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cdeacon

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  1. UM Auto Attendant

    in Microsoft OCS

    Posted

    Can you please get us the PBX log (with the SIP logging enabled)? or the wireshark network trace will be okay too.

     

    Here are my logs from the point that the auto attendant forwards the call to the extension:

     

    [7] 2008/09/29 14:12:48: SIP Rx tcp:172.16.116.104:5065:

    REFER sip:999@172.16.116.105:1181;transport=tcp SIP/2.0

    FROM: <sip:999@172.16.116.104;user=phone>;epid=06A5C4BC0B;tag=c19311cfd

    TO: <sip:999@172.16.116.104;user=phone>;tag=53059

    CSEQ: 1 REFER

    CALL-ID: 88725d22@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 172.16.116.104:5065;branch=z9hG4bK6d898cb2

    CONTACT: <sip:OBSRNDVMMX01.obsrnd.obsglobal.com:5065;transport=Tcp;maddr=172.16.116.104

    ;ms-opaque=ede52a3158c87bd0>;automata

    CONTENT-LENGTH: 0

    REFER-TO: <sip:212@172.16.116.105:1181;transport=tcp;user=phone>

    REFERRED-BY: <sip:999@172.16.116.104;user=phone>

    USER-AGENT: RTCC/3.0.0.0

     

    [7] 2008/09/29 14:12:48: SIP Tx tcp:172.16.116.104:5065:

    SIP/2.0 202 Accepted

    Via: SIP/2.0/TCP 172.16.116.104:5065;branch=z9hG4bK6d898cb2

    From: <sip:999@172.16.116.104;user=phone>;tag=c19311cfd;epid=06A5C4BC0B

    To: <sip:999@172.16.116.104;user=phone>;tag=53059

    Call-ID: 88725d22@pbx

    CSeq: 1 REFER

    Contact: <sip:999@172.16.116.105:1181;transport=tcp>

    User-Agent: pbxnsip-PBX/3.0.0.2998

    Content-Length: 0

     

    [5] 2008/09/29 14:12:48: Redirecting call to 212

    [5] 2008/09/29 14:12:48: Call 88725d22@pbx#53059: Last request not finished

    [7] 2008/09/29 14:12:48: SIP Tx tcp:172.16.116.104:5065:

    BYE sip:OBSRNDVMMX01.obsrnd.obsglobal.com:5065;transport=Tcp;maddr=172.16.116.104 SIP/2.0

    Via: SIP/2.0/TCP 172.16.116.105:1181;branch=z9hG4bK-d1b5d0ea5696fcc1fcaf23c8260956a0;rport

    From: "OBS Global" <sip:999@172.16.116.104;user=phone>;tag=53059

    To: <sip:999@172.16.116.104;user=phone>;tag=c19311cfd

    Call-ID: 88725d22@pbx

    CSeq: 25625 BYE

    Max-Forwards: 70

    Contact: <sip:999@172.16.116.105:1181;transport=tcp>

    RTP-RxStat: Dur=15,Pkt=528,Oct=90816,Underun=0

    RTP-TxStat: Dur=14,Pkt=699,Oct=117264

    Content-Length: 0

     

    [5] 2008/09/29 14:12:48: Dialplan ThinkTel: 212 goes to extension

    [8] 2008/09/29 14:12:48: Play audio_moh/noise.wav

    [7] 2008/09/29 14:12:48: UDP: Opening socket on port 54846

    [7] 2008/09/29 14:12:48: UDP: Opening socket on port 54847

    [7] 2008/09/29 14:12:48: SIP Tx tcp:172.16.116.102:5060:

    INVITE sip:+12049759212@172.16.116.102;transport=tcp SIP/2.0

    Via: SIP/2.0/TCP 172.16.116.105:1182;branch=z9hG4bK-c6c4c5fff867991162b4cdac8dfdb667;rport

    From: "Chris Deacon" <sip:2049820218@159.18.161.101:5060;transport=udp;user=phone>;tag=62205

    To: "OBS Global" <sip:999@obsrnd.obsglobal.com>

    Call-ID: 9b698194@pbx

    CSeq: 11279 INVITE

    Max-Forwards: 70

    Contact: <sip:212@172.16.116.105:1182;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/3.0.0.2998

    Alert-Info: <http://127.0.0.1/Bellcore-dr2>

    P-Asserted-Identity: "OBS Global" <sip:999@obsrnd.obsglobal.com>

    Content-Type: application/sdp

    Content-Length: 294

     

    v=0

    o=- 14904 14904 IN IP4 172.16.116.105

    s=-

    c=IN IP4 172.16.116.105

    t=0 0

    m=audio 54846 RTP/AVP 0 8 9 2 3 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:9 g722/8000

    a=rtpmap:2 g726-32/8000

    a=rtpmap:3 gsm/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=sendrecv

    [7] 2008/09/29 14:12:48: UDP: Opening socket on port 51212

    [7] 2008/09/29 14:12:48: UDP: Opening socket on port 51213

    [7] 2008/09/29 14:12:48: SIP Tx udp:172.16.116.200:2051:

    INVITE sip:212@172.16.116.200:2051;line=rpwp3qf5 SIP/2.0

    Via: SIP/2.0/UDP 172.16.116.105:5060;branch=z9hG4bK-6b60059434809ee17999ff5b626bdef1;rport

    From: "Chris Deacon" <sip:2049820218@159.18.161.101:5060;transport=udp;user=phone>;tag=11559

    To: "OBS Global" <sip:999@obsrnd.obsglobal.com>

    Call-ID: 7ec7d03a@pbx

    CSeq: 22046 INVITE

    Max-Forwards: 70

    Contact: <sip:212@172.16.116.105:5060;transport=udp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/3.0.0.2998

    Alert-Info: <http://127.0.0.1/Bellcore-dr2>

    P-Asserted-Identity: "OBS Global" <sip:999@obsrnd.obsglobal.com>

    Content-Type: application/sdp

    Content-Length: 294

     

    v=0

    o=- 54913 54913 IN IP4 172.16.116.105

    s=-

    c=IN IP4 172.16.116.105

    t=0 0

    m=audio 51212 RTP/AVP 0 8 9 2 3 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:9 g722/8000

    a=rtpmap:2 g726-32/8000

    a=rtpmap:3 gsm/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=sendrecv

    [7] 2008/09/29 14:12:48: 1B9E0136@159.18.161.101#b6d60dcb50: Media-aware pass-through mode

    [7] 2008/09/29 14:12:48: SIP Rx tcp:172.16.116.102:5060:

    SIP/2.0 100 Trying

    FROM: "Chris Deacon"<sip:2049820218@159.18.161.101:5060;transport=udp;user=phone>;tag=62205

    TO: "OBS Global"<sip:999@obsrnd.obsglobal.com>

    CSEQ: 11279 INVITE

    CALL-ID: 9b698194@pbx

    VIA: SIP/2.0/TCP 172.16.116.105:1182;branch=z9hG4bK-c6c4c5fff867991162b4cdac8dfdb667;rport

    CONTENT-LENGTH: 0

     

    [7] 2008/09/29 14:12:48: SIP Rx udp:172.16.116.200:2051:

    SIP/2.0 180 Ringing

    Via: SIP/2.0/UDP 172.16.116.105:5060;branch=z9hG4bK-6b60059434809ee17999ff5b626bdef1;rport=5060

    From: "Chris Deacon" <sip:2049820218@159.18.161.101:5060;transport=udp;user=phone>;tag=11559

    To: "OBS Global" <sip:999@obsrnd.obsglobal.com>;tag=yv82uxv0iq

    Call-ID: 7ec7d03a@pbx

    CSeq: 22046 INVITE

    Contact: <sip:212@172.16.116.200:2051;line=rpwp3qf5>;flow-id=1

    Require: 100rel

    RSeq: 1

    Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

    Allow-Events: talk, hold, refer, call-info

    Content-Length: 0

     

    [7] 2008/09/29 14:12:48: SIP Tx udp:172.16.116.200:2051:

    PRACK sip:212@172.16.116.200:2051;line=rpwp3qf5 SIP/2.0

    Via: SIP/2.0/UDP 172.16.116.105:5060;branch=z9hG4bK-636dea0941850e26b19f08a829dc50f9;rport

    From: "Chris Deacon" <sip:2049820218@159.18.161.101:5060;transport=udp;user=phone>;tag=11559

    To: "OBS Global" <sip:999@obsrnd.obsglobal.com>;tag=yv82uxv0iq

    Call-ID: 7ec7d03a@pbx

    CSeq: 22047 PRACK

    Max-Forwards: 70

    Contact: <sip:212@172.16.116.105:5060;transport=udp>

    RAck: 1 22046 INVITE

    P-Asserted-Identity: "OBS Global" <sip:999@obsrnd.obsglobal.com>

    Content-Length: 0

     

    [8] 2008/09/29 14:12:48: Play audio_en/ringback.wav

    [8] 2008/09/29 14:12:48: Call 88725d22@pbx#53059: Response does not correspond to open request

    [7] 2008/09/29 14:12:48: SIP Rx udp:172.16.116.200:2051:

    SIP/2.0 200 Ok

    Via: SIP/2.0/UDP 172.16.116.105:5060;branch=z9hG4bK-636dea0941850e26b19f08a829dc50f9;rport=5060

    From: "Chris Deacon" <sip:2049820218@159.18.161.101:5060;transport=udp;user=phone>;tag=11559

    To: "OBS Global" <sip:999@obsrnd.obsglobal.com>;tag=yv82uxv0iq

    Call-ID: 7ec7d03a@pbx

    CSeq: 22047 PRACK

    Contact: <sip:212@172.16.116.200:2051;line=rpwp3qf5>;flow-id=1

    Content-Length: 0

     

    [7] 2008/09/29 14:12:48: Call 7ec7d03a@pbx#11559: Clear last request

    [7] 2008/09/29 14:12:48: SIP Rx tcp:172.16.116.102:5060:

    SIP/2.0 183 Session Progress

    FROM: "Chris Deacon"<sip:2049820218@159.18.161.101:5060;transport=udp;user=phone>;tag=62205

    TO: OBS Global<sip:999@obsrnd.obsglobal.com>;epid=4414844359;tag=389ae44573

    CSEQ: 11279 INVITE

    CALL-ID: 9b698194@pbx

    VIA: SIP/2.0/TCP 172.16.116.105:1182;branch=z9hG4bK-c6c4c5fff867991162b4cdac8dfdb667;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0 MediationServer

     

    [7] 2008/09/29 14:12:49: SIP Rx tcp:172.16.116.104:5065:

    SIP/2.0 200 OK

    FROM: "OBS Global"<sip:999@172.16.116.104;user=phone>;tag=53059

    TO: <sip:999@172.16.116.104;user=phone>;tag=c19311cfd;epid=06A5C4BC0B

    CSEQ: 25625 BYE

    CALL-ID: 88725d22@pbx

    VIA: SIP/2.0/TCP 172.16.116.105:1181;branch=z9hG4bK-d1b5d0ea5696fcc1fcaf23c8260956a0;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0

     

    [8] 2008/09/29 14:12:49: Call 88725d22@pbx#53059: Response does not correspond to open request

    [5] 2008/09/29 14:12:49: BYE Response: Terminate 88725d22@pbx

    [7] 2008/09/29 14:12:49: Other Ports: 3

    [7] 2008/09/29 14:12:49: Call Port: 1B9E0136@159.18.161.101#b6d60dcb50

    [7] 2008/09/29 14:12:49: Call Port: 7ec7d03a@pbx#11559

    [7] 2008/09/29 14:12:49: Call Port: 9b698194@pbx#62205

    [7] 2008/09/29 14:12:49: SIP Rx tcp:172.16.116.102:5060:

    SIP/2.0 180 Ringing

    FROM: "Chris Deacon"<sip:2049820218@159.18.161.101:5060;transport=udp;user=phone>;tag=62205

    TO: OBS Global<sip:999@obsrnd.obsglobal.com>;epid=4414844359;tag=389ae44573

    CSEQ: 11279 INVITE

    CALL-ID: 9b698194@pbx

    VIA: SIP/2.0/TCP 172.16.116.105:1182;branch=z9hG4bK-c6c4c5fff867991162b4cdac8dfdb667;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0 MediationServer

     

    [7] 2008/09/29 14:12:50: SIP Tr udp:159.18.161.67:5060:

    SIP/2.0 200 Ok

    Via: SIP/2.0/UDP 159.18.161.67;branch=z9hG4bK8751.c2ab7894.0

    Via: SIP/2.0/UDP 159.18.161.101:5060;rport=5060;branch=z9hG4bK-0c26c6f69825fcf9328601343cba48ea-159.18.161.101-1

    Record-Route: <sip:2049758698@159.18.161.67;ftag=159.18.161.101+1+1ba2d9+59b98b75;lr=on>

    From: Chris Deacon <sip:2049820218@159.18.161.101:5060;transport=udp>;tag=159.18.161.101+1+1ba2d9+59b98b75;isup-oli=00

    To: <sip:2049758698@205.200.204.5>;tag=b6d60dcb50

    Call-ID: 1B9E0136@159.18.161.101

    CSeq: 684567237 INVITE

    Contact: <sip:2049758698@172.16.116.105:5060;transport=udp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/3.0.0.2998

    Content-Type: application/sdp

    Content-Length: 208

     

    v=0

    o=- 38924 38924 IN IP4 172.16.116.105

    s=-

    c=IN IP4 172.16.116.105

    t=0 0

    m=audio 62836 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=sendrec

  2. UM Auto Attendant

    in Microsoft OCS

    Posted

    Hey there. I've been setting up an OCS2007 with Exchange UM environment with success thus far with routing calls between SNOM320 sets and OC clients. I'm now looking into the UM portion, specifically the Auto Attendant. I can call in, receive the greeting and ask to be routed to an extension. Once it dials the extension the phone & communicator ring, which is all fine. My issue is that if I hang up the phone while the AA is calling the extension, the extension keeps ringing and then forwards to voice mail. How can I get the auto attendant to terminate the call when the outside caller has ended the call?

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