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Lync Server integration with Asterisk

    Question

  • Has anyone had success in setting up communication between the two, or found any reference materials to do so? I realize Lync is still new territory, but hoping there has been some success.

    Thanks in advance for any insight!

    Tuesday, October 05, 2010 4:44 PM

Answers

  • Why do so many people choose Trixbox?  I tried it, found it confusing with all the stuff it takes control of.  Anyway, I'll voice my usual disclaimer - I know Asterisk not TrixBox - but they are close enough that we can normally work something out.

    OK - sounds like Trixbox is setup OK - your placing calls from Trixbox to Lync.

    There are settings in the Lync control panel that specifies the port number for the mediation server and the port for the gateway you are sending to.  I do not exactly where it is without seeing my Lync server and I'm not near it today.  I can reference it later.

    Secondly, in asterisk, in the sip.conf file, you can double check what port Asterisk is using AND what port it is using to talk to the Mediation server.  Sounds like they are set correctly already, but it's one more thing to check.

    Regards

    Paul Adams

    PS - I'm starting to think I need to write a step by step Asterisk to Lync guide (and I mean step by step)...  It's just time right?  :-)


    pauladamsit.com
    • Marked as answer by yeahbuddyia Monday, October 11, 2010 3:45 PM
    Friday, October 08, 2010 4:48 PM

All replies

  • Couple of things to keep in mind. Set your sip trunk on asterisk to TCP (I think this requires a patch on Asterisk to support) and set encryption on the Lync client to supported with powershell if you are going to try media bypass with it.

     

    Cheers

    Chris


    http://voipnorm.blogspot.com/
    Tuesday, October 05, 2010 5:16 PM
  • I have this setup in-place - not direct media.  I'd get round to testing that someday.

    Use Asterisk 1.6 or higher, supports TCP 'out the box'.  Set the details of the trunk in the sip.conf file to include the line transport = tcp

    Geoff's blog covers most of it - http://blogs.technet.com/b/gclark/archive/2008/10/09/asterisk-1-6-with-office-communications-server-2007.aspx

    It's the same settings for Lync as OCS in Asterisk.  In addtion to that - you'll need to setup the call flows in & out of Lync, but once you actually get it installed and use the control panel, it's easier to work with then OCS when setting up how to handle calls.

    If you have anything you get stuck on, just ask.

    Regards

    Paul Adams
    pauladamsit.com

    Tuesday, October 05, 2010 5:35 PM
  • Thank you both. I got everything installed as a standard edition front end with collocated mediation server and an edge. You're right Paul - night and day improvement for voice routing administration. I will see what I can try to figure out, but the one piece I'm not immediately finding are the equivalent of the logging / tracing tools from 07 R2. Big help in validating the routing in trying to get things working. Have either of you found them that you could point me in the right direction?

    Thanks again.

    Jeremy

    Tuesday, October 05, 2010 7:46 PM
  • Taking you up on your offer, Paul. :)

     

    To be more specific, I'm using Trixbox 2.8.0.4 in reference to our Asterisk box. I have calls making it to Lync from Trixbox successfully with the SIP trunk between the two of them and sending calls over it based on an Outbound route. When I try to send calls to it from Lync, I get the following: 

     

    Noteworthy piece in setting up the call is that it references the IP of the trixbox correct, but using port 5070:

    TL_INFO(TF_COMPONENT) [0]1C84.1DB4::10/07/2010-21:35:54.554.0000446e (MediationServer,MediationCall.Terminate:mediationcall.cs(271))(0000000003B7CF57)$$START-MEDIATIONSERVER

    MediationCall: 9cb1c6ace18849509a353ab8d02e1db3

    CallId: 50801b60d3b5465cb64bf733750d8d38

    From: sip:jward@involta.com

    To: sip:92132014@10.128.12.5:5070;user=phone;maddr=invmaocsvm01.involtadc.local

    Direction: Outbound

    Start-Line: Mediation Call Terminate successfully.

    $$END-MEDIATIONSERVER

     

    The final failure message:

    TL_INFO(TF_PROTOCOL) [0]0A54.0FD4::10/07/2010-21:35:54.558.000044a7 (SIPStack,SIPAdminLog::TraceProtocolRecord:SIPAdminLog.cpp(125))$$begin_record

    Trace-Correlation-Id: 1184543054

    Instance-Id: 0000F61B

    Direction: outgoing

    Peer: 10.128.12.105:51858

    Message-Type: response

    Start-Line: SIP/2.0 504 Cannot connect to gateway. Socket error: ConnectionRefused

    From: "Jeremy Ward"<sip:jward@involta.com>;tag=b132ae220b;epid=8371fd6545

    To: <sip:2132014;phone-context=defaultprofile@involta.com;user=phone>;tag=6dbec9d898;epid=8BF6B67ACE

    CSeq: 1 INVITE

    Call-ID: 50801b60d3b5465cb64bf733750d8d38

    Authentication-Info: TLS-DSK qop="auth", opaque="080A8014", srand="BC46040C", snum="76", rspauth="4e3fa5d82b71107d66c3e2d37a1c1141229b94ef", targetname="invmaocsvm01.involtadc.local", realm="SIP Communications Service", version=4

    Via: SIP/2.0/TLS 10.128.12.105:51858;ms-received-port=51858;ms-received-cid=16E000

    CONTENT-LENGTH: 0

    P-ASSERTED-IDENTITY: <sip:92132014;phone-context=DefaultProfile@involta.com;user=phone>

    SERVER: RTCC/4.0.0.0 MediationServer

    ms-diagnostics: 10504;source="invmaocsvm01.involtadc.local";reason="Gateway responded with 504 Server Time out";component="MediationServer";SipResponseCode="504";SipResponseText="Cannot connect to gateway. Socket error: ConnectionRefused";GatewayFqdn="10.128.12.5"

    ms-diagnostics-public: 10504;reason="Gateway responded with 504 Server Time out";component="MediationServer";SipResponseCode="504";SipResponseText="Cannot connect to gateway. Socket error: ConnectionRefused"

    ms-trunking-peer: 10.128.12.5

    ms-trunking-peer-state: down

    ms-endpoint-location-data: NetworkScope;ms-media-location-type=intranet

    Message-Body: –

    $$end_record

     

    Any suggestions?

    Thursday, October 07, 2010 9:54 PM
  • Why do so many people choose Trixbox?  I tried it, found it confusing with all the stuff it takes control of.  Anyway, I'll voice my usual disclaimer - I know Asterisk not TrixBox - but they are close enough that we can normally work something out.

    OK - sounds like Trixbox is setup OK - your placing calls from Trixbox to Lync.

    There are settings in the Lync control panel that specifies the port number for the mediation server and the port for the gateway you are sending to.  I do not exactly where it is without seeing my Lync server and I'm not near it today.  I can reference it later.

    Secondly, in asterisk, in the sip.conf file, you can double check what port Asterisk is using AND what port it is using to talk to the Mediation server.  Sounds like they are set correctly already, but it's one more thing to check.

    Regards

    Paul Adams

    PS - I'm starting to think I need to write a step by step Asterisk to Lync guide (and I mean step by step)...  It's just time right?  :-)


    pauladamsit.com
    • Marked as answer by yeahbuddyia Monday, October 11, 2010 3:45 PM
    Friday, October 08, 2010 4:48 PM
  • Thank you Paul! Your response got me thinking in the right direction again. I went back and during the initial deployment, it was setup that the mediation would use "All IP addresses" for the roles vs. defining what to use. As soon as I designated one NIC to the PSTN connection, I was back talking to the PBX. At that point I had to figure out how to route calls in the PBX to either an internal extension or out our PRI. I hadn't done a lot in the extension_custom.conf previously, but I did find a combination of 2 existing strings that I can now reach an internal extension, or if it doesn't match that condition, passes it to the trunk for the PRI and dials successfully. I'm happy to share any details if you or anyone else is curious.

     

    Thanks again all!

    Jeremy

    Monday, October 11, 2010 3:49 PM
  • could you share the details of the extension_custom.conf ?

     

    Thanks

    Wednesday, November 03, 2010 10:39 AM
  • Hello, could you share the settings for the trunk in Asterisk ?
    Tuesday, June 07, 2011 8:24 PM
  • Sorry - must have missed that requst for the extension_custom.conf.  I use Asterisk, which is just extensions.conf, I believe extension_custom.conf is trixbox.

    3rd posting in this thread has a link for Geoff Clark's blog on link Asterisk to OCS / Lync.  Is contains a sip.conf and an extensions.conf

    All the details you need are in those 2 listings. In the sip,.conf file is a section for the truck to OCS - copy those settings and change to use your IP numbers.

    Regards

    Paul Adams


    pauladamsit.com
    Tuesday, June 07, 2011 8:41 PM
  • Here you go for the trunk:

     

    host=10.128.12.9
    
    transport=tcp
    
    port=5060
    
    insecure=very
    
    type=peer
    
    fromdomain=involtadc.local
    
    promiscredir=yes
    
    qualify=no
    
    context=from_lync
    
    canreinvite=yes
    
    dtmfmode=RFC2833
    
    disallow=all
    
    allow=alaw&ulaw

     

    Tuesday, June 07, 2011 8:41 PM
  • OCS / Lync does uLaw only, so allow=ulaw

    And if you have trouble with one way audio, try adding the line
    NAT=yes

    Regards

    Paul Adams


    pauladamsit.com
    Tuesday, June 07, 2011 8:50 PM
  • Thank you all for your great contributions.

    Could you please share your step-by-step configuration settings for

    LYNC 2010 integration with ASTERISK/TRIXBOX.

    My email address is iyewo1@hotmail.com


    Bawo.

     


    Saturday, June 25, 2011 2:53 PM
  • Hi Paul,

    Thanks for your excellent effort. Have you written your step by step Asterisk to Lync guide? If yes can you please share the link to it?

    Thank You


    Favad Q

    Monday, March 05, 2012 4:52 AM
  • Yes, I worked on it with Geoff Clark of Microsoft and it became a chapter of the Lync Resource Kit, released June 2011.

    http://blogs.technet.com/b/drrez/archive/2011/06/07/microsoft-lync-server-2010-resource-kit-interoperability-with-third-party-systems.aspx

    Or try Adam's blog on Lync, Asterisk, Skype.

    http://imaucblog.com/archive/2010/10/09/step-by-step-microsoft-lync-2010-asterisk-and-skype-installationintegration-guide/

    regards

    Paul Adams


    MS Lync Server 2010 Resource Kit writer | website: www.pauladamsit.com

    • Proposed as answer by Agile-Favad Monday, March 05, 2012 5:33 AM
    Monday, March 05, 2012 5:30 AM
  • Hi Paul,

    Thank You for your prompt reply. I'll have a look at it. Also I'm already on the second link in which Adam is using a Command Line interface of Asterisk, is there a guide using the Asterisk GUI or FreePBX GUI?

    Thank You Again :-)


    Favad Q

    Monday, March 05, 2012 5:38 AM
  • There maybe, but not that I'm aware of.

    Asterisk GUI, FreePBX, TrixBox, PIAF - they are all based on Asterisk and are all somewhat similar.

    Google is your friend

    http://lmgtfy.com/?q=Asterisk+GUI+FreePBX+lync

    Regards

    Paul Adams


    MS Lync Server 2010 Resource Kit writer | website: www.pauladamsit.com

    Monday, March 05, 2012 4:25 PM
  • Hi Paul,

    Thank You for your reply. Its indeed an honour to see the author of Lync Resource Kit is active on the forum.

    I'm not sure if this is the right place to ask questions? I can create a new post if you like. I read it and set it up they way it explained. Early days probably but unfortunately as of now I've been unable to establish a successful call from Lync to Client or vice-versa.

    Also, my final objective is to establish Lync inbound and outbound connection with PSTN, through asterisk and a Sip Trunk Provider (which doesn't work directly with Lync). Although the guide mentions this scenario under Asterisk Direct SIP Internals I cannot really find the configuration for that.

    Hope you can guid me please.

    Thank You


    Favad Q

    Monday, March 05, 2012 5:18 PM
  • Thank you for saying so but I don't think it's an honour.  I'm just a one of about 25+ people who contributed work to the book.  There are a number of people who contribute to these forums who all add value.

    Your objective is totally reasonable.  I recently helped an MS Gold Partner who did just that; they used Asterisk-GUI as the gateway for Lync.

    In my limited use of the "Asterisk GUI" interfaces, I have found PIAF (google PIAF) to be good.  It uses the FreePBX interface along with other 'tricks'.

    Feel free to ask any questions you like on the forum, but if you are looking for me to guide you, I am happy to work for you directly.  My rates are reasonable.  If this interest you, please contact me at paul [at] pauladamsit [dot] com and we can discuss it further.

    Regards

    Paul Adams


    MS Lync Server 2010 Resource Kit writer | website: www.pauladamsit.com

    Monday, March 05, 2012 7:16 PM
  • I'm having a similar issue to the on originally in this thread.  Lync 2010 install with Asterisk (FreePBX) as my PSTN gateway.  connection inbound to Lync work perfectly.  All roles but Edge are collocated.  Lync server and PBX are on different VLAN's.  communication between both systems seems great but calls never come outbound from the lync environment.  

    What I view as the important aspect of the logs:

    FQDN/State:pbx-cluster.mydomain.com-Down
    TL_INFO(TF_COMPONENT) [0]0BE4.0888::03/19/2012-21:27:58.662.00026f67 (OutboundRouting,OutboundTarget.IsAvailable:outboundtarget.cs(212))(00000000026D8851)Gateway was Down but has waited enough. Set state back to Up
    TL_VERBOSE(TF_COMPONENT) [0]0BE4.0888::03/19/2012-21:27:58.662.00026f68 (OutboundRouting,OutboundGateway.GetTranslatedNumber:gateway.cs(218))(00000000026D8851)No matches found. (GW=pbx-cluster.mydomain.com, number=5117)
    TL_VERBOSE(TF_COMPONENT) [0]0BE4.0888::03/19/2012-21:27:58.662.00026f69 (OutboundRouting,OutboundRoutingDispatcher.ReplaceCallerIdInRequest:outboundroutingdispatcher.cs(1355))[2576752566]Nothing to replace.
    TL_VERBOSE(TF_COMPONENT) [0]0BE4.0888::03/19/2012-21:27:58.662.00026f6a (OutboundRouting,OutboundRoutingDispatcher.PrepareRequest:outboundroutingdispatcher.cs(1511))[2576752566]New request line: sip:5117@pbx-cluster.mydomain.com:5070;user=phone;maddr=lync-01.mydomain.com. Exit.

    Any help is greatly appreciated as I've been struggling with this for awhile now.  Thanks

    Monday, March 19, 2012 9:47 PM
  • Hmmm - not easy to give an exact answer - could be any number of things and I assume it's FreePBX related, (it might not be).

    As an experienced guess, I'd say FreePBX cannot process the + being sent on the dial string from Lync.  Add a rule to your PSTN gateway in Lync to remove the + and then send the number out to the gateway.

    Regards

    Paul Adams


    MS Lync Server 2010 Resource Kit writer | website: www.pauladamsit.com

    Tuesday, March 20, 2012 6:10 PM
  • Hi there.

    I also have problems with Asterisk. I check my sip.conf and extensions.conf like previous links in other posts. I try to find any solution in google but i cannot solve my issue. When I call to Asterisk extension everything rules, but from Asterisk to Lync I can take call but it hang up in one second. In event viewer can see this as the last message:

    0  (null)
    ms-diagnostics-public:  22;reason="Call failed to establish due to a media connectivity failure when both endpoints are internal";component="MediationServer";Exception="Call failed to establish due to a media connectivity failure when both endpoints are internal";OriginalPresenceState="0";CurrentPresenceState="0";MeInsideUser="Yes";ConversationInitiatedBy="0";SourceNetwork="0";RemotePartyCanDoIM="No"
    
    I posted this answer in Planning threads cause I though it could be about multiple domains issue. I have only a single external SIP domain so I had to replicate my external DNS records in my internal DNS server. Everything works fine, sharing, IM, videoconference but with Asterisk I'm breaking my head with a wall.

    This is the original post: http://social.technet.microsoft.com/Forums/en-US/ocsplanningdeployment/thread/03377a78-c5e2-410d-815a-23a807bf984f

    Could you help me please.

    Regards.

    Wednesday, March 21, 2012 9:10 AM
  • Qoute: "When I call to Asterisk extension everything rules" - I assume you mean calls to Asterisk from Lync are OK.

    Quote: "But from Asterisk to Lync I can take call but it hang up in one second"

    Calls from Asterisk to Lync setup OK, (this is Asterisk and Lync working together to establish a call using SIP), but when it is answered, (moves to the media part = RTP), it hangs up.  OK - so it sounds like you have a media issue (RTP) from Asterisk to Lync.

    If the call is being routed correctly to Lync, we can ignore the extensions.conf

    I assume you've checked this, but in your Asterisk sip.conf, check for the lines directmedia=no and nat=yes in the definition section for your Lync server.  If you have to change anything, restart Asterisk.

    Still not working?  Post your entire sip.conf file.

    Regards

    Paul Adams


    MS Lync Server 2010 Resource Kit writer | website: www.pauladamsit.com


    • Edited by Paul Adams Wednesday, March 21, 2012 2:50 PM changed canreinvite to directmedia
    Wednesday, March 21, 2012 2:47 PM
  • Thanks for answer.

    Sorry... there is a mistake... Asterisk to Lync -> OK, Lync to Asterisk -> FAIL

    Thank you so much for help.

    Sip.conf  (I used all parameters that I found in google. Started with 4 or 5 and try with each of them as you can see in the text above)

    [general]
    context=default
    allowoverlap=no
    udpbindaddr=0.0.0.0
    bindport=5060
    bindaddr=0.0.0.0
    tcpenable=yes
    tcpbindaddr=0.0.0.0
    srvlookup=yes
    notifyhold=yes
    dtmfmode=rfc2833
    
    [1001]
    type=friend
    callerid=1001
    secret=1001
    dtmfmode=rfc2833
    disallow=all
    allow=ulaw,alaw
    transport=udp
    canreinvite=no
    host=dynamic
    mailbox=1001
    nat=yes
    
    [Lync_Trunk]
    type=friend
    port=5068
    host=10.1.1.217
    dtmfmode=rfc2833
    context=from-lync
    qualify=yes
    transport=tcp,udp
    fromdomain=infostock.com.es
    canreinvite=no
    disallow=all
    allow=ulaw,alaw
    nat=yes
    directmedia=no
    

    and extensions.conf

    [general]
    static=yes
    writeprotect=no
    
    [globals]
    
    [default]
    
    ;dialling other extensions starting with 1 followed by three digits
    exten=>_1XXX,1,Dial(SIP/${EXTEN},20)
    exten=>_1XXX,n,hangup()
    
    ;dialling other extensions starting with 2 followed by three digits
    exten=>_5XXX,1,Dial(SIP/Lync_Trunk/${EXTEN},20)
    exten=>_5XXX,n,hangup()
    
    [from-lync]
    ;dialling other extensions starting with 1 followed by three digits
    exten=>_1XXX,1,Dial(SIP/${EXTEN},20)
    exten=>_1XXX,n,hangup()
    

    Wednesday, March 21, 2012 4:50 PM
  • OK - so the problem is the other way around: Lync -> Asterisk

    Put a semi-colon (;) in front of the fromdomain=infostock.com.es line and restart Asterisk.

    Try that.

    Still not working? Try opening an Asterisk console and watching what happens when you place the call.

    For an Asterisk console, type

    asterisk -vvvvvvr

    from a linux command line. You should now be at an Asterisk console with Verbose set to 6.

    What happens when you place the call from Lync.  Look out for WHAT dial string is being dialled and then look as Asterisk works through your dialplan (extensions.conf).  It will work through each step in turn.

    Regards

    Paul Adams


    MS Lync Server 2010 Resource Kit writer | website: www.pauladamsit.com

    Thursday, March 22, 2012 2:56 PM
  • I think is a NAT issue. All seems to be fine but I just found when I try to call to Lync voicemail same occurs. In this case, only Lync infrastructure plays so it must be a misswrong Lync configuration o NAT issue.

    This is asterisk console when i call from Lync to Asterisk.

     == Using SIP RTP CoS mark 5
        -- Executing [1001@from-lync:1] Dial("SIP/Lync_Trunk-0000000e", "SIP/1001,20") in new stack
      == Using SIP RTP CoS mark 5
        -- Called 1001
        -- SIP/1001-0000000f is ringing
        -- SIP/1001-0000000f answered SIP/Lync_Trunk-0000000e
        -- Packet2Packet bridging SIP/Lync_Trunk-0000000e and SIP/1001-0000000f
      == Spawn extension (from-lync, 1001, 1) exited non-zero on 'SIP/Lync_Trunk-0000000e'

    Thanks for your time Paul.

    Regards.

    Friday, March 23, 2012 8:54 AM
  • Another piece of the puzzle - "when I try to call to Lync voicemail same occurs."  I agree that is seems to be related to Lync sending RTP.  This is a new issue for me - not seen this before.

    You could try this which seems to be the same issue.  The fix semed to be apply all patches to Lync - and then restart.

    http://social.technet.microsoft.com/Forums/sv-SE/ocsvoice/thread/683b6263-de4b-4188-a222-ec33bffb92f8

    Are you trying to do some sort of media bypass?

    Can you post more of the Lync log from a failed call?

    Regards

    Paul Adams


    MS Lync Server 2010 Resource Kit writer | website: www.pauladamsit.com


    • Edited by Paul Adams Friday, March 23, 2012 3:44 PM Ask for logs...
    Friday, March 23, 2012 3:13 PM
  • Thank you so much Paul. I finally solve this issue activating NAT in its firewall rule, between PSTN GW (FreePBX based) and Lync FE. Now i can call directly to user IP phones. Lync -> FreePBX.
    From IP phones to Lync I have pass through an Asterisk, cause FreePBX based system is pretty closed and it isn't be able to send traffic over TCP, just UDP. I solve this with 2 trunks more: one between FreePBX and Asterisk and the other one between Asterisk and Lync. By this way, I have 3 SIP trunks, Asterisk-PBX, Asterisk-Lync and Lync-PBX :)

    Voicemail runs now too.

    Cheers.

    Wednesday, March 28, 2012 3:47 PM
  • Awesome!  :-)

    Arhh - I can see what's going on here.

    Not trying to cause you any further work, just offering my 2 cents.

    Freepbx is Asterisk, just a different front end.

    I run PIAF (PBX In A Flash) on a virtual sertup with Lync.  That is Asterisk based also and uses FreePBX as the front end.

    You can add tcp to freepbx in the same as Asterisk, by editing the sip.conf file.  If you've got Asterisk working using TCP then you must have done this already.

     

    Technically, you are running one more server then you need to, but hey, it's working!  :-)

     

    Regards

    Paul Adams

     


    MS Lync Server 2010 Resource Kit writer | website: www.pauladamsit.com

    Wednesday, March 28, 2012 5:36 PM
  • Hi All,

    Hope you are all well. I've set up a Lync 2010 environment in a vmware lab following various blogs and posts. My network is a back to back TMG perimiter network and I setup FreePBX as a PSTN Gatewat for the Lync 2010 Consolidated server in the perimeter. Again looking through this and other posts managed to setup calls from both FreePBX to Lync users and Lync to FreePBX users.

    At the moment I can't seem to make any calls out from the FreePBX, doesnt matter if I try a Lync user or a FreePBX user. I trired asterisk -vvvvvvr from the commad line as Paul suggested earlier in this post. and I got this

    ---------------------------

    Executing [8+442077003233@from-internal :6] Macro ("SIP/3001-0000004d", "outisbusy,") in new stack

    Executing [s@macro-outisbusy:1] Progress ("SIP/3001-0000004d", "") in new stack 

    Executing [s@macro-outisbusy:2] GotoIf ("SIP/3001-0000004d", "0?emergency,1") in new stack

    Executing [s@macro-outisbusy:3] GotoIf ("SIP/3001-0000004d", "0?intracompany,1") in new stack

    Executing [s@macro-outisbusy:4] Playback ("SIP/3001-0000004d", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack

    <SIP/3001-0000004d. Playing 'all-circuits-busy-now.ulaw'

    <SIP/3001-0000004d. Playing 'pls-try-call-later.ulaw'

    Executing [s@macro-outisbusy:5] Congestion ("SIP/3001-0000004d", "20") in new stack

    [2012-05-21 20:32:07] WARNING[-1] channel .c:4742 ast_prod: Prodding Channel 'SIP/3001-0000004d' failed

    Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/3001-0000004d' in macro 'outisbusy'

    Spawn extension (from-internal, 8+442077003233, 6) exited non-zero on 'SIP/3001-0000004d'

    Executing [h@from-internal:1] Hangup ("SIP/3001-0000004d", "") in new stack

    Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/3001-0000004d'

    ----------------------

    Anyone able to help out please or point me in the right direction? Much appriciated.

    Kind regards

    Hammad

    Monday, May 21, 2012 7:54 PM
  •  

    Hi Hammad,

    Firstly - if that UK Inner London phone number listed in your logs is personal to you (like your cell or home number)- I'd advise you not to list it in a public forum on the Internet.

     

    Secondly - sounds like you have an outbound FreePBX issue, not actually related to Lync.  Specifically, it seems you do not have any outbound routes specified.  Note how it checks emergency, then inter-company, then fails - nothing else.

    I'd suggest this page - http://www.freepbx.org/support/documentation/module-documentation/outbound-routes

    NOTE - do not just specify an outbound route, specify the dialling patterns to match to use that outbound route.  Freepbx makes it easy with some drop down choices.  Chose local 7/10 digits, plus international.

    Some of it you might have to edit to fit UK numbers. 

     

    If you are struggling, the guys in the FreePBX forums are very helpful.

    Regards

    Paul Adams


    MS Lync Server 2010 Resource Kit writer | website: www.pauladamsit.com

    Wednesday, May 23, 2012 5:04 AM