Random Thoughts from TechEd NA 2012

I attended TechEd NA this year, making this the seventh year in a row I attended.  This year, I was a paid attendee, rather than working the Hands-On Lab area, which meant I was able to interact with more booths and see more sessions than usual.  My various unstructured thoughts follow in the hope you will find them useful.

  • Products Featured.  This show was all about two things:
    • 2012 releases – Windows 8, Windows Server 2012, System Center 2012, Visual Studio 2012, and SQL Server 2012
    • “The Cloud” – Office 365, cloud-based management including Windows Intune, and Windows Azure including the new Virtual Machine offerings.  In fact, the custom hotel room keys this year put The Cloud very clearly front and center:
      69e41244-7ea4-46c9-894f-4589fe5e37a0
    • Not really talked about much:
      • Windows Phone (although there was a large show floor presence, there was no track and only a few sessions; expect a lot more next year when Windows Phone 8 is released)
      • Windows 7/Windows Server 2008 R2 (some sessions but not a big push)
      • SharePoint 2010/Exchange 2010/Lync 2010 (sessions but not a big push; expect more next year for the new releases)
      • Office 2010 (again expect more next year when Office 2012 is the hot product)
  • Specific session notes for the ones I attended in person… because a lot of what falls under my “official” subject area was not new material for me (when you work for Bennett Adelson you’re required to be ahead of the curve in your area!) I went a bit out of scope for some of the sessions…
    • AAP313 | Scrum Under a Waterfall (Benjamin Day) – A good discussion of how to do agile (or should I say “Agile” – big “A” – since it was Scrum-focused) in an environment where old school waterfall planning is required.
    • AAP401 | Real World Developer Testing with Visual Studio 2012 (David Starr, Peter Provost) – Ultimately I had mixed feelings about this session. I was not convinced the idea of “submit your problems and we’ll solve them” really worked as only a few problems were gotten to, and my real-world question of “how do you expect developers who can’t afford Visual Studio Ultimate to do these things” wasn’t answered. Yes, it was snarky, but it is a real-life problem faced by many. Further, a lot of time was spent on “that’s the wrong way” coding.
    • DEV370 | Nokia with Windows Phone: Learning How to Tile (David Middleton, David Mason, Kalle Lehtinen) – Ultimately very disappointing as the practical content was virtually zero in my opinion. If this was “DEV170” that would have been okay…
    • VIR317 | Lessons from the Field: 22 VDI and RDS Mistakes You’ll Want to Avoid (Greg Shields) – A good, honest session about real-life implementation of VDI and RDS on Windows Server 2008 R2 and how changes/improvements in Windows Server 2012 help.  Highly recommended if you’re looking at these technologies or have already implemented them.
    • WCL290 | Microsoft Application Virtualization 5.0: Introduction (Andy Cerat, Matthijs Gates) – A very nice intro to App-V 5.0 (part of the upcoming MDOP release) showing some of the great changes and improvements. No more Q: drive? Apps can work with each other (think: Visio available from Word)? Updated, cleaner UI? Check, check, and check!
    • WSV325 | DNSSEC Deployment with Windows Server 2012 (Rob Kuehfus) – Presented by a member of the Wireless Networking and Services team and the owner of the DNS Server offering in Windows Server 2012, this is a nice overview of how DNSSEC works in general and how to use it in Windows Server 2012 (hint: it’s very easy), including practical guidance on the steps to implement in order. Highly recommended if secure DNS is important to you or if you work in an environment where it is mandatory (e. g. US Federal Government).
    • WSV331 | What’s New with Internet Information Services (IIS) 8: Open Web Platform for Cloud (Won Yoo) – A solid presentation on new expansion and control capabilities in IIS 8 including mention of features that have been or will be back-ported to IIS 7.5. Nice demos. Some amazing performance improvements demonstrated – for one case, the first GET on IIS 7.5 with SSL demo took 10.9 seconds and over 500 MB of RAM, while the same page first GET on IIS 8 with the new central file-based certificate store capability took 0.14 seconds (under 1/6 of a second is not a typo) with 44 KB of RAM (again KB not MB is not a typo) – and that was with with 20x as many instances of the site running under IIS 8!
    • WSV332 | What’s New with Internet Information Services (IIS) 8: Performance, Scalability, and Security (Robert McMurray) – A nice companion to the previous session.  Includes discussion of new dynamic security features in the web and FTP (yes, FTP!) service. Also discusses changes in warm-up functionality that can make it possible to show users a “I’m warming up, be with you soon” message while waiting for the ASP.NET hamsters to spin up.
  • Product/Exhibitor Booths – note that the “Attendee” badge gets you treated “seriously” – many exhibitors and Microsoft staff ignore a “Staff” badge holder or are even borderline hostile even at silly things like book signings (I won’t name names but I will say that it takes zero day-s for this to happen) so it’s nice to be treated appropriately for once… 
    • I visited the Windows Phone area to find out what was up with the Windows Phone Summit in San Francisco this week. They all made it sound like a press-only event, despite earlier talk of a two day developer event. The whole thing felt like a bit of a CF to be honest. At least the announcements – which world + dog expect to be all around Windows Phone 8 – will be streamed.
    • I visited the Office 365 area to discuss an issue a customer was having with their proof of concept where Lync Online refused to federate with anyone, even after everything was clearly configured right. It turned out to be some kind of Microsoft-side provisioning screw-up, although the Office 365 booth was not helpful figuring that out.
    • I visited the Windows 8 “Access Everywhere” booth to ask, essentially, “WTF is with Consumer Preview being Professional instead of Enterprise? You know we don’t get DirectAccess with that, right?” The answer was essentially:
      • “yes, we know it sucks, everyone is yelling at us [field people]”
      • “we hope to have some kind of resolution soon, maybe even in the next week, or at least an official acknowledgement that there will be no resolution”
      • “no one seems to know why that was the decision made by the client team”
      • “only TAP people have Enterprise right now.”
        So a major ball drop there.
    • I visited the System Center Enterprise Protection booth to ask, “why does SCEP turn off Security Center on Windows 8 every time the machine boots?” The answer was essentially “it won’t install on Windows 8 prior to CTP2 [note: that is wrong… speaking from office experience here], so do CTP2 and see what happens.” because going from CTP2 to the final Beta or RTM is painful we’re just leaving SCEP off Windows 8 machines right now and using the built-in Windows Defender instead, which gets us the same protection but loses us the management/reporting functionality.
  • Certification!  There were multiple free exams this year – specifically two Private Cloud exams and three beta exams (Windows Server 2012, Windows 8, and Developing with HTML5/CSS3).
    • The Private Cloud exams (70-246 and 70-247) felt tough but fair to me. That said, there were MANY 70-246 failures… so be warned. You really need to have worked with System Center 2012 as a suite and know how the pieces work together and individually to pass these!
    • The Installing and Configuring Windows Server 2012 exam had a nice mix of old and new. You will need to have worked with the product at least a bit, or have been exposed to it a lot without touching it, to have any chance of this one.
    • The Configuring Windows 8 exam also had a nice mix of old and new, and just like the server exam, you will need to have worked with the product at least a bit, or have been exposed to it a lot without touching it, to have any chance of this one.
    • The Programming in HTML5 with JavaScript and CSS3 exam had almost nothing Microsoft-specific on it, and did an okay job of covering the field, although I was surprised on some of what I was NOT tested on (although it’s a beta and I’m sure has a large pool so others may be different!). Of course I can’t tell you more details – test NDA and all 🙂

I am sure I am leaving many things out, but I think this is a reasonably-complete high-level brain dump. Please feel free to comment with thoughts or questions!

— Michael C. Bazarewsky
Principal Consultant, Windows Server and Security

Using UDP-SIP with Exchange UM and Lync 2010

Attachment: https://bennettadelson.wordpress.com/2012/06/04/using-udp-sip-with-exchange-um-and-lync-2010/kamailio-cfg/ (remember to change extension)

Attachment: asterisk.tar.gz (remember to change extension)

I am working on and off with a client that is deploying Exchange 2010 Unified Messaging and Lync 2010 in their environment. They want to use Exchange UM with a hosted SIP-based VoIP system from a provider that I will refer to as “PhoneCo” for the sake of discussion. Furthermore, they want their Lync environment to work with the Exchange voicemail, and by the way, think it would be nice if they could experiment with Enterprise Voice functionality. Luckily, PhoneCo offers SIP trunks, and will trunk from the hosted VoIP environment to Exchange UM. So all is good, right?

The Problem Statement

Ha ha, of course I am joking. Because although Microsoft talks SIP, and PhoneCo talks SIP, we hit upon a long-standing issue. Microsoft refuses to support UDP SIP (they have their reasons, I won’t debate the point here) while PhoneCo refuses to support TCP SIP. Thus, we have an impasse.

Solution Overview

The official, standard answer to this is to use a Session Border Controller (SBC), which is essentially a SIP middleman box that can do UDP on one end and TCP on the other. A typical SBC also includes firewalling intelligence to prevent denial-of-service and other such nasty behavior. As a result, they generally start at thousands and quickly get into tens of thousands of dollars. In this customer’s case, the SIP trunk is going to be over a private MPLS connection directly between the hosted PBX and the on-premises Microsoft tools, so the customer didn’t want to pay for a lot of security they didn’t need just to deal with this issue.

The customer found a commercial product named Brekeke SIP Server that appears to be $500 to start. This is nice in that (1) it is commercial and (2) it can run on Windows, although it is Java-based so it’s a little messy and gives you one more thing to deal with patching every day or two.

We wanted to see if there was an open-source way to solve this problem. We found a way, and this post documented what we came up with. I have replicated the scenario in a lab, and have since actually simplified things a bit. I have also corrected something we had done to work around an Asterisk “bug” (in quotes because the bug states it’s not really an Asterisk bug) that came up while we were simulating the PhoneCo setup.

So first, here’s the list of VMs that are in the UC Lab:

Hostname IP Description
dc.uclab.local 172.30.1.10 Domain Controller
exchange.uclab.local 172.30.1.12 Exchange 2010
freepbx.uclab.local 172.30.1.11 PhoneCo stand-in
lync.uclab.local 172.30.1.13 Lync 2010
siprouter.uclab.local 172.30.1.14 SIP middleman
tmg.uclab.local 172.30.1.1 TMG 2010
internalclient.uclab.local 172.30.1.100 Test Lync/SIP client

The PhoneCo stand-in is a FreePBX installation using the FreePBX Linux distribution. I am not going to go into details on installing that into a VM because there are plenty of guides on getting that to work. For the purposes of this post I’m going to pretend Asterisk can’t do TCP SIP because that’s what we are looking at with PhoneCo. This also means ignoring all the online info about getting Asterisk to talk to Lync and Exchange using TCP SIP. (Note: Some of these guides assuming port 5065 for talking to Exchange, which is a partial solution. I’ll get into why that’s wrong later on.)

The SIP middleman – SIP router – is a Linux-based CentOS machine running the Kamailio open-source SIP router package. Kamailio is a mature, solid package that is quite amazing in some of what it can do, but I’m ignoring about 99% of it, I think. We may end up needing some of the NAT support eventually at the client, which I’m not getting into here and don’t need for the lab, but otherwise a lot of functionality is actually not in play here.

Preparing the CentOS Machine

So let’s get to it.

  1. I began with a basic minimal CentOS 6.2 installation. Note that I’ve had repeated issues with the Hyper-V Integration Components on this OS so far, so I didn’t bother with them – for a lab it’s not critical. For production you’d care a lot more – the customer uses VMware so this particular issue did not come up.
  2. Next, I logged in as root via SSH (PuTTY is your friend here) and accepted the key when prompted:
    image
    image
  3. I ran yum updateto get all of the current updates for the OS, and rebooted to get the updated kernel loaded.
  4. Using vi, I created /etc/yum.repos.d/kamailio.repowith:
    [kamailio]
    name=Kamailio
    baseurl=http://download.opensuse.org/repositories/home:/kamailio:/telephony/CentOS_CentOS-6/
    enabled=1
    gpgcheck=0

    This looks like this:

    clip_image001

  5. I confirmed that the new repository was visible with yum repolist:clip_image002
  6. I then confirmed that there was a package I could install in that repository with yum list kamailio:
    clip_image003
  7. After confirming the package, I installed it with yum install kamailio:
    clip_image004

    clip_image005
  8. So now I need to configure the beast. Kamailio comes with a very long sample configuration file. Most of it is noise for my use. I tried to trim it down as safely as possible, as well as better fit what I wanted. So using the following commands I saved the shipped file:
    cd /etc/kamailio
    mv kamailio.cfg kamailio.cfg.original
    vi /etc/kamailio.cfg

    And then made mine, which I will explain later after finishing the build instructions:

    #!KAMAILIO
    
    # Remote Hosts
    #!subst "/SIP_UDP_HOST/172.30.1.11/"
    #!subst "/EXCHANGE_UM/172.30.1.12/"
    #!subst "/LYNC_MEDIATION/172.30.1.13/"
    
    listen=172.30.1.14:5060
    listen=172.30.1.14:5065
    listen=172.30.1.14:5067
    
    ####### Global Parameters #########
    
    memdbg=5
    memlog=5
    
    debug=2
    
    log_facility=LOG_LOCAL0
    
    fork=yes
    children=4
    
    disable_tcp=no
    
    auto_aliases=no
    
    /* uncomment and configure the following line if you want Kamailio to
       bind on a specific interface/port/proto (default bind on all available) */
    #listen=udp:10.0.0.10:5060
    
    # life time of TCP connection when there is no traffic
    # - a bit higher than registration expires to cope with UA behind NAT
    tcp_connection_lifetime=3605
    
    ####### Modules Section ########
    
    mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/"
    
    loadmodule "kex.so"
    loadmodule "tm.so"
    loadmodule "tmx.so"
    loadmodule "sl.so"
    loadmodule "pv.so"
    loadmodule "maxfwd.so"
    loadmodule "usrloc.so"
    loadmodule "textops.so"
    loadmodule "siputils.so"
    loadmodule "xlog.so"
    loadmodule "sanity.so"
    loadmodule "ctl.so"
    loadmodule "cfg_rpc.so"
    loadmodule "mi_rpc.so"
    
    # ----- tm params -----
    # auto-discard branches from previous serial forking leg
    modparam("tm", "failure_reply_mode", 3)
    # default retransmission timeout: 30sec
    modparam("tm", "fr_timer", 30000)
    # default invite retransmission timeout after 1xx: 120sec
    modparam("tm", "fr_inv_timer", 120000)
    
    server_header="Server: PhoneCo Intransigence Coping Solution (PICS) 2.0";
    
    ####### Routing Logic ########
    route {
            if(is_method("OPTIONS")) {
                    xlog("L_INFO","OPTIONS from $si");
                    sl_send_reply("200", "Yes, Microsoft, I am alive");
                    exit();
            }
    
            xlog("L_INFO", "*** M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci");
    
            # Route Exchange extensions
            if((to_uri=~"sip:5992") || (to_uri=~"sip:5999")) {
                    xlog("L_NOTICE", "EXCHANGE UM call, $proto port $op, $ru, $fU");
                    t_on_reply("1");
    
                    # https://issues.asterisk.org/jira/browse/ASTERISK-16862
                    # http://imaucblog.com/archive/2009/10/03/part-1-how-to-integrate-exchange-2010-or-2007-with-trixbox-2-8/
                    replace("Diversion: <sip:5999@SIP_UDP_HOST>;reason=unconditional","MCB-Stripped-Header: Diversion");
    
                    switch($op) {
                            case 5060:
                                    xlog("L_NOTICE", "Redirecting to TCP 5060");
                                    t_relay_to("tcp:EXCHANGE_UM:5060");
                                    exit();
                                    break;
                            case 5065:
                                    xlog("L_NOTICE", "Redirecting to TCP 5065");
                                    t_relay_to("tcp:EXCHANGE_UM:5065");
                                    exit();
                                    break;
                            case 5067:
                                    xlog("L_NOTICE", "Redirecting to TCP 5067");
                                    t_relay_to("tcp:EXCHANGE_UM:5067");
                                    exit();
                                    break;
                    }
            }
    
            # Route Lync extensions
            if(to_uri=~"sip:5...") {
                    replace("To: <sip:", "To: <sip:+");
                    xlog("L_NOTICE", "LYNC call to $tu");
                    t_relay_to("tcp:LYNC_MEDIATION:5068");
                    exit();
            }
    
            # Route the rest to Asterisk
            xlog("L_NOTICE", "Asterisk call to $tu");
            forward_udp("SIP_UDP_HOST", 5060);
    }
    
    onreply_route[1] {
            xlog("L_NOTICE", "Handling reply from Exchange relay, status $rs");
            switch($rs) {
                    case 302:
                            xlog("L_NOTICE", "Saw 302 Redirect response, checking details...");
                            if(search(";transport=Tcp")) {
                                    xlog("L_NOTICE", "Saw TCP redirection, changing redirection to UDP");
                                    replace(";transport=Tcp", ";transport=Udp");
                            } else {
                                    xlog("L_NOTICE", "302 was not matched (!)");
                            }
                            exit();
                            break;
                    case 100:
                            xlog("L_NOTICE", "Saw 100, leaving alone...");
                            exit();
                            break;
            }
    
    }

     

  9. I stared the daemon (read: service) with /etc/rc.d/init.d/kamailio start and confirmed that it started  with /etc/rc.d/init.d/kamailio status:clip_image001
  10. I confirmed it was listening (netstat –an | grep 506):clip_image002
  11. I then opened up the firewall to allow those ports in (okay, thats a lie – I floundered a bit before remembering I had to do this) by editing /etc/sysconfig/iptables and adding after the --dport 22line:
    		-A INPUT -p tcp -m state --state NEW -m tcp --dport 5060 -j ACCEPT
    		-A INPUT -p tcp -m state --state NEW -m tcp --dport 5065 -j ACCEPT
    		-A INPUT -p tcp -m state --state NEW -m tcp --dport 5067 -j ACCEPT
    		-A INPUT -p udp -m state --state NEW -m udp --dport 5060 -j ACCEPT
    		-A INPUT -p udp -m state --state NEW -m udp --dport 5065 -j ACCEPT
    		-A INPUT -p udp -m state --state NEW -m udp --dport 5067 -j ACCEPT

    This looks like this when it’s done:
    image

  12. I then made this kick in by restarting the firewall with /etc/rc.d/init.d/iptables restart.
  13. I next added system logger support for the configured log source by editing /etc/rsyslog.confand adding:
    local0.*                                                 /var/log/kamailio.log

    image

  14. I then made this kick in by reloading the logger configuration with /etc/rc.d/init.d/syslog reload.
    image
  15. I don’t want this log to grow uncontrollably so I configured the logrotate daemon to make a new log every day and save seven of them by creating /etc/logrotate.d/kamailiowith:
    /var/log/kamailio.log {
    	rotate 7
    	missingok
    	daily
    }

    image

Preparing Exchange 2010 and Lync 2010

This is normal Exchange and Lync SIP configuration so I’m not going to get into great detail here. The following are the key points:

  • Make sure Lync has a TCP listener on port 5068 on the mediation server of your choice. There’s no high availability here so pick one and go. As quick hints of where this is done in Topology Builder:
    clip_image001[7]
    clip_image002[8]
    After publishing and running Bootstrapper (Lync Setup) on the Mediation Server as instructed by Topology Builder I ran into (what I consider to be) a bug in Lync shown via the event log – there were LS Mediation Server messages 25075 and 25031 indicating no TCP port is enabled, then that the TCP port was requested but ignored. Restarting the Mediation Server service sorted it out. The Kamailio log will show this working (e. g. tail /var/log/kamailio.log):
    image
  • For Exchange, make sure you have TCP enabled on the UM server (requires a service restart to kick in) and that you have an appropriate IP gateway and unsecured telephone extension dial plan configured against that gateway:
    clip_image001[9]
    clip_image002[10]

And that’s it!

So What Does the Configuration Mean?

OK, so what the heck does the configuration I gave you above mean?  Let’s go through it:

#!KAMAILIO

This is a signature for the configuration file.

# Remote Hosts

#!subst "/SIP_UDP_HOST/172.30.1.11/"
#!subst "/EXCHANGE_UM/172.30.1.12/"
#!subst "/LYNC_MEDIATION/172.30.1.13/" 

listen=172.30.1.14:5060
listen=172.30.1.14:5065
listen=172.30.1.14:5067

This is the super important customization part. The three subst lines replace all references to those text strings with the appropriate IP addresses, while the listen lines allow the router to accept traffic on its IP on three ports – 5060, 5065, and 5067. The latter two are because Exchange – for reasons known to Microsoft but not me – takes UM connections on port 5060 but then redirects them to 5065 or 5067. Remember how above I said that some sites use 5065 and that’s wrong?  That’s because they are assuming all redirects are to 5065, but Exchange might want 5067.

Anyway, the next lines are some configuration stuff that is from the default that I left alone mainly because either the settings were fine (e. g. the syslog facility used) or because I didn’t know the implications in changing them (e. g. the children process count); there’s also the enabling of TCP (normally disabled):

####### Global Parameters ######### 
memdbg=5

memlog=5 
debug=2
log_facility=LOG_LOCAL0 
fork=yes

children=4 
disable_tcp=no 
auto_aliases=no 

# life time of TCP connection when there is no traffic
# - a bit higher than registration expires to cope with UA behind NAT
tcp_connection_lifetime=3605

Next are the modules that I am loading. I know I need some of these for sure – there are others I don’t know about so I left well-enough alone and kept them there:

####### Modules Section ######## 
mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" 
loadmodule "kex.so"
loadmodule "tm.so"
loadmodule "tmx.so"
loadmodule "sl.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "textops.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
loadmodule "ctl.so"
loadmodule "cfg_rpc.so"
loadmodule "mi_rpc.so" 

# ----- tm params -----
# auto-discard branches from previous serial forking leg
modparam("tm", "failure_reply_mode", 3)
# default retransmission timeout: 30sec
modparam("tm", "fr_timer", 30000)
# default invite retransmission timeout after 1xx: 120sec
modparam("tm", "fr_inv_timer", 120000)

The next line sets a server header seen in the SIP headers. It is a fun way to point out that PhoneCo was annoying me as well as to hide the actual software being used:

server_header="Server: PhoneCo Intransigence Coping Solution (PICS) 2.0"

Now comes the real meat. It starts the routing logic for incoming SIP calls looking for the OPTIONS call that Lync and Exchange make every nanosecond (approximately) to check to see if their SIP peers are alive. Hence the status text – the code is all that really matters:

####### Routing Logic ########

route {
        if(is_method("OPTIONS")) {
                xlog("L_INFO","OPTIONS from $si");
                sl_send_reply("200", "Yes, Microsoft, I am alive");
                exit();
        }

The next line just acts as a debugging log showing what came in:

        xlog("L_INFO", "*** M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci");

The dollar-sign pseudo-variables are documented here, should you care: http://www.kamailio.org/wiki/cookbooks/3.2.x/pseudovariables

Anyway, moving on, we have the Exchange routing. Looking at this now, I probably want the two extensions (one for the auto-attendant and one for subscriber access) to be substituted variables, but that will be 2.1 I guess:

# Route Exchange extensions
        if((to_uri=~"sip:5992") || (to_uri=~"sip:5999")) {
                xlog("L_NOTICE", "EXCHANGE UM call, $proto port $op, $ru, $fU");
                t_on_reply("1");

This basically says “if a SIP call is made to extension 5992 or extension 5999, then do this…” and starts by indicating that we are going to do a transactional SIP redirect that, when we see a reply, should go to reply handler “1“, which will come later. After that, we have:

        # https://issues.asterisk.org/jira/browse/ASTERISK-16862
        # http://imaucblog.com/archive/2009/10/03/part-1-how-to-integrate-exchange-2010-or-2007-with-trixbox-2-8/
        replace("Diversion: <sip:5999@SIP_UDP_HOST>;reason=unconditional","MCB-Stripped-Header: Diversion");

Why is this here? Basically, Asterisk does something we don’t want it to do on the Exchange redirect – adds an extra SIP Diversion header – and we want that extra header to go away. I need to replace it with something though, so I just made up a vendor header and used that. This is safe as SIP agents – like HTTP server and clients – ignore headers that they don’t know. Next, we take the UDP session and do a transactional redirect to TCP:

        switch($op) {
                case 5060:
                        xlog("L_NOTICE", "Redirecting to TCP 5060");
                        t_relay_to("tcp:EXCHANGE_UM:5060");
                        exit();
                        break;
                case 5065:
                        xlog("L_NOTICE", "Redirecting to TCP 5065");
                        t_relay_to("tcp:EXCHANGE_UM:5065");
                        exit();
                        break;
                case 5067:
                        xlog("L_NOTICE", "Redirecting to TCP 5067");
                        t_relay_to("tcp:EXCHANGE_UM:5067");
                        exit();
                        break;
                }
        }

I couldn’t come up with a “smart” way to do this better; this is a little wordy but it is clear what is happening. I next route the Lync calls (adding the E.164 “+” sign along the way) based on extension pattern (all other 5xxx extensions besides the two special case ones above), with all others going to the Asterisk side:

        # Route Lync extensions
        if(to_uri=~"sip:5...") {
                replace("To: <sip:", "To: <sip:+");
                xlog("L_NOTICE", "LYNC call to $tu");
                t_relay_to("tcp:LYNC_MEDIATION:5068");
                exit();
        }

        # Route the rest to Asterisk
        xlog("L_NOTICE", "Asterisk call to $tu");
        forward_udp("SIP_UDP_HOST", 5060);
}

Notice that I do forward_udpinstead of t_relay_to because I don’t care about maintaining transactional state in the case of going back to Asterisk, so there’s no reason to waste resources on it. I just tell Kamailio to throw it over the wall and forget about it.

Finally, I handle the reply from Exchange. This is why I made the Exchange piece transactional:

onreply_route[1] {
        xlog("L_NOTICE", "Handling reply from Exchange relay, status $rs");
        switch($rs) {
                case 302:
                        xlog("L_NOTICE", "Saw 302 Redirect response, checking details...");
                        if(search(";transport=Tcp")) {
                                xlog("L_NOTICE", "Saw TCP redirection, changing redirection to UDP");
                                replace(";transport=Tcp", ";transport=Udp");
                        } else {
                                xlog("L_NOTICE", "302 was not matched (!)");
                        }
                        exit();
                        break;
                case 100:
                        xlog("L_NOTICE", "Saw 100, leaving alone...");
                        exit();
                        break;
        }

Notice if I get a redirect from Exchange (which I will for port 5060) I change that from a Tcp redirect to a Udp redirect, then send it on its way.

So, this is what is in the lab right now. I think this works – until PhoneCo gets the line in place we won’t know 100% but I think this is close if it isn’t completely right. We’ll see.

Hope this helps you in your integration scenarios!

— Michael C. Bazarewsky
Principal Consultant, Windows Server and Security