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Lync 2013 RGS - PSTN RRS feed

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    Hi, everyone!

    I have been searching solution of my problem for half a year and I've not found anything. So the problem is:

    External Call -> PSTN (Asterisk PBX 11.8.1) ->  Lync Front-End -> RGS (ext. 1000) -> RGS Agent (aAstra LPE)

    Asterisk is used for UDP to TCP converting. Sometimes external call dropped and I can't realy understand why ( . On the reporting server i see than Diagnostic ID for dropped call is 51004 - seems everything is OK and this is expected. I turned on CLS on Front-End and turned on debug-log on Asterisk:

    FE:

    xx.xx.xx.11 - FE IP
    xx.xx.xx.50 - Asterisk IP


    xx.xx.xx.11:5068-> xx.xx.xx.50:35162
    BYE sip:<external_number>@xx.xx.xx.50:5060;transport=TCP SIP/2.0
    FROM: <sip:1000@xx.xx.xx.11:5068>;epid=1F1F675A78;tag=1b534dd234
    TO: <sip:<external_number>@xx.xx.xx.50>;tag=as79f5e121
    CSEQ: 1 BYE
    CALL-ID: 0596b18b3a3c372e4aa2d8955de7c87c@xx.xx.xx.50:5060
    MAX-FORWARDS: 70
    VIA: SIP/2.0/TCP xx.xx.xx.11:5068;branch=z9hG4bKea412ed
    CONTACT: <sip:lync_pool_name:5068;transport=Tcp;maddr=xx.xx.xx.11>
    CONTENT-LENGTH: 0
    USER-AGENT: RTCC/5.0.0.0 MediationServer

    I can't understand why Mediation Server sent BYE from RGS ext/number (1000) if call had been transfered to RGS Agent ?

    Please, help.

     

    Friday, October 17, 2014 9:25 AM

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    I think, I've solved my problem - I replaced aAstra LPE on Polycom CX300. Two days without dropped calls.
    • Marked as answer by LyncDummy Thursday, October 23, 2014 5:45 AM
    Thursday, October 23, 2014 5:45 AM

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    Are the mediation server collocated with the front end?

    The callflow is

    External Call -> PSTN (Asterisk PBX 11.8.1) ->  Lync mediation -> lync front end.

    I would suggest to take the logs using clslogging command for incoming and outoging calls and analyse whether any error reported.

    - Muralidharan. Please mark as answer/useful if my contribution helps you.

    Friday, October 17, 2014 10:57 AM
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    Yep, Mediation collocated with Front-End

    Logs contains:

    Selected by CALL-ID

    xx.xx.xx.11 - FE IP
    xx.xx.xx.50 - Asterisk IP

    Time: 13:34:54.484

    <<<<<<<<<<<<Incoming SipMessage c=[<SipTcpConnection_303BCC8>], xx.xx.xx.11:5068<-xx.xx.xx.50:35162
    INVITE sip:1000@xx.xx.xx.11:5068 SIP/2.0
    FROM: "<external_number>" <sip:<external_number>@xx.xx.xx.50 >;tag=as79f5e121
    TO: <sip:1000@xx.xx.xx.11:5068>
    CSEQ: 102 INVITE
    CALL-ID: 0596b18b3a3c372e4aa2d8955de7c87c@xx.xx.xx.50:5060
    MAX-FORWARDS: 70
    VIA: SIP/2.0/TCP xx.xx.xx.50:5060;branch=z9hG4bK0bf92c3c
    CONTACT: <sip:<external_number>@xx.xx.xx.50:5060;transport=TCP>
    CONTENT-LENGTH: 256
    DATE: Wed, 15 Oct 2014 13:34:54 GMT
    SUPPORTED: replaces, timer
    USER-AGENT: Asterisk PBX 11.8.1
    CONTENT-TYPE: application/sdp
    ALLOW: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

    v=0
    o=root 843370910 843370910 IN IP4 xx.xx.xx.50 
    s=Asterisk PBX 11.8.1
    c=IN IP4 xx.xx.xx.50 
    t=0 0
    m=audio 10164 RTP/AVP 8 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:off - - - -
    a=ptime:20
    a=sendrecv

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

    Time:13:34:54.485

    >>>>>>>>>>>>Outgoing SipMessage c=[<SipTcpConnection_303BCC8>], xx.xx.xx.11:5068->xx.xx.xx.50:35162
    SIP/2.0 100 Trying
    FROM: "<external_number>"<sip:<external_number>@xx.xx.xx.50>;tag=as79f5e121
    TO: <sip:1000@xx.xx.xx.11:5068>
    CSEQ: 102 INVITE
    CALL-ID: 0596b18b3a3c372e4aa2d8955de7c87c@xx.xx.xx.50 :5060
    VIA: SIP/2.0/TCP xx.xx.xx.50 :5060;branch=z9hG4bK0bf92c3c
    CONTENT-LENGTH: 0
    ------------

    Time: 13:34:55.004

    >>>>>>>>>>>>Outgoing SipMessage c=[<SipTcpConnection_303BCC8>], xx.xx.xx.11:5068->xx.xx.xx.50:35162
    SIP/2.0 183 Session Progress
    FROM: "<external_number>"<sip:<external_number>@xx.xx.xx.50>;tag=as79f5e121
    TO: <sip:1000@xx.xx.xx.11:5068>;tag=1b534dd234;epid=1F1F675A78
    CSEQ: 102 INVITE
    CALL-ID: 0596b18b3a3c372e4aa2d8955de7c87c@xx.xx.xx.50:5060
    VIA: SIP/2.0/TCP xx.xx.xx.50:5060;branch=z9hG4bK0bf92c3c
    CONTACT: <sip:lync_pool_name:5068;transport=Tcp;maddr=xx.xx.xx.11>
    CONTENT-LENGTH: 0
    ALLOW: CANCEL
    ALLOW: BYE
    ALLOW: UPDATE
    ALLOW: PRACK
    SERVER: RTCC/5.0.0.0 MediationServer
    -------------

    Time: 13:34:55.009

    >>>>>>>>>>>>Outgoing SipMessage c=[<SipTcpConnection_303BCC8>], xx.xx.xx.11:5068->xx.xx.xx.50:35162
    SIP/2.0 180 Ringing
    FROM: "<external_number>"<sip:<external_number>@xx.xx.xx.50>;tag=as79f5e121
    TO: <sip:1000@xx.xx.xx.11:5068>;tag=1b534dd234;epid=1F1F675A78
    CSEQ: 102 INVITE
    CALL-ID: 0596b18b3a3c372e4aa2d8955de7c87c@xx.xx.xx.50:5060
    VIA: SIP/2.0/TCP xx.xx.xx.50:5060;branch=z9hG4bK0bf92c3c
    CONTACT: <sip:lync_pool_name:5068;transport=Tcp;maddr=xx.xx.xx.11>
    CONTENT-LENGTH: 0
    ALLOW: CANCEL
    ALLOW: BYE
    ALLOW: UPDATE
    ALLOW: PRACK
    SERVER: RTCC/5.0.0.0 MediationServer
    -------------

    Time: 13:34:55.344

    >>>>>>>>>>>>Outgoing SipMessage c=[<SipTcpConnection_303BCC8>], xx.xx.xx.11:5068->xx.xx.xx.50:35162
    SIP/2.0 200 OK
    FROM: "<external_number>"<sip:<external_number>@xx.xx.xx.50>;tag=as79f5e121
    TO: <sip:1000@xx.xx.xx.11:5068>;tag=1b534dd234;epid=1F1F675A78
    CSEQ: 102 INVITE
    CALL-ID: 0596b18b3a3c372e4aa2d8955de7c87c@xx.xx.xx.50:5060
    VIA: SIP/2.0/TCP xx.xx.xx.50:5060;branch=z9hG4bK0bf92c3c
    CONTACT: <sip:lync_pool_name:5068;transport=Tcp;maddr=xx.xx.xx.11>
    CONTENT-LENGTH: 249
    SUPPORTED: timer
    SUPPORTED: 100rel
    CONTENT-TYPE: application/sdp
    ALLOW: ACK
    ALLOW: CANCEL,BYE,INVITE,PRACK,UPDATE
    SERVER: RTCC/5.0.0.0 MediationServer
    Session-Expires: 1800;refresher=uas
    Min-SE: 90

    v=0
    o=- 914 1 IN IP4 xx.xx.xx.11
    s=session
    c=IN IP4 xx.xx.xx.11
    b=CT:1000
    t=0 0
    m=audio 49214 RTP/AVP 8 101
    c=IN IP4 xx.xx.xx.11
    a=rtcp:49215
    a=label:Audio
    a=sendrecv
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    ------------

    Time: 13:34:55.347

    <<<<<<<<<<<<Incoming SipMessage c=[<SipTcpConnection_303BCC8>], xx.xx.xx.11:5068<-xx.xx.xx.50:35162
    ACK sip:lync_pool_name:5068;transport=Tcp;maddr=xx.xx.xx.11 SIP/2.0
    FROM: "<external_number>" <sip:<external_number>@xx.xx.xx.50>;tag=as79f5e121
    TO: <sip:1000@xx.xx.xx.11:5068>;tag=1b534dd234
    CSEQ: 102 ACK
    CALL-ID: 0596b18b3a3c372e4aa2d8955de7c87c@xx.xx.xx.50:5060
    MAX-FORWARDS: 70
    VIA: SIP/2.0/TCP xx.xx.xx.50:5060;branch=z9hG4bK2a18f400
    CONTACT: <sip:<external_number>@xx.xx.xx.50:5060;transport=TCP>
    CONTENT-LENGTH: 0
    USER-AGENT: Asterisk PBX 11.8.1
    --------------

    Time: 13:35:16.757

    >>>>>>>>>>>>Outgoing SipMessage c=[<SipTcpConnection_303BCC8>], xx.xx.xx.11:5068->xx.xx.xx.50:35162
    BYE sip:<external_number>@xx.xx.xx.50:5060;transport=TCP SIP/2.0
    FROM: <sip:1000@xx.xx.xx.11:5068>;epid=1F1F675A78;tag=1b534dd234
    TO: <sip:<external_number>@xx.xx.xx.50>;tag=as79f5e121
    CSEQ: 1 BYE
    CALL-ID: 0596b18b3a3c372e4aa2d8955de7c87c@xx.xx.xx.50:5060
    MAX-FORWARDS: 70
    VIA: SIP/2.0/TCP xx.xx.xx.11:5068;branch=z9hG4bKea412ed
    CONTACT: <sip:lync_pool_name:5068;transport=Tcp;maddr=xx.xx.xx.11>
    CONTENT-LENGTH: 0
    USER-AGENT: RTCC/5.0.0.0 MediationServer
    ----------------

    Time: 13:35:16.760

    <<<<<<<<<<<<Incoming SipMessage c=[<SipTcpConnection_303BCC8>], xx.xx.xx.11:5068<-xx.xx.xx.50:35162
    SIP/2.0 200 OK
    FROM: <sip:1000@xx.xx.xx.11:5068>;epid=1F1F675A78;tag=1b534dd234
    TO: <sip:<external_number>@xx.xx.xx.50>;tag=as79f5e121
    CSEQ: 1 BYE
    CALL-ID: 0596b18b3a3c372e4aa2d8955de7c87c@xx.xx.xx.50:5060
    VIA: SIP/2.0/TCP xx.xx.xx.11:5068;branch=z9hG4bKea412ed;received=xx.xx.xx.11
    CONTENT-LENGTH: 0
    SUPPORTED: replaces, timer
    ALLOW: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    SERVER: Asterisk PBX 11.8.1

    Start time (INVITE) : 13:34:54

    End Time (BYE) : 13:35:16

    And no errors are in log report



    • Edited by LyncDummy Friday, October 17, 2014 6:05 PM
    Friday, October 17, 2014 1:37 PM
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    Hi,

    Please double check the network between FE Server and Media Gateway.

    Please also try to change the Touting method to others to test the issue again.

    Best Regards,

    Eason Huang

    Eason Huang
    TechNet Community Support

    Monday, October 20, 2014 9:04 AM
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    Hi, thank you for your reply

    FE Server and Media Gateway are in the same subnet, telnet command works as expected (ports are available)
    As to Routing methods - I didn't clearly understand where I have to change it ? 
    I'm not an expert in Lync (as you can see =)), but is it possible that my problem connected with aAstra LPE ? External call is coming to 2 endpoints: PC and aAstra LPE.
    Reception operator take a call on LPE (so call on PC ended). After some time (it may be 20sec or more than 1 minute) the call ended and on FE I can see SIP BYE from extension of RGS.
    Maybe I should troubleshoot RGS more deep ? Any ideas ?

    Monday, October 20, 2014 9:52 AM
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    I think, I've solved my problem - I replaced aAstra LPE on Polycom CX300. Two days without dropped calls.
    • Marked as answer by LyncDummy Thursday, October 23, 2014 5:45 AM
    Thursday, October 23, 2014 5:45 AM