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Problem with direct call to Lync client from PSTN with one line. Gateway Audiocodes MP-114 RRS feed

  • Question

  • Hello all!

    I have a problem.

    I have a homelab with gateway Audiocodes MP-114 2 FXS, 2 FXO, Mediation Server with FE colocated, one telephone line which is connected to Port 3 FXO in the Gateway. I have a task that inbound call from PSTN will go directly to Lync client.

    User has tel:801 in the LINE URI: Parameter.

    I have read lots of documentation, setup proxy on the gateway, automatic dialing to 801 on port 3.

    Nothing works.

    Please help!
    Friday, November 30, 2012 7:39 AM

Answers

  • Hi,

    Does the call from Lync to PSTN work properly?

    Have you tried to turn off media bypass or disable refer support to test the issue?

    I suggest trying to enable logging tool to get detailed report:

    http://blog.schertz.name/2011/06/using-the-lync-logging-tool/

    Note: Microsoft is providing this information as a convenience to you. The sites are not controlled by Microsoft. Microsoft cannot make any representations regarding the quality, safety, or suitability of any software or information found there. Please make sure that you completely understand the risk before retrieving any suggestions from the above link.


    Kent Huang
    TechNet Community Support

    • Proposed as answer by Kent-Huang Tuesday, December 4, 2012 2:28 AM
    • Marked as answer by Kent-Huang Wednesday, December 5, 2012 6:14 AM
    Monday, December 3, 2012 5:35 AM
  • have you turned on syslogging on the AudioCodes device itself (especially if the call is never making it to Lync. If the call is making it to Lync then the logging tool mentioned above is a great resource) and seen what comes through when you dial in from outside? Are the certificates set and working between the AudioCodes box and the Lync servers? There are a ton of ways for this to go wrong so maybe start with a little more info about how your topology is setup

    Steve

    • Marked as answer by Kent-Huang Wednesday, December 5, 2012 6:14 AM
    Wednesday, December 5, 2012 12:25 AM

All replies

  • Hi,

    Does the call from Lync to PSTN work properly?

    Have you tried to turn off media bypass or disable refer support to test the issue?

    I suggest trying to enable logging tool to get detailed report:

    http://blog.schertz.name/2011/06/using-the-lync-logging-tool/

    Note: Microsoft is providing this information as a convenience to you. The sites are not controlled by Microsoft. Microsoft cannot make any representations regarding the quality, safety, or suitability of any software or information found there. Please make sure that you completely understand the risk before retrieving any suggestions from the above link.


    Kent Huang
    TechNet Community Support

    • Proposed as answer by Kent-Huang Tuesday, December 4, 2012 2:28 AM
    • Marked as answer by Kent-Huang Wednesday, December 5, 2012 6:14 AM
    Monday, December 3, 2012 5:35 AM
  • have you turned on syslogging on the AudioCodes device itself (especially if the call is never making it to Lync. If the call is making it to Lync then the logging tool mentioned above is a great resource) and seen what comes through when you dial in from outside? Are the certificates set and working between the AudioCodes box and the Lync servers? There are a ton of ways for this to go wrong so maybe start with a little more info about how your topology is setup

    Steve

    • Marked as answer by Kent-Huang Wednesday, December 5, 2012 6:14 AM
    Wednesday, December 5, 2012 12:25 AM
  • I will try and send the results.

    Thanks.

    Thursday, December 6, 2012 3:34 PM
  • I got the log.

    TL_INFO(TF_PROTOCOL) [0]1668.0120::12/09/2012-21:33:39.933.00016c5c (SIPStack,SIPAdminLog::TraceProtocolRecord:SIPAdminLog.cpp(125))$$begin_record


    Trace-Correlation-Id

    : 2995789077


    Instance-Id

    : 000011CB


    Direction

    : incoming


    Peer

    : LYNC01-LAB.HOMELAB.LOCAL:57847


    Message-Type

    : request


    Start-Line

    : INVITE sip:801;phone-context=DefaultProfile@nantonov.ru;user=phone SIP/2.0

    From

    : "Gateway"<sip:801;phone-context=DefaultProfile@nantonov.ru;user=phone>;epid=78AB20D280;tag=b148b939c1


    To

    : <sip:801;phone-context=DefaultProfile@nantonov.ru;user=phone>


    CSeq

    : 66 INVITE


    Call-ID

    : af8ce164-3359-44a3-baef-4f92e538483a


    MAX-FORWARDS

    : 70


    VIA

    : SIP/2.0/TLS 172.30.2.100:57847;branch=z9hG4bK2c54b425


    CONTACT

    : <sip:LYNC01-LAB.HOMELAB.LOCAL@nantonov.ru;gruu;opaque=srvr:MediationServer:DiFqpV_JC1SlciAA1gTEIQAA;grid=48d8945e64e048dfb079d42663dc8f86>;isGateway


    CONTENT-LENGTH

    : 2338


    SUPPORTED

    : replaces


    SUPPORTED

    : ms-safe-transfer


    SUPPORTED

    : ms-bypass


    SUPPORTED

    : ms-dialog-route-set-update


    SUPPORTED

    : timer


    SUPPORTED

    : 100rel


    SUPPORTED

    : gruu-10


    USER-AGENT

    : RTCC/4.0.0.0 MediationServer


    CONTENT-TYPE

    : multipart/alternative; boundary=5H4TmPTzP6p3Oq3MolTwc65eT4qcIidM


    ALLOW

    : ACK


    ms-endpoint-location-data

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


    ms-call-source

    : non-ms-rtc


    ms-trunking-peer

    : 172.30.2.2;User-Agent="Audiocodes-Sip-Gateway-MP-114 FXS_FXO/v.5.60A.014.009"


    Session-Expires

    : 1800


    Min-SE

    : 90


    Allow

    : CANCEL,BYE,INVITE,REFER,NOTIFY,PRACK,UPDATE

    TL_INFO(TF_DIAG) [0]1668.0120::12/09/2012-21:33:39.937.0001763a (SIPStack,SIPAdminLog::TraceDiagRecord:SIPAdminLog.cpp(147))$$begin_record


    LogType

    : diagnostic


    Severity

    : information


    Text

    : Response successfully routed


    SIP-Start-Line

    : SIP/2.0 404 No matching rule has been found in the dial plan for the called number.


    SIP-Call-ID

    : af8ce164-3359-44a3-baef-4f92e538483a


    SIP-CSeq

    : 66 INVITE


    Peer

    : LYNC01-LAB.HOMELAB.LOCAL:57847


    Data

    : destination="LYNC01-LAB.HOMELAB.LOCAL"

    $$end_record

    TL_INFO(TF_PROTOCOL) [1]1668.0120::12/09/2012-21:33:44.947.000191a8 (SIPStack,SIPAdminLog::TraceProtocolRecord:SIPAdminLog.cpp(125))$$begin_record


    Trace-Correlation-Id

    : 2187484564


    Instance-Id

    : 000011D1


    Direction

    : outgoing;source="local"


    Peer

    : LYNC01-LAB.HOMELAB.LOCAL:57850


    Message-Type

    : response


    Start-Line

    : SIP/2.0 404 No matching rule has been found in the dial plan for the called number.


    From

    : "Gateway"<sip:801;phone-context=DefaultProfile@nantonov.ru;user=phone>;epid=78AB20D280;tag=7f7ca1c1c


    To

    : <sip:801;phone-context=DefaultProfile@nantonov.ru;user=phone>;tag=43ACE4D0C0D16CB175409714110A28C0


    CSeq

    : 68 INVITE


    Call-ID

    : 314b5e5f-b108-487a-a2d2-f4e5bc9f737c


    Via

    : SIP/2.0/TLS 172.30.2.100:57850;branch=z9hG4bKc166a6f3;ms-received-port=57850;ms-received-cid=3AB00


    ms-diagnostics

    : 14006;reason="Unable to find a potentially matching regex";source="LYNC01-LAB.HOMELAB.LOCAL";CalledNumber="801";ProfileName="DefaultProfile";appName="TranslationService"


    Server

    : TranslationService/4.0.0.0


    Content-Length

    : 0


    Message-Body

    :


    $$end_record

    Sunday, December 9, 2012 9:48 PM
  • I have created the normalization rule:

    Exact 3 digits = $1

    But nothing changed

    Sunday, December 9, 2012 9:48 PM
  • not sure what 3 digits = $1 does for you but you need to have the 801 extension normalized to a full E.164 number (ex: +14251234801) for it to work in Lync. Is that what you are doing?
    Monday, December 10, 2012 9:48 PM
  • Hello Scarr.

    I changed Line URI to +1434454433;ext=801.

    Created normalization Rule -

    ^(\d{3})$

    +1434454433;ext=$1


    $$end_record

    Nothing works again.

    TL_INFO(TF_PROTOCOL) [0]1668.1B34::12/11/2012-18:39:54.615.001080af (SIPStack,SIPAdminLog::TraceProtocolRecord:SIPAdminLog.cpp(125))$$begin_record


    Trace-Correlation-Id

    : 3229493997


    Instance-Id

    : 00005D28


    Direction

    : outgoing;source="local"


    Peer

    : LYNC01-LAB.HOMELAB.LOCAL:61363


    Message-Type

    : response


    Start-Line

    : SIP/2.0 404 No matching rule has been found in the dial plan for the called number.


    From

    : <sip:801;phone-context=DefaultProfile@nantonov.ru;user=phone>;epid=25D293BF6E;tag=20ab9f9fab


    To

    : <sip:801;phone-context=DefaultProfile@nantonov.ru;user=phone>;tag=43ACE4D0C0D16CB175409714110A28C0


    CSeq

    : 47 SERVICE


    Call-ID

    : a9463db617c645029d17c24481208955


    Via

    : SIP/2.0/TLS 172.30.2.100:61363;branch=z9hG4bKf430a263;ms-received-port=61363;ms-received-cid=1A4800


    ms-diagnostics

    : 14006;reason="Unable to find a potentially matching regex";source="LYNC01-LAB.HOMELAB.LOCAL";CalledNumber="801";ProfileName="DefaultProfile";appName="TranslationService"


    Server

    : TranslationService/4.0.0.0


    Content-Length

    : 0


    Message-Body

    :

    Tuesday, December 11, 2012 5:55 AM
  • hmmm, the other thought I had then is that your from address isn't in E.164 format either and I've seen issues with this when a call comes through an AudioCodes device so if you could build a translation rule on the MP-114 to convert the from number into E.164 too that may do the trick as Lync can't convert that part.
    Tuesday, December 11, 2012 9:52 PM
  • Could you please send me the link how to do that?

    I think its a manipulation rule in Audiocodes Topology?

    Wednesday, December 12, 2012 7:10 AM
  • nevermind, I didn't remember correctly. The source number should be fine and clearly your error logs show an issue with the destination number. One question then is whether your routing is setup on the default profile or do you have a site or user profile you are working with? Other than that I'm not sure why your routing rules aren't picking up that 801 number and connecting it to a user. Have you done the dialing tests in the Control Panel of Lync to see if the global rule set translates 801 into something that can be connected to a user in that it matches how you actually set up the user themselves (whether as a straight +801 or as an extension ex: +14251234444;ext=801).

    Steve

    Wednesday, December 12, 2012 10:39 PM