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    Hey everyone. 

    I'm currently SIP Trunked from Avaya SES.  The trunk is built and i'm able to dial out and all that good jazz.  however, me like everyone else it seems, is unable to get incoming dialing to work.  I have the Avaya team sending me the full 11 digits.  i'm trying to add the + at my end.  They have tried to tell it to send the + as well, however, i don't see it on my traces.  So i asked them to completely remove the + from any of their configurations and I would try to add it.  so my question is, do I add it in the Dial Plan Normalization rules, or do I need to use the trunk configuration tab?  I've tried both and I cannot get it to add the + sign.  So my guess, is it is unable to look me up in AD.  Any ideas??!!  Thanks!

    Tuesday, December 21, 2010 5:02 PM

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    Hi Jason,

    Rather than just write a reply here I have created a blog post on the subject

    http://voipnorm.blogspot.com/2010/12/lync-inbound-normalization-rules.html

    Hope this helps.

     

    Cheers

    Chris

    http://voipnorm.blogspot.com/
    • Proposed as answer by jwdberlin Tuesday, December 21, 2010 8:23 PM
    • Marked as answer by Jason Sloan Wednesday, December 22, 2010 10:04 PM
    Tuesday, December 21, 2010 7:07 PM
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    Hey guys.  i got the answer.  So apparently, at some point one of the Avaya guys rebuilt the trunk, and created a trunk directly off CM 6 and bypassed the Session manager....and in addition, he had Avaya to Lync Signaling setup wrong with the domain names...so now IT WORKS!

    so let me run it down.

    Avaya CM 6 TCP 5070 to Lync TCP 5066 SIP Trunk bypassing SES

    Avaya to Lync Signal Far-end Domain = SIP domain; Far-end Node Name = FQDN of Lync server; Far-end Listen Port = TCP 5066 lync server listening port

    Lync to Avaya Signal Far-end Domain = FQDN of Mediation Server

     

    THANKS GUYS!!!

    • Marked as answer by Jason Sloan Wednesday, December 22, 2010 10:04 PM
    Wednesday, December 22, 2010 5:52 PM

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    Hi Jason,

    Rather than just write a reply here I have created a blog post on the subject

    http://voipnorm.blogspot.com/2010/12/lync-inbound-normalization-rules.html

    Hope this helps.

     

    Cheers

    Chris

    http://voipnorm.blogspot.com/
    • Proposed as answer by jwdberlin Tuesday, December 21, 2010 8:23 PM
    • Marked as answer by Jason Sloan Wednesday, December 22, 2010 10:04 PM
    Tuesday, December 21, 2010 7:07 PM
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    Hi Chris,

    Thanks for the Link, I appreciate it.

    I did do just that, however, i'm having issues still when a call is coming in.  I did a trace and my avaya team is sending 19135437158@lync.domain.com   when i dial 7158 on my avaya phone.   lync.domain.com is the name of my Lync server.  My sip domain is domain.com  is he sending the digits correctly?  or should it be 19135437158@domain.com ?

     

    When i read the traces it is not showing the +19135437158@lync.skccom.com.  it's rather confusing why it will not normalize when it gets to me.

    Tuesday, December 21, 2010 9:55 PM
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    Is Lync returning a error message? Might want to make sure that the outbound SIP trunk from Avaya is hitting port 5060 TCP. Below is a snipit from the Avaya documentation from the SES (unless you have a later version) for OCS. The FQDN of the mediation server should be appearing in the inbound SIP invite.

     

    For SIP Invite messages matched by the address map configured in Steps 2 – 4, the

    Contact

    replaces the matched Request-URI and specifies where the SIP Invite messages are to be routed.

    Configure a

    Contact and click on “Add”. In the example below, the Contact is

    “

    sip:+$(user)@msmedia1.sitlms.net:5060;transport=tcp” (latter characters not visible in the

    screenshot below) and is interpreted as follows:

    •

    The user portion of the outgoing Request-URI contains “+” followed by the user portion, as

    embodied in “$(user)”, of the matched Request-URI. Essentially, the called party number is

    pre-pended with a “+” to comply with E.164 format.

    •

    The host portion of the outgoing Request-URI contains the FQDN of the Microsoft

    Mediation Server.

    •

    The outgoing SIP Invite message is to be sent to port 5060 over a TCP connection.

    http://voipnorm.blogspot.com/
    Wednesday, December 22, 2010 2:13 AM
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    Did you use logging tool with SipStack and S4 (Level All, all flags)? TCP port for Lync mediation server role is 5068 - is INVITE arriving there?

    Johann Deutinger | MCTS Exchange 2007/2010 / OCS 2007 | ucblog.deutinger.de
    Wednesday, December 22, 2010 11:36 AM
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    hey guys....I appreciate everyone helping me with this.  My telco talents are still RAW.

    So, my Mediation is collocated on my front end "lync.skccom.com".  Mediation Port is listening on TCP 5066. 

    My PSTN Gateway is listening on 5070 and is configured to point to my 5066 on my lync.skccom.com.  now, should I put the Avaya SES as the PSTN Gateway or the Avaya CM?  The sip trunk is off the SES but the CM is the call handler.  In the Invite i'm seeing the IP of the CM shown below

    TL_INFO(TF_PROTOCOL) [0]1150.19D4::12/21/2010-22:01:32.172.0001b61b (S4,SipMessage.DataLoggingHelper:sipmessage.cs(686))[459914669]

    <<<<<<<<<<<<Incoming SipMessage c=[<SipTcpConnection_2281665>], 192.168.10.121:5066<-192.168.20.93:32620

    INVITE sip:19135437158@lync.skccom.com SIP/2.0

    FROM

    TO

    CSEQ

    CALL-ID

    MAX-FORWARDS

    VIA

    ROUTE

    RECORD-ROUTE

    CONTACT

    CONTENT-LENGTH

    SUPPORTED

    USER-AGENT

    CONTENT-TYPE

    ACCEPT-LANGUAGE

    ALERT-INFO

    ALLOW

    P-ASSERTED-IDENTITY

    Min-SE

    Session-Expires

    v=0

    o=- 1 1 IN IP4 192.168.20.93

    s=-

    c=IN IP4 192.168.20.95

    b=AS

    t=0 0

    a=avf

    a=csup

    m=audio 15864 RTP/AVP 0 8 101 13

    a=rtpmap

    a=rtpmap

    a=rtpmap

    a=rtpmap

    ------------EndOfIncoming SipMessage

     

    After I get the SIP 100 Trying the next 4 In's then are followed by the number changing to this: SERVICE sip:19135437158;phone-context=dialstring@skccom.com;user=phone  Start-Line: Received new Gateway incoming session with requestUri: sip:19135437158@lync.skccom.com

     

    Start-Line: Mapping GatewayListener to incoming session

    Start-Line: There was an inbound call error, sending a CER

    Start-Line: Valid phone number for inbound call.

    TL_INFO(TF_PROTOCOL) [0]1150.1748::12/21/2010-22:01:32.173.0001b629 (S4,SipMessage.DataLoggingHelper:sipmessage.cs(613))[459914669]

    >>>>>>>>>>>>Outgoing SipMessage c=[<SipTcpConnection_2281665>], 192.168.10.121:5066->192.168.20.93:32620

    SIP/2.0 488 Not Acceptable Here

    FROM

    TO

    CSEQ

    CALL-ID

    VIA

    CONTENT-LENGTH

    SERVER

    ------------EndOfOutgoing SipMessage

     

    : RTCC/4.0.0.0 MediationServer    : 0    : SIP/2.0/TCP 192.168.20.93:5070;branch=z9hG4bK096c0112e14e01a0104d20d0cb00    : 096c0112e14e019f104d20d0cb00    : 1 INVITE    : <sip:19135437158@lync.skccom.com>;epid=B79DF75C94;tag=4e744a50ad    : "Jason Sloan"<sip:+7239@skccom.com>;tag=096c0112e14e019e104d20d0cb00    :13 CN/8000    :101 telephone-event/8000    :8 PCMA/8000    :0 PCMU/8000    :avf-v0    :avc=n prio=n    :64    : 1200;refresher=uac    : 1200    : "Jason Sloan" <sip:+7239@skccom.com>    : INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,INFO,PRACK,PUBLISH    : <cid:internal@lync.skccom.com>;avaya-cm-alert-type=internal    : en    : application/sdp    : Avaya CM/R016x.00.0.345.0    : 100rel,histinfo,join,replaces,sdp-anat,timer    : 249    : "Jason Sloan" <sip:+7239@192.168.20.93:5070;transport=tcp>    : <sip:192.168.20.93:5070;transport=tcp;lr>    : <sip:192.168.10.121:5066;transport=tcp;lr;phase=terminating>    : SIP/2.0/TCP 192.168.20.93:5070;branch=z9hG4bK096c0112e14e01a0104d20d0cb00    : 71    : 096c0112e14e019f104d20d0cb00    : 1 INVITE    : <sip:19135437158@lync.skccom.com>    : "Jason Sloan" <sip:+7239@skccom.com>;tag=096c0112e14e019e104d20d0cb00
    Wednesday, December 22, 2010 2:39 PM
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    Oh yeah, 1 more point.  I can call out and have media streams just fine from Lync to Avaya End point, or out the Trunk to the PSTN.  it's only incoming i'm having issues with.

    Wednesday, December 22, 2010 2:46 PM
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    Have you tried configuring 5068 for TCP in topology? 5066 may be used for E9-1-1:

    http://technet.microsoft.com/en-us/library/gg398833.aspx

     

    Johann Deutinger | MCTS Exchange 2007/2010 / OCS 2007 | ucblog.deutinger.de
    Wednesday, December 22, 2010 3:27 PM
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    I do not have 911 configured.  Wouldn't my media streams fail when i call from Lync to avaya if Avaya wasn't able to communicate back to me on 5066?
    Wednesday, December 22, 2010 3:30 PM
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    Hey guys.  i got the answer.  So apparently, at some point one of the Avaya guys rebuilt the trunk, and created a trunk directly off CM 6 and bypassed the Session manager....and in addition, he had Avaya to Lync Signaling setup wrong with the domain names...so now IT WORKS!

    so let me run it down.

    Avaya CM 6 TCP 5070 to Lync TCP 5066 SIP Trunk bypassing SES

    Avaya to Lync Signal Far-end Domain = SIP domain; Far-end Node Name = FQDN of Lync server; Far-end Listen Port = TCP 5066 lync server listening port

    Lync to Avaya Signal Far-end Domain = FQDN of Mediation Server

     

    THANKS GUYS!!!

    • Marked as answer by Jason Sloan Wednesday, December 22, 2010 10:04 PM
    Wednesday, December 22, 2010 5:52 PM
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    Hi Jason,

    The SES should be a PSTN Gateway and the ports can be the same at both ends to make it simple. I do not think its your phone number format that is the issue. To me it looks like a different issue, it may be the fact you have not got the  trunk pointed at the SES but I would think if you dont have that setup right you would not be able to make outbound calls. 

    http://voipnorm.blogspot.com/
    Wednesday, December 22, 2010 6:04 PM