice article. I’m currently looking for any Remote Call Control help on Lync.

@Doug Hi Doug, are you referring to RCC functionality within Asterisk?

Thanks for this – much appreciated!

Nice job

Awesome Doug, let me try it and wait for my comments..

[...] [...]

I’m having some trouble routing calls even between Lync and Asterisk. How do I

Step-by-step Microsoft Lync 2010, Asterisk and Skypeinstallation/integration guide | I’m a UC Blog « JC’sBlog-O-Gibberish

October 11, 2010

Doug October 11, 2010

Adam [I'm a UC Blog] October 11, 2010

Deane October 12, 2010

Khani October 12, 2010

Diego October 14, 2010

Asterisk 1.8.0 is released to web | I'm a UC Blog October 23, 2010

Fredrik October 28, 2010

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configure the mediation server? I cant route from the Asterix out to skype either itseems, but first thing’s first, the internal route.

Thank you for your great blog posts BTW!!

@Fredrik Can you make calls Asterisk ext to ext? Which version of Asterisk are you using?

Hey there, I am having trouble with the dial in. It answers fine, put in the conf id,thats fine. When it try to join the confernce, it says i can’t connect to theconfernce. In the event viewer, all I get is:

User failed to join the conference.

Microsoft.Rtc.Collaboration.ConferenceFailureException:The operation failed due toa response from the server. For more information, examine the properties on theexception and inner exception.at Microsoft.Rtc.Signaling.SipAsyncResult`1.ThrowIfFailed()at Microsoft.Rtc.Signaling.Helper.EndAsyncOperation[T](Object owner,IAsyncResult result)at Microsoft.Rtc.Collaboration.McuSession.EndSendCommandInternal(IAsyncResultresult)atMicrosoft.Rtc.Collaboration.AudioVideo.AudioVideoMcuSession.EndTransfer(IAsyncResultresult)at Microsoft.LiveServer.Caa.CaaCall.EndTransfer(IAsyncResult asyncResult,Boolean& retry, Exception& caught)Detected at System.Environment.get_StackTrace()at Microsoft.Rtc.Collaboration.ConferenceFailureException..ctor(String message,Exception innerException)atMicrosoft.Rtc.Collaboration.Conferencing.SendCommandAsyncResult.ProcessCccpResponse(SipMessageDatamessageData, responsetype response, Boolean& isPendingResponse)atMicrosoft.Rtc.Collaboration.Conferencing.SendCommandAsyncResult.ProcessStatusMessage(SipMessageDatastatusMessageData, responsetype response)

Adam [I'm a UC Blog] October 28, 2010

Calum MacRawe November 23, 2010

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atMicrosoft.Rtc.Collaboration.Conferencing.StatusMessageReceivedWorkItem.Process()at Microsoft.Rtc.Signaling.AsyncWorkitemQueue.ProcessItems()at Microsoft.Rtc.Signaling.SerializationQueue`1.ResumeProcessing()at Microsoft.Rtc.Signaling.SerializationQueue`1.ResumeProcessingCallback(Objectstate)atMicrosoft.Rtc.Signaling.QueueWorkItemState.ExecuteWrappedMethod(WaitCallbackmethod, Object state)at System.Threading.ExecutionContext.Run(ExecutionContext executionContext,ContextCallback callback, Object state)atSystem.Threading._ThreadPoolWaitCallback.PerformWaitCallbackInternal(_ThreadPoolWaitCallbacktpWaitCallBack)at System.Threading._ThreadPoolWaitCallback.PerformWaitCallback(Object state)Cause: Administration issues.Resolution:Verify that Conferencing Attendant is installed in a supported topology and thedependant Front End servers are functioning correctly.

Help please!!!

I’m wondering if I need to include Mediation server into this scenario or just PSTNGateway… Please, let me know…

@TreeFox The mediation service is deployed during the install, for PSTN termination agateway or SIP trunking is required.

Call forwarding and simul ring to PSTN/Asterisk number does not work with thissetup.

TreeFox November 23, 2010

Adam [I'm a UC Blog] November 24, 2010

BlaNon November 30, 2010

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@BlaNon Actually, just the Asterisk extension.Try forwarding your OCS extension to PSTN or Asterisk extension. If you dial fromthe phone connected to Asterisk to the OCS extension, the call will not beforwarded. OCS sends the “INVITE” with from=@lyncserver, and Asterisk will notauthorize it, except if you set insecure=invite on the Asterisk extension.There must be a better way tho.

For me trunk between Asterisk & Lync is not showing UP…

asterisk*CLI> sip show peersName/username Host Dyn Nat ACL Port Status1001/1001 192.168.100.100 D 29082 UnmonitoredLync_Trunk 192.168.100.101 5060 UNREACHABLE2 sip peers [Monitored: 0 online, 1 offline Unmonitored: 1 online, 0 offline]

I am getting below error for my Lync 2010 RTM trunk. I have configured PSTNgateway with TCP 5060 port without any issues.. I am able to make calls betweenasterisk extensions & as well as in between Lync extension… is I am missinganything?????

[Dec 1 11:11:36] ERROR[3294]: tcptls.c:350 ast_tcptls_client_start: Unable toconnect SIP socket to 192.168.100.101:5060: Connection refused

Now I am able to get my trunk UP,But not able to make calls in between Asterisk & Lync vice versa…

asterisk*CLI> sip show peersName/username Host Dyn Nat ACL Port Status1001/1001 192.168.100.100 D 51890 Unmonitored1002/1002 192.168.100.102 D 5060 UnmonitoredLync_Trunk 192.168.100.101 5060 OK (2 ms)

3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 0 offline]== Using SIP RTP CoS mark 5— Executing [2001@default:1] Dial(“SIP/1001-0000000a”,

BlaNon December 1, 2010

Johnny December 1, 2010

Johnny December 1, 2010

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“SIP/Lync_Trunk/2001,20″) in new stack== Using SIP RTP CoS mark 5— Called Lync_Trunk/2001— SIP/Lync_Trunk-0000000b is circuit-busy== Everyone is busy/congested at this time (1:0/1/0)— Executing [2001@default:2] Hangup(“SIP/1001-0000000a”, “”) in new stack== Spawn extension (default, 2001, 2) exited non-zero on ‘SIP/1001-0000000a’

Getting “SIP/Lync_Trunk-0000000b is circuit-busy” error

Can you call out from Lync or Asterisk? Which scenarios work for you?

@Johnny I noticed an error in my Asterisk SIP.CONF, the default Lync Server TCP listeningport is 5068. Change your port and you should be all good (fingers crossed!)

Thanks Adam for this Awesome post. Now I am able to make calls from Asterisk toLync extension without any issues. Problem was with my Lync extension telephonenumber previously I used default format (i.e. tel:+2001) that was causing theproblem.. I have changed extension to tel:2001 & it works…

Still I have one issue, I am not able to make calls from Lync extension to Asteriskextension, I am working on it…

I would like if you post more on Lync Dial Plan.. This will help newbies like me…

Cheerssss!!!!!!!!!!!!

Hi Johnny,

This is more than likely a dial-plan issue (as correctly identified) – did you permitcalls to route to 2xxx extentions via your Lync trunk configuration?

- Adam

Adam [I'm a UC Blog] December 1, 2010

Adam [I'm a UC Blog] December 1, 2010

Johnny December 2, 2010

Adam [I'm a UC Blog] December 3, 2010

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Thanks Adam,

It works by adding PSTN gateway route for Asterisk extensions.

Cheerss!!!!!!!

Adam,

Much thanks. You got me rolling on Lync. Now to play and really dig in.

Going to do a little more on your Asterisk integration. I’m pretty familiar withAsterisk and FreePBX (trixbox, AsteriskNow, even my own UI). Perhaps we canadd a little more on that config. to a real world scenario for those using FreePBX.Should be able to get some alone time in a week or so.

Hi Adam, I’m glad you have picked up Lync! If you have anything you’d like tocontibute please let me know – happy to run a “special guest” article

All the best, another Adam

@Adam

Adam,

This is almost exactly what i’m looking to do and as a proff of concept was able toget things working with this. However we currently have a decent sized asteriskimplementaion that hooks to the pstn. I want to hook things in to make mymigration seemless but all the articles i’ve seen say rip out all that is in the cfg filesright now.

Do you or anyone else have some guidance on how to integrate lync into anexisting asterisk deployment?

Hi Jim,

Johnny December 8, 2010

Adam December 12, 2010

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You may be able to take elements of my code and merge with your existing config(taking into account changes to suit your environment). Is there a specific areaswhere you are struggling?

- Adam

Hello,

i geht the same errorERROR[31807]: tcptls.c:367 ast_tcptls_client_start: Unable to connect SIP socketto 172.16.1.155:5068: Connection refused

but if i make telnet 172.16.1.155 5068 (to my Lync Server)i get the following messageTrying 172.16.1.155…telnet: Unable to connect to remote host: Connection refused

can anybody help me ?

Adam,

I’m setting up Lync in our test environment. I don’t have my SIP trunk yet, but dohave enterprise voice enabled. I have X-Lite 4 downloaded but cannot get it toconnect to the Lync server. A sniffer shows X-Lite trying to connect to port 5060as expected, but when I go to my Lync server, Netstat shows that it is notlistening on port 5060? Did I miss a step in installation/setup? I have no devicesdefined in the Clients tab.

Also could you share a screenshot of your X-Lite account setup page so that I cancheck my syntax?

Thanks for your help and for a great article.

Bryan Hunt

Hi Bryan, you can’t connect the x-lite client to Lync (only the Lync client can beused here) – configure it to point to the Asterisk server.

Goofy December 16, 2010

Bryan Hunt December 16, 2010

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- Adam

Hi Goofy, is the firewall running on the server? Are all the Lync services running?

- Adam

Thanks for the response Adam. The Lync client is working perfectly.

Does this imply that MS plans for Lync to be a somewhat “closed” platform as faras SIP devices go? In other words, only phones and soft clients with firmwaredesigned specifically for Lync will be able to connect to it? Not all devices that“speak” SIP?

Thanks.

Bryan Hunt

@Bryan Hunt Glad to hear it is working Bryan!

Now onto your excellent question, OCS/Lync does utilise SIP, but as with mostmajor telephony vendors it has been “enhanced” to include product specificfunctionality – as such device firmware needs to support the OCS or Lync SIPspecification. Typically these devices will only work with these Microsoft products,however there are signs of change – one example are Snom’s IP Phones, thesehave dual firmware support. See here for more information.

I hope this helps?

- Adam

Hi Adam, have you perhaps experiences that PSTN Media Gateways for HomeOffice and are available and working with Lync? Audiocodes and Ferrari SBA’s areout of the question. These are too expensive for me just for testing @ home. Alsoa matching card from Divacom (for example Diva BRI-2 PCI v2, Diva V-BRI-2 PCI

Adam [I'm a UC Blog] December 16, 2010

Bryan Hunt December 17, 2010

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v2) eliminated for cost reasons.

Thanks for suggestions,

Ralf

Hi Ralf,

I am playing around with an AudioCodes MediaPack 114 gateway (chances are I willprobably upload config guides in due course) – this device isn’t what I would callexpensive approximately £250?

- Adam

Thank you Adam for the tip with the small audio codes device and of course a bigcompliment for your How-Tos on this Site. Thanks especially for the Lync AsteriskHowto.

Ralf

@Johnny

Hi Adam!

Where do I add the PSTN gateway route for Asterisk extensions?

ThanksChris

Chris :@Johnny Hi Adam!Where do I add the PSTN gateway route for Asterisk extensions?ThanksChris

Solved – On Lync Server under Voice Routing -> Route -> Add Associatedgateways

Adam [I'm a UC Blog] December 19, 2010

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I just implemented this for OCS 2007r2, and, after a few modifications, it workslike a charm!

When I bring up the Lync Control Panel, under Topology, what should my AsteriskPBX status be listed as?

Right now I am able to communicate internally between Lync Clients and Tanguaydesk phones, but I seem to be having some troubles integrating Asterisk.

I currently use PBX-In-A-Flash for my Asterisk needs. It is Asterisk v1.8 withFreePBX v2.8.

I changed the dial plan to the following (since I live in North America, and myAsterisk extensions are in the 7xx range):http://img24.imageshack.us/img24/500/18534005.jpg

Right now when I try to call externally through my Tanguay phone I get a “Callunsuccessful. Cannot complete the call due to restrictions on your account” error.

And when I try to call through the Lync Client I get this:http://img443.imageshack.us/img443/540/captureodf.jpg

Not sure where to start in troubleshooting this problem.

Any help would be appreciated.

Thanks.

Hi Wayne,

First things first is there a SIP trunk successfully established betweenLync/Asterisk? In Asterisk you can check this via the CLI – on Lync the serverevent log will report trunk related errors. Also ensure your PBXIAF is set to allowTCP – this is not enabled by default.

Let me know how you get on.

- Adam

Michael December 22, 2010

Wayne December 24, 2010

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Hi Adam,

I am using Linksys 3102 as my asterisk gateway to make outgoing calls to externalworld.. I am able to make calls in between asterisk & Lync extension vice versawithout any issue.Also I am able dial outside numbers i.e. local mobile number from my asteriskextension by dialing 8 then local mobile number & phones .. But i am not able todial 8 + local mobile number/phone number via Lync client…I have addedd requiredroute on lync control panel..is I am missing anything?

thanks in advance..

Cheers!!!!

Hi Johnny,

There a number of ways this can be achieved, my lab setup was configured toautomatically prefix Lync PSTN routes with “9″ as per my config – this will route viaSFA. I hope this helps?

- Adam

@Calum MacRawe – did you ever get this figured out…I’m having the exact sameissue on the final release of Lync.

Calum MacRawe :Hey there, I am having trouble with the dial in. It answers fine,put in the conf id, thats fine. When it try to join the confernce, it says i can’tconnect to the confernce. In the event viewer, all I get is:User failed to join the conference.Microsoft.Rtc.Collaboration.ConferenceFailureException:The operation failed due toa response from the server. For more information, examine the properties on theexception and inner exception.at Microsoft.Rtc.Signaling.SipAsyncResult`1.ThrowIfFailed()at Microsoft.Rtc.Signaling.Helper.EndAsyncOperation[T](Object owner,IAsyncResult result)at Microsoft.Rtc.Collaboration.McuSession.EndSendCommandInternal(IAsyncResultresult)at

Adam [I'm a UC Blog] December 31, 2010

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Microsoft.Rtc.Collaboration.AudioVideo.AudioVideoMcuSession.EndTransfer(IAsyncResultresult)at Microsoft.LiveServer.Caa.CaaCall.EndTransfer(IAsyncResult asyncResult,Boolean& retry, Exception& caught)Detected at System.Environment.get_StackTrace()at Microsoft.Rtc.Collaboration.ConferenceFailureException..ctor(String message,Exception innerException)atMicrosoft.Rtc.Collaboration.Conferencing.SendCommandAsyncResult.ProcessCccpResponse(SipMessageDatamessageData, responsetype response, Boolean& isPendingResponse)atMicrosoft.Rtc.Collaboration.Conferencing.SendCommandAsyncResult.ProcessStatusMessage(SipMessageDatastatusMessageData, responsetype response)atMicrosoft.Rtc.Collaboration.Conferencing.StatusMessageReceivedWorkItem.Process()at Microsoft.Rtc.Signaling.AsyncWorkitemQueue.ProcessItems()at Microsoft.Rtc.Signaling.SerializationQueue`1.ResumeProcessing()at Microsoft.Rtc.Signaling.SerializationQueue`1.ResumeProcessingCallback(Objectstate)atMicrosoft.Rtc.Signaling.QueueWorkItemState.ExecuteWrappedMethod(WaitCallbackmethod, Object state)at System.Threading.ExecutionContext.Run(ExecutionContext executionContext,ContextCallback callback, Object state)atSystem.Threading._ThreadPoolWaitCallback.PerformWaitCallbackInternal(_ThreadPoolWaitCallbacktpWaitCallBack)at System.Threading._ThreadPoolWaitCallback.PerformWaitCallback(Object state)Cause: Administration issues.Resolution:Verify that Conferencing Attendant is installed in a supported topology and thedependant Front End servers are functioning correctly.Help please!!!

@Goofy

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- firewall is down- Restart “Lync Server Mediation service” or all Lync service after publishing thetopology- you could be able to telnet your lync server on 5068

This guide is absolutely awesome. I used your Lync step-by-step guide to get abasic Lync server up and running and I’ll probably use this to get some awesomeintegration going.

Thanks for your feedback!

Hi adamfirst i want to say this and the previous guide for exchange um are amazing:)the coolest thing i ever saw:)BUT in this guide i just wanted to test lync and asterisk going(no skype)so i did exactly what you said(besides registering skype part)and i just cant get this thing to work:(i cannot dial not from the asterisk and not from lynci get network busy and busy dial tone.do you have any idea what i am missing here:)?(can provide access to my lab if u have time:))Thanks

Turbomcp :Hi adamfirst i want to say this and the previous guide for exchange umare amazing:)the coolest thing i ever saw:)BUT in this guide i just wanted to testlync and asterisk going(no skype)so i did exactly what you said(besides registeringskype part)and i just cant get this thing to work:(i cannot dial not from theasterisk and not from lynci get network busy and busy dial tone.do you have anyidea what i am missing here:)?(can provide access to my lab if u havetime:))Thanks

Rosewood January 18, 2011

Adam [I'm a UC Blog] January 18, 2011

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small update, i rebooted the lync server and now i can call from lync to asteriskusing 1001but i still cannot dial from asterisk and x-lite to lync, i get busy signal like before

Hi there, firstly thanks for the feedback!

Chances are you have issues with your call routing, specifically within Lync. Doesyour Lync enabled user have an extension assigned? When calling what is theerror reported within the Asterisk CLI – type asterisk -r on your Asterisk commandline. Let me know how you get on.

- Adam

mm,my user name is testand his line uri is tel:2001on the asterisk i dont see any specific error(i think) besides:unable to connect sip socket to 192.168.10.4:5068: network is unreachable.so i checked my lync server(forgot i had to reinstall my asterisk)my subnet is 192.168.25.0255.255.255.0i changed the lync server entry ofcourse to point to my lync server which is192.168.25.4and set all other options and rebooted and still busy signal from x-lite 2001 tolync.is there something else i need to do like register the 2001 number at asterisks(thisis a brand new lab so not the one used for exchange um)

Turbomcp :mm,my user name is testand his line uri is tel:2001on the asterisk idont see any specific error(i think) besides:unable to connect sip socket to192.168.10.4:5068: network is unreachable.so i checked my lync server(forgot ihad to reinstall my asterisk)my subnet is 192.168.25.0255.255.255.0i changedthe lync server entry ofcourse to point to my lync server which is 192.168.25.4and

Adam [I'm a UC Blog] February 2, 2011

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set all other options and rebooted and still busy signal from x-lite 2001 to lync.isthere something else i need to do like register the 2001 number at asterisks(this isa brand new lab so not the one used for exchange um)

quick question(i know its stupid but still)when im configuring the x-lite phone i configure it to register/login to the asteriskas 1001 right?so when i dial 2001 how does it know to forward that to lync because of the trunkand rule we confiured?another thing:on my lab i only have 2 things under pstn usage and not 4 like your printscreenshowsi have local route and local extensions(both as global)

now for the weirdest thing:)i added another xp machine with test2 user.duplicated the 1001 to be 1002(so i have 2 locally attachwed extensions)i configured it and logged on and bamthat one works perfect:)how weird is that:)i dial 1001 from lync get to x-lite dial 2002 from x-lite get to 1002:)at this point im clueless as to what the heck is going on with the first user:)

found it:)i configured user one(test) with topologhy tabstupid i knowanyway when is set this thing to none all is good.Thanks againyour hard work is highly appreciateddaily watcher of this web site:)

Used this article (which is brilliant by the way) and have got skypeout working fine,however I have got an online number setup on the skype account but having some

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issues routing the inbound to the lync client.

Here is what I see in the debug…

[Feb 21 20:13:49] NOTICE[3362]: core.cpp:2146 sfa_call_hangup: ending call— Executing [2001@default:1] Dial(“Skype/{skype-signin-name}-1f762d98″,“SIP/Lync_Trunk/2001,20″) in new stack== Using SIP RTP CoS mark 5— Called Lync_Trunk/2001— SIP/Lync_Trunk-00000006 is circuit-busy== Everyone is busy/congested at this time (1:0/1/0)— Executing [2001@default:2] Hangup(“Skype/{skype-signin-name}-1f762d98″,“”) in new stack== Spawn extension (default, 2001, 2) exited non-zero on ‘Skype/{skype-signin-name}-1f762d98′

Any help would be much appreciated!

@Craig Gauntlett

Hi Craig, now if you are trying to get assistance on account of your flattery thenyou are going the right way about it!

First off have you given the extension number 2001 to your Lync client? Next howhave you triggered your trunk, the busy tone usually indicates a trunk/routingissue.

- Adam

Dear Adam, I have interesting issue.

My Setup:“Asterisk” “Lync” “AudiocodesMP”No direct routes between “Asterisk” and “AudiocodesMP”, both configured as voicegate in Lync.

From Lync I can call extensions on both “Asterisk” and “AudiocodesMP”.

From “AudiocodesMP” I was not able to call any extensions on “Asterisk” until i’veconfigured New-CSAnalogDevice account in Lync for “AudiocodesMP” (TEL:+900).

Adam [I'm a UC Blog] February 21, 2011

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Now using +900 as CallerID I can call any number from “AudiocodesMP”. Whencalling +900 from Lync I can reach “AudiocodesMP”.

But when I call +900 from “Asterisk”, always get “SIP 404 Not Found” error. Whilemonitoring Asterisk SIP log, i’ve noticed what when I call +900 from “Asterisk”,Lync is trying to call +900 extension on “Asterisk” instead of “AudiocodesMP”.

I’ve tried to create CSAnalogDevice account for “Asterisk”, when use its TEL:+950us CallerID when I call from “Asterisk” to TEL:901 (which is routed to“AudiocodesMP”, but this number is not assigned to any Lync account). This Call isrouted. But in this case i’m loosing my original Asterisk CallerID and this is notacceptable. Also, call to +900 still fails.

How can I solve my problems: enable calls from “Asterisk” to “AudiocodesMP” thru“Lync” and keep original CallerID?

Wow Igor, this one threw me a little (let’s put it this way I had to get Visio out!)

Q. For you, why are you not using the AudioCodes as a gateway for Lync andAsterisk? If you took this approach you could establish cross extension capabilityvia a Lync to Asterisk trunk.

- Adam

I’m currently testing, but is there any way for Asterisk and Lync to share the sameextension? What I’d like to accomplish is to ring our existing Asterisk IP deskphone and ring/show the incoming call on the users Lync client, basically asimultaneous ring using the same extension. Can’t figure out how to do it or if it’seven possible.

Thanks!

Would a simultaneous ring to yor Lync extension not achieve this? I don’t havethe code to hand, let me know if I am missing something on this?

Adam [I'm a UC Blog] March 20, 2011

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Does Skype Connect or SFA (described in article) allow two-way audiocommunication between cloud of Lync users and cloud of Skype users?

What about the same question about messages exchange between that clouds?

Hi Alex,

Unfortunately Skype does not allocate “cloud-based” DIDs, therefore Lync isunable to normalise the calls for Lync-2-Skype purposes (Asterisk can handlenames so this is not an issue here). Potentially you might be able to setup aspeedial within Asterisk and assign a number that could be reached via Lync butthis starts to get complex.

The biggest win is being able to utilse Skype’s PSTN for outbound and for inboundyou can use Skype-in or your Skype ID for other Skype based clients.

I hope this helps?

- Adam

Thanks, Adam,

Surely, we got an comfortable long distance prices from our provider, so costoptimisation via Skype is not an issue.

Problem is that users are requiring Skype to talk, chat and use videoconferencewith contragents and to minimise long-distance expenses, but this requires Skypeclient installation and interfere with security policies. Lync 2010 instead, fits thecorporate security, meets all the internal corporate needs, but we’re stuck withSkype interoperability.

Potentially, there are number of solutions to integrate Skype and Lync, forexample Skystone and Skystone Video. Will evaluate it.

Let me know how you get on Alex.

- Adam

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Hi,

I am trying to set up a test environment (XLite–>Asterisk–>MS Lync Server 2010–>Lync Client) similar to that in this excellent guide,but I’m having trouble calling from XLite to the Lync client.

1. According to wireshark on MS Lync Server, the header checksum of the InternetProtocol in each SIP packet sent by MS Lync Server is 0×0000.2. I’m also getting the following error according to Lync Server Logging Tool,:TL_WARN(TF_COMPONENT) [2]143C.133C::04/04/2011-22:45:37.959.00000746(SIPStack,CSIPRequest::RouteRequestUriOnNonEdgeProxy:SIPRequest.cpp(3441))(0000000006370440 ) User [2001@appserver.com] is not in enterprise or we donot serve this domain – using 404TL_WARN(TF_DIAG) [2]143C.133C::04/04/2011-22:45:37.977.00000747(SIPStack,SIPAdminLog::TraceDiagRecord:SIPAdminLog.cpp(145))$$begin_recordLogType: diagnosticSeverity: warningText: Non-trusted source sent an FQDN/IP that doesn’t match a routing table ruleResult-Code: 0xc3e93c5e SIPPROXY_E_ROUTINGSIP-Start-Line: INVITE sip:2001@appserver.com;user=phone SIP/2.0SIP-Call-ID: 5f803ac8-ece1-47f8-9ac4-0981e5b3b617SIP-CSeq: 57 INVITEData: user=”2001@appserver.com”$$end_record

Any idea what might be causing these problems? Thanks.

Hi Max,

Stating the obvious first, have you properly assigned the extension no? Next haveyou looked at the Asterisk logging? Type asterisk -r via the command prompt. Isthe trunk up? (again you can check this via the asterisk command prompt “sipshow peers”)

Let me know how you get on?

- Adam

Max April 5, 2011

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Hi, thanks for a great guide, just walked through this and got it all working,Thanks!

The only problem I ran into was that I got no audio from the Skype client to Lync,X-lite -> Lync and mobile (via SkypeIn) -> Lync where fine. The fix I found was toadd the following to to chan_skype.conf in the user section

allow=ulaw,alaw

-Stephen.

I have disabled the firewall on Lync Server now the error is

Unable to connect SIP socket to 192.168.22.243:5068: Connection refused

Thanks for posting your workaround Stephen – was this not enabled by default?

- Adam

Hi Syed,

Apologies for not getting back to you sooner (I was on vacation) and thanks foryour previous message, given that Windows Firewall is disabled you may want tocheck the ports – sometimes there can be some confusion on incoming/listeningports.

I’ld suggest you follow defaults and refer to this TechNet documentation fordefault port definitions.

- Adam

[...] for Skype (R.I.P. my Skype-to-Lync integration blog post) – okay so it was aworkaround and PIC would be far [...]

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I don’t have a Chan_skype.conf file in the asterisk directory. Is this something Ishould create manually?

Hi David,

This is usually created when you add SFA – is this installed correctly?

- Adam

Far as I can tell. I added the License Key for the SFA and that didn’t give me anyerrors and it created the License File.

I’d strongly suggest re-installing something has gone wrong. This shoulddefinately be created as a part of the install.

- Adam

[...] http://imaucblog.com/archive/2010/10/09/step-by-step-microsoft-lync-2010-asterisk-and-skype-installa… [...]

GREAT POST!!!!!! Thanks to your hard work I got my environment up andrunning. Lync users can call X-Lite. X-Lite users can call Lync. X-Lite users canconnect out via SIP. Lync users can connect out via SIP. All working like a charm.Not that it happened easily. Some changes were required.

As I’m using another SIP provider and not Skype my dial plan looks like this:[general]static=yeswriteprotect=no

David May 11, 2011

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David May 11, 2011

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Step-by-step Microsoft Lync 2010, Asterisk and Skypeinstallation/integration guide : Lync Guru

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[globals]

[default]

;receive incoming calls to a connected phone

exten => s,1,Dial(SIP/1001)

; outbound calls (outside of your own PBX)

exten => _1XXX,1,Dial(SIP/${EXTEN},20)exten => _1XXX,2,hangup()

exten => _0.,1,Dial(SIP/${EXTEN:1}@2talk)exten => _0.,2,hangup()

exten => _2XXX,1,Dial(SIP/Lync_Trunk/${EXTEN},20)exten => _2XXX,2,hangup()

[from-lync]

;dialling other extensions starting with 1 followed by three digitsexten=>_1XXX,1,Dial(SIP/${EXTEN},20)exten=>_1XXX,n,hangup()

;send other calls to 2talk for Asteriskexten => _.,1,Dial(SIP/${EXTEN}@2talk)exten => _.,2,hangup()

This way all calls get routed nicely.

I found following commands very useful:enter Asterisk: asterisk -rReload all config files: reloadReload only the dial plan: dialplan reload

rebooting my asterisk server takes AGES. So reloading the configs is much muchfaster. Especially when testing different dial plans.

Thanks Alex (sorry for not getting back to you sooner), really appreciate yousharing this info for other readers.

- Adam

Adam [I'm a UC Blog] July 18, 2011

Andres July 22, 2011

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Hi Adam

Nortel CS1000M/Great post, I will try this on a topology Elastix – Cisco\Nortel BCM50

I guess it will be enough setting up the Lync-Asterisk part, I only have my doubtsin the following:

1. Is it possible to have trouble using Asterisk 1.8?2. Is there any codec support issue with Lync, like using codec g.729?3. Will this have the same behavior when calling to an Asterisk Queue? because I’musing Elastix call center module and it works based on queues.4. Is there a how-to for connecting Lync Server with Microsoft Dynamics CRM andOutlook?

If someone has any suggestion on this to be of consideration I will be great full ifyou share before I start deploying. I shall share the results and hopefully goodanswers to all this.

Thanks again for the great post.

Andres :Hi AdamNortel CS1000M/Great post, I will try this on a topology Elastix – Cisco\Nortel BCM50

hehe I guess it did’nt show up as I meant. I will try this in a topology with Elastixworking with Cisco, Nortel CS1000M and Nortel BCM50. Wish me luck xD

Hi Andres,

A1. Asterisk 1.8 is absolutely possible for Lync IntegrationA2. Lync supports g.711 ulaw/alaw – (supported by Asterisk too)A3. I have not used Elastix, but I don’t see why not?

Andres July 22, 2011

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A4. So far as I am aware only an OCS -> CRM 4 out of the box integration isavailable at this time, see here

- Adam

I am really trying to work through this in my lab. I am running 2008 R2 SP1 Hyper-v server. The Linux Integration Tools do not install at all!

I can install Centos 5.2 fine but when I get to the integration tools I cannot accessthe command line nor install the tools. Any ideas?

Hi Bob,

I didn’t need to install the Linux integration tools, sorry.

- Adam

Hello Adam pls i will like if u can suggest to me a solution to this i got this fromeventvwr

The Mediation Server service has received a call that does not support comfortnoise. This event is throttled after 5 calls from a single Gateway peer.

The Mediation Server service has received a call that does not support comfortnoise from the Gateway peer, 192.168.10.105Cause: The Gateway peer does not support comfort noise.Resolution:Please ensure the comfort noise option on the Gateway has been enabled.

I cant call from my xlite to lync but i can call from lync to xlite

Hi Olay,

The comfort noise issue is unlikely to be the root cause, suggest as a first port ofcall you look into the Asterisk routing error. Access the Asterisk command line andexamine the logging generated wen you try to initiate a call failure. The command

Bob K August 10, 2011

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line can be accessed by typing “asterisk -r” via the terminal.

- Adam

hi AdamsThanks a lot i av gptten the error but what am i to dothe error is tcptls.c 350 ast_tcptls_client_start unable to connect sip socket to192.168.10.20:5060 : no route to host

Within the Asterisk CLI type “sip show peers” – is your Lync trunk up?

- Adam

AdamI type the command but it returns an error (-bash: sip : command not found)

Hi Ojay,

Try the Asterisk CLI, this is enabled by typing “asterisk -r” from your Linuxterminal.

- Adam

HI Adam i am so sorry for disturbing you. i can get the sip peers now the trunk isunreachable. pls go through my configuration

[general]context=default ; Default context for incoming callsallowoverlap=no ; Disable overlap dialing support. (Default is yes)udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds toall)bindport=5060

ojay August 16, 2011

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ojay August 17, 2011

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bindaddr=0.0.0.0tcpenable=yes ; Enable server for incoming TCP connections (default is no)tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to allinterfaces)srvlookup=yes ; Enable DNS SRV lookups on outbound callsnotifyhold = yes ; Notify subscriptions on HOLD state (default: no)

[1001] ; A locally attached SIP extensiontype=friendcallerid=1001canreinvite=nodtmfmode=rfc2833mailbox=1001disallow=allallow=ulawtransport=udpsecret=passwordhost=dynamiccontext=default

[Lync_Trunk] ; Our Lync trunktype=friendport=5068 ; This is the default Lync Server TCP listening porthost=192.168.10.20 ; This should be the IP address of your Lync Serverdtmfmode=rfc2833context=from-lyncqualify=yestransport=tcp,udp

Name Host Port StatusLync_trunk 192.168.10.20 5068 unreacheable

My PSTN GATE is using port 5060My Mediation Tls listening port:5067My Mediation TCP listeneing port:5068

i even try to change the PSTN gateway port to 5068 all to no avail all i can get is acall from lync to xlite and not the other way round

Adam [I'm a UC Blog] August 22, 2011

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Ojay,

You config looks good, something is stopping your Lync trunk from beingestablished (from the Asterisk to Lync) – this is the root cause. Not sure if I haveasked, do you have Windows Firewall enabled, could this be blocking the port?Also is 192.168.10.20 the IP address of your Lync server?

- Adam

Here is my issueI have freepbx and OCS 2007r2I just upgraded to Lync in parralelnote my connect-with-OCS trunk works finemy connect-with-Lync trunk for some reason does notI did reset the TCP listener from 5068 to 5060 on Lync and published it andverified ityet when I route from-internal to Connect-with-Lync context I get no loveNo firewalls are on and there on the same subnet.

If i revert to the working Connect-with-OCS trunk here is what works and whatdoesntand is my real issue.from PBX sip client (PSC) to Lync user (LU) ext to ext work finefrom LU to PSC client ext to ext work finefrom LU to dial out PBX to my mobile phone works fineBLOCKERon the LU i set call forwardingOCS to LU call forwarding works to mobile fineLU to LU call forwarding works to mobile finePSC client then calls LU ext with call forwarding enabledand it rings tries to route and fails.

So first issue is How do I get Lync to accept traffic from freepbxwhen it has the same settings of my OCS trunks and configsand my context from internal is pointed to the right trunk?

How do I get Lync to forward calls from a PSC client out tothe public mobile number.

Should I just revert and try to use port 5068 and see what happens?

bacmallard August 31, 2011

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SIP trace on Lync shows the forward traffic happening but my guess isThe routing rules did not result in a final response for PSTN caller and non-UMenabledcallee”;source=”site1pool.foo.local”;Callee=”jdaugherty@foo.com”;appName=”InboundRouting”

Im just thinking since lync anc ocs calls forward correctly that somehowI have to tell Lync or OCS to look at the PCS client number and that its ok toforward it off.

Hi Bacmallard,

I’ll be honest you have lost me a little! Nevertheless I would definately suggest yourevert to 5068, I have this configuration setup and working with the default portconfiguration.

- Adam

@Bob K

How I got Hyper-V Integration Tools to work with AsteriskNOW (Hyper-V R2 SP1Host)Download http://www.microsoft.com/download/en/details.aspx?id=24247 andextract .exeMount extraced .iso file in Hyper-V

*Login as root*yum -y groupinstall “Development Tools”yum -y updateyum -y install kernel-devel*Reboot**Login as root*mkdir -p /mnt/cdrommount /dev/cdrom /mnt/cdromcp -rp /mnt/cdrom /opt/linux_icunmount /mnt/cdromcd /opt/linux_ic/make

Adam [I'm a UC Blog] August 31, 2011

TiamaT September 9, 2011

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make install*Reboot**Done*

Btw: Fantastic article. Now I can finally get some real hands-on experience withEnterprise Voice.

Adam, just attempted to purchase license for AsteriskNOW and it seems that it isno longer available? Am I missing something on there website?

BOO!

Why is Skype for Asterisk no longer available?Skype for Asterisk was developed as a result of an agreement between Digium andSkype to allow distribution of Skype proprietary software. This software enabledAsterisk to make use of proprietary Skype protocols and participate as a nativeclient on the Skype network. Skype decided not to renew this agreement in 2011,so Digium had to cease sales of the product. Skype for Asterisk will be supporteduntil July 26, 2013.

http://www.digium.com/en/docs/SFA/sfa_faq.php

Dang it!!!!

Hi Ben,

AsteriskNOW is free, download links here

- Adam

Error : SIP/Lync_Trunk-00000004 is circuit-busy Please help me….

Reloading SIP— Unregistered SIP ’1001′— Registered SIP ’1001′ at 192.168.1.47 port 23436

Ben September 15, 2011

Ben September 15, 2011

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Bikash September 17, 2011

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[Sep 17 17:41:37] NOTICE[3195]: chan_sip.c:23658 sip_poke_noanswer: Peer‘Lync_Trunk’ is now UNREACHABLE! Last qualify: 1[Sep 17 17:41:47] NOTICE[4035]: chan_sip.c:18912 handle_response_peerpoke:Peer ‘Lync_Trunk’ is now Reachable. (3ms / 2000ms)== Using SIP RTP CoS mark 5— Executing [2861@default:1] Dial(“SIP/1001-00000003″,“SIP/Lync_Trunk/2861,20″) in new stack== Using SIP RTP CoS mark 5— Called Lync_Trunk/2861[Sep 17 17:41:48] WARNING[4035]: chan_sip.c:17726 reply_digest: missingDigest.[Sep 17 17:41:48] NOTICE[4035]: chan_sip.c:18451 handle_response_invite:Failed to authenticate on INVITE to ‘”1001″ ;tag=as790ee8f5′— SIP/Lync_Trunk-00000004 is circuit-busy== Everyone is busy/congested at this time (1:0/1/0)— Executing [2861@default:2] Hangup(“SIP/1001-00000003″, “”) in new stack== Spawn extension (default, 2861, 2) exited non-zero on ‘SIP/1001-00000003′localhost*CLI>[Sep 17 17:42:51] NOTICE[3195]: chan_sip.c:23658 sip_poke_noanswer: Peer‘Lync_Trunk’ is now UNREACHABLE! Last qualify: 3[Sep 17 17:43:01] NOTICE[4039]: chan_sip.c:18912 handle_response_peerpoke:Peer ‘Lync_Trunk’ is now Reachable. (3ms / 2000ms)[Sep 17 17:44:05] NOTICE[3195]: chan_sip.c:23658 sip_poke_noanswer: Peer‘Lync_Trunk’ is now UNREACHABLE! Last qualify: 3[Sep 17 17:44:15] NOTICE[4042]: chan_sip.c:18912 handle_response_peerpoke:Peer ‘Lync_Trunk’ is now Reachable. (3ms / 2000ms)[Sep 17 17:45:19] NOTICE[3195]: chan_sip.c:23658 sip_poke_noanswer: Peer‘Lync_Trunk’ is now UNREACHABLE! Last qualify: 3[Sep 17 17:45:29] NOTICE[4045]: chan_sip.c:18912 handle_response_peerpoke:Peer ‘Lync_Trunk’ is now Reachable. (3ms / 2000ms)[Sep 17 17:46:33] NOTICE[3195]: chan_sip.c:23658 sip_poke_noanswer: Peer‘Lync_Trunk’ is now UNREACHABLE! Last qualify: 3localhost*CLI>

ThanksBikash

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@ojay I have same problem trunk is unreachable Please help me

HI Adam i am so sorry for disturbing you. i can get the sip peers now the trunk isunreachable. pls go through my configuration

[general]context=default ; Default context for incoming callsallowoverlap=no ; Disable overlap dialing support. (Default is yes)udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds toall)bindport=5060bindaddr=0.0.0.0tcpenable=yes ; Enable server for incoming TCP connections (default is no)tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to allinterfaces)srvlookup=yes ; Enable DNS SRV lookups on outbound callsnotifyhold = yes ; Notify subscriptions on HOLD state (default: no)

[1001] ; A locally attached SIP extensiontype=friendcallerid=1001canreinvite=nodtmfmode=rfc2833mailbox=1001disallow=allallow=ulawtransport=udpsecret=passwordhost=dynamiccontext=default

[Lync_Trunk] ; Our Lync trunktype=friendport=5068 ; This is the default Lync Server TCP listening porthost=192.168.10.20 ; This should be the IP address of your Lync Serverdtmfmode=rfc2833context=from-lync

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qualify=yestransport=tcp,udp

Name Host Port StatusLync_trunk 192.168.10.20 5068 unreacheable

My PSTN GATE is using port 5060My Mediation Tls listening port:5067My Mediation TCP listeneing port:5068

i even try to change the PSTN gateway port to 5068 all to no avail all i can get is acall from lync to xlite and not the other way round

Hi Bikash,

At a guess you have some sort of connectivity/performance related issueimpacting the Asterisk -> Lync trunk.

- Adam

Hi Adam,

Great site. Have you ever tried this with Sipxecs?

I setup a gateway to my sipxecs server and a dial plan that routes the calls (with a46 prefix) but I can’t connect to the server. The client gets a fast busy. One of theerrors I see in event viewer on the server is “There was no response from agateway to an OPTIONS request sent by the Mediataion Server”. Any suggestions?

Thanks for the feedback Mike. I have not played with sipXecs but I know it ispopular, without looking into it is your trunk UP?

- Adam

Thanks for reply Now Asteris is Workinf Fine and Also I have configure Elastix GUIfor LYNC.

Adam [I'm a UC Blog] September 19, 2011

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ThanksBikash

Glad to hear it Bikash!

- Adam

Hi!Thanks for this great guide.I’m totally new to Lync and SIP telephony in general and have just installed Lyncand Asterisk for the first time.

I followed your tutorial though using another SIP provider and was able to makecalls to PSTN from X-lite client as well as from Lync clients for a while.Then this morning I was not able to enter numbers longer than 7 digits excludingthe national prefix (in my case +46).When entering a number in the Lync client it starts searching the address bookdirectly and I am able to press the call button as long as I enter less than 8 digits.But when I enter the 9th digit nothing happens when I press the call button.I’m not sure but it seems like this happend after I entered my phone numbers inthe Lync client configuration. I removed them but the problem persists.

Does anybody have a clue where to start troubleshooting this? Cause I’m clueless.The only thing that differs from your Lync conf is the regexp for national callswhich is ^(\+46\d{7}\d+)$ instead. A typical Swedish mobile phone number is+46734123456.

Any help would be greatly appreciated.

Seems like I solved it.Deleted the GalContacts.db and GalContacts.db.idx files and ran reg addHKLM\Software\Policies\Microsoft\Communicator /v GalDownloadInitialDelay /tREG_DWORD /d 0 /f to force a new address book sync. Also ran the Update-CsAddressBook cmd on the Lync server before starting the client again.And now it seems like I can make >8 digits calls again. Have not been able to try it

Adam [I'm a UC Blog] September 24, 2011

Nils October 18, 2011

Nils October 18, 2011

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since I’m on remote with my lab environment.

Thanks for sharing Nils.

Incidentally you’re doing pretty good considering you’re new to both Lync andAsterisk (my guides are obviously working out too!)

- Adam

Hello Adam and Nils,

Can you please let me know the other SIP vendors which I can use to call PSTN.Also please let me know the configuration.

I am trying to integrate Microsoft Lync 2010 with Trixbox. I have public linepurchased from VoiP operator. I followed guide from this url to configure Trixbox:http://blogs.breezetraining.com.au/mickb/2009/07/31/FinallyConnectedOCS2007R2ToTrixboxAsteriskToAPSTNPBX.aspxI am able to receive call from public network to my Lync client. However I amunable to call from Lync to anybody in outside world.I examined the logs on Lync side. There is and error:Start-Line: SIP/2.0 504 Cannot connect to gateway. Socket error:ConnectionRefusedI also examined logs on Trixbox side with doing: asterisk -r; sip set debug on.When i was placing a call there was no SIP traffic captured on Trixbox server.I did network trace. I noticed that Trixbox is dropping connection from Lync. Lyncsends SYN packet to port 5060 and Trixbox is replying with RST,ACK packet.

Here is trunk configuration from Trixbox to Lync and vice-versa:[Connect-with-Lync]disallow=allhost=10.48.22.182 ; Lync server IPtype=friendport=5068 ; Lync listening portinsecure=port,invitedtmfmode=rfc2833

Adam [I'm a UC Blog] October 18, 2011

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qualify=yestransport=tcp,udpcanreinvite=yesallow=ulawcontext=from-lync

[from-Lync]host=10.48.22.182 ;Lync IPtransport=tcpport=5060 ;Trixbox portinsecure=port,invitetype=friendcontext=from-Lync

Could you please help?

If I understand this configuration correctly you have two Trixbox trunks defined(each for calls in/out) – this is not neccessary and will confuse things. I’d suggestyou create one and configure Lync accordingly.

- Adam

skype…

[...]Step-by-step Microsoft Lync 2010, Asterisk and Skype installation/integrationguide | I'm a UC Blog[...]…

@JOhnny hi, My Lync is still showing unreachable on 5060. can help me

I apologize for perhaps a dumb question here. I am aware that Asterisk basedsolution is about telephony; SIP/IAX/PSTN, etc. Now, Lync, is pure presence/IMplatform. What benefit exactly do I get with integrating the two? If the answer is

Adam [I'm a UC Blog] November 5, 2011

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Jeff November 27, 2011

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presence of the phone and IM’s then I still do not understand, because I can haveOpenFire server on my Asterisk box providing IM/presence for those sameextension via AsteriskIM plugin. Again, please correct me if I am wrong or missingsomething. IF there is a feature I can further enhance my solution, I am open toadditional information. thank you.

I am using Bria which has integration with Asterisk, Outlook, and containspresence, click to dial, IM features, etc. What’s the purpose of having a hugeproduct like Lync integrated with Asterisk? From ease of deployment, licensingcost, complexity of management, it seems to be an overkill.

Hi Jeff,

Lync is whole lot more than just presence and IM, since Office CommunicationsServer 2007 the platform has very much incorporated voice, a/v conferencing andcollaboration. OCS/Lync integration with Asterisk can deliver a number of benefits,such as the leverage of existing Asterisk PSTN break-out. I can also tell you(based upon traffic statistics that there is a very keen interest – around 150vistors daily!

- Adam

Hi Jordon,

Asterisk is a great solution and it certainly offers a lot of capability found withinOCS or Lync, I’d also hasten to add that it would never be my recommendation todeploy, support and maintain both products within a production environment. Itmay however be of value if you are looking to migrate to Lync or leverage existingservices already available within your existing Asterisk deployment i.e. PSTN break-out.

Being familiar with both products I will say that you get what you pay for withAsterisk and Lync is now a mature enterprise-ready product with deep integrationwith the Microsoft stack. Of course if you are a start-up business or ContactCentre running Open Source technology, Asterisk will fit right in.

Jordan Turner November 28, 2011

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I hope this helps?

- Adam

[...] VoIP & Gadgets Blog, VoIP & Gadgets Blog. I came across an excellent tutorialon installing and integrating Microsoft Lync 2010, Asterisk and Skype. The tutorialcovers [...]

[...] Integration Tutorial Posted on December 28, 2011 by VoIP & Gadgets Blog Icame across an excellent tutorial on installing and integrating Microsoft Lync 2010,Asterisk and Skype. The tutorial covers [...]

Adam, a short question – I am trying to integrate 2 x Asterisk as gatewaysassosiated with a Mediation Pool consisting of the 3 FEs (collocated MediationServers) for some failover/redundancy testing. Do you have an idea how toconfigure the peer part of the sip.conf to let Asterisk communicate to all 3 DNSload balanced FEs or is this configuration not achievable at all.Thanks

Hi Paul,

Interesting scenario. I’d probably try using the DNS name as a host (pointed atthe pool name) – not sure how Asterisk will handle this though i.e. will it utiliseround-robin or just fail…

Let me know how you get on?

- Adam

IT Business Reviews » Archive » Microsoft Lync 2010,Asterisk & Skype Integration Tutorial

December 28, 2011

Microsoft Lync 2010, Asterisk & Skype IntegrationTutorial | All Things VoIP - VoIP Phones, VoIP Products,VoIP Services, etc...

December 28, 2011

Paul January 28, 2012

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Hi Paul,

Probably one of the best Lync Asterisk integration guides out there. I’m workingdedicatedly on Lync since past couple of years, but totally new to Asterisk. I’mtrying to do what Alex did in Post #73 below i.e set this up with a Asterisk SipTrunk Provider which unfortunately doesn’t work with Lync directly.

I’m using a Public IP on Asterisk and Lync Mediation and using TCP 5060 forcommunication between Lync and Asterisk. I’m unable to dial from X-lite or Lyncand get an error Unable to create channel of type ‘SIP’ (cause 20 – Unknown)Also sip show peers command shows Lync using 5060 correctly but also shows1001 client also using 5060 and unmonitored, which should be some random portin my opinion.

Alex Dean :GREAT POST!!!!!! Thanks to your hard work I got my environment upand running. Lync users can call X-Lite. X-Lite users can call Lync. X-Lite users canconnect out via SIP. Lync users can connect out via SIP. All working like a charm.Not that it happened easily. Some changes were required.As I’m using another SIP provider and not Skype my dial plan looks likethis:[general]static=yeswriteprotect=no[globals][default];receive incoming calls to a connected phoneexten => s,1,Dial(SIP/1001); outbound calls (outside of your own PBX)exten => _1XXX,1,Dial(SIP/${EXTEN},20)exten => _1XXX,2,hangup()exten => _0.,1,Dial(SIP/${EXTEN:1}@2talk)exten => _0.,2,hangup()exten => _2XXX,1,Dial(SIP/Lync_Trunk/${EXTEN},20)exten =>_2XXX,2,hangup()[from-lync];dialling other extensions starting with 1 followed by threedigitsexten=>_1XXX,1,Dial(SIP/${EXTEN},20)exten=>_1XXX,n,hangup();send other calls to 2talk for Asteriskexten =>_.,1,Dial(SIP/${EXTEN}@2talk)exten => _.,2,hangup()This way all calls get routed nicely.I found following commands very useful:enter Asterisk: asterisk -rReload all configfiles: reloadReload only the dial plan: dialplan reloadrebooting my asterisk server takes AGES. So reloading the configs is much muchfaster. Especially when testing different dial plans.

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Hi Favad,

Is the trunk up between your Asterisk server and Lync? As you previously stated,“sip show peers” will confirm.

- Adam

Hi Adam,

Thank You for your reply. Yes it is up according to the sip show peers commandon TCP 5060. Also when dialing from Lync to X-Lite I get 480 temporarilyunavailable on Lync Mediation and Sip 404 Not found when I call from X-Lite toLync

Thank You

Try “asterisk -r” at the command line and replicate a call to see what is going on –it may be a route issue?

- Adam

Thanks for comprehensive – and funny howto. Great work!

arrrgghh okay so I have installed PIAF purple and trying to move off my old pbxthat I had working with Lync just fine.

Where im at… Lync snooper logs simply state when placing a call to PIAFStart-Line: SIP/2.0 503 Service UnavailableCSeq: 1 INVITECall-ID: 37181e8685284502afb58faa5b279eb4VIA: SIP/2.0/TLS10.15.2.39:57786;branch=z9hG4bK62FB59E9.62B9C8E23E28080F;branched=FALSE,SIP/2.0/TLS

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10.15.16.52:50364;ms-received-port=50364;ms-received-cid=3A49500CONTENT-LENGTH: 0SERVER: RTCC/4.0.0.0 MediationServerms-endpoint-location-data: NetworkScope;ms-media-location-type=intranetms-trunking-peer-state: downms-trunking-peer: 10.15.3.2ms-enable-dns-failover: yesms-diagnostics: 10001;source=”site1pool”;reason=”Gateway did not respond in atimely manner (timeout)”;component=”MediationServer”ms-diagnostics-public: 10001;reason=”Gateway did not respond in a timelymanner (timeout)”;component=”MediationServer”Message-Body: –$$end_record

I have successful results in my sip peers that show OK in my Lync Trunk

My objective as it was before was to have all calls from lync route to either outsideif a public number dialed and be able to call PBX extensions as well.

Im not worried about anything right now but having the Lync be able to connectI did switch my old ports from 5060 to 5068 triple checked Lync ports and configpatched and restarted Mediation services

As long as I can see traffic I can get through the rest. But for the life of me rightnowI cant dechiper why Lync is stating the its trunking peer state is down.

Also in PIAF purple sip.conf is explicity stated not to edit it and to add your configsto sip_general_custom.conf instead

mine looks like this

tcpenable=yestcpbindaddr=0.0.0.0alwaysauthreject=no

[Connect-with-Lync]type=peerhost=10.15.2.39qualify=yestransport=tcp,udpcanreinvite=yesallowexternalinvites=yes

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port=5068dtmfmode=rfc2833disallow=allallow=ulawcontext=from-lync

My Lync trunk has two entries as it did on my old pbx that was working just fineexcept for the port change from 5060 to updated 5068

Connect-with-Lync

Outgoing settingstype=peerhost=10.15.2.39qualify=yestransport=tcp,udpcanreinvite=yesallowexternalinvites=yesport=5068dtmfmode=rfc2833disallow=allallow=ulawcontext=from-lync

INCOMING Settings:User context from-lync

type=peerhost=10.15.2.39qualify=yestransport=tcp,udpcanreinvite=yesallowexternalinvites=yesport=5068disallow=allallow=ulawcontext=from-lync

From this it looks at though your Asterisk server is talking to Lync, but Lync is not

Adam [I'm a UC Blog] May 10, 2012

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talking to Asterisk. Check your Lync SIP trunk out within topology builder – is itset to 5060?

- Adam

ok switched it to 5060 that seems to have done the trickThanks ADAMI got traffic now just on to call routing from here so ill see how PIAF purple placescontexts. Ill add my update here as well.

Question: Subject SIP Trunks:My current sip trunk has only one DID is it possible for a SIP provider to havemulitple DIDs supported on a single sip trunk?

If so whats the average capacity per sip trunk to support multiple call volume. Iknow its based off of internet bandwidth but that aside whats the ball parkcapacity?

Call forwarding from Asterisk to Lync then forwarding calls from Lync back outAsterisk PBX

So say I want Lync users to utilize there client as there soft phone. That userwants to forward calls from Lync to their cell phone. That call will route out myAsterisk box.

I discovered that if you set a password on your Asterisk extension and have Lyncset to Forward the call. It translates the call back to Asterisk as a known extensionfrom an outside source (Lync) in this case that is attempting to say hey Im one ofyour extensions please forward this call.

The only work around I found wich is not ideal but does work is if you do not set apassword on the extension. This way Asterisk does not try to validate a knownextension from outside and allow it to pass the traffic.

Anyone try this or have a different way of engineering it to work with the Asteriskextension and having success forwarding Lync calls out?

bacmallard May 10, 2012

bacmallard May 10, 2012

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Absolutely, you can have a range of DDIs assigned to a trunk. Capacity is basedupon network capability and any associated service provider restrictions.

- Adam

Last at last, Okay so I have everything back on 5060 and now all seems to behappy again.

My last nudge here is. I want to set the follow me settings on the PBX to the LyncExtension.

I can dial the Lync number from PBX just fine but when I set the follow mesettings to5516 (lync extension) or 5516# it fails to forward over. Any ideas on how to getthe follow me to ring to the Lync extension?

@bacmallard Okay so couldnt find any hacks for the follow me to worki set my Lync ext and my mobile in the PBX follow me list and ringall

It would go to Lync but then roll over to the cell

bit of duct tape here

What I did was create to local extensions 5516 and changed the context SIP todial my lync extension over the trunk

I then set up extension 4416 and changed the context SIP to dial my cell phonenumber over the ExternalProvider trunk

set confirm calls

and now everything works just fine with ringall forward me

my office sounds like a ring tone party

[...] would like to give a big thanks to Adam Jacobs who wrote How to integrateLync 2010, Asterisk and Skype as inspiration for this post. This entry was posted

bacmallard May 10, 2012

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How to Integrate Microsoft Lync 2010, Asterisk, and asip trunk. | Sip Stories

June 25, 2012

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@bacmallard

I’m trying something similar with PIAF. I have a working PBX with a Google trunk..works really well. Now I’m trying to get the SIP trunk between Lync and the PBXup but just having a horrible time!. when you said you switched everything to5060 do you mean even the config on Lync? did you switch 5068 to 5060? Causefor the life of me I can’t get the trunk to come up.

Have you enabled SIP over TCP on PIAF?

A

Javier October 25, 2012

Adam [I'm a UC Blog] October 27, 2012