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callsanjeevat

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Posts posted by callsanjeevat

  1. Supervised Transfer Not working

    in Microsoft Speech Server

    Posted

    The issue is this - [7] 20110617102007: REFER from device type "RTCC/3.0.0.0" is not supported in this product.

     

    snomONE blue/yellow/free editions do not support REFER from 3rd party devices. We will inform the marketing / licensing group to see if they have any solution/workaround.

     

    can you please let me know as soon as possible? Also, can you please let me know if this works on snomONE Plus Edition. If Yes, what is the cost of this device for ~150 extensions.

     

    Thanks,

     

    Sanjeev.

  2. Supervised Transfer Not working

    in Microsoft Speech Server

    Posted

    Looking at the above observation & as user "snom ONE" suggested, the firewall on the PBX or on the gateway is dropping the UDP SIP packets. Is port 5060 opened for both TCP & UDP protocols?

     

    BTW, just FYI, if the logs are attached as a file, it is easier to read the post ;)

     

    Thanks for feedback. I turned off the windows firewall on the server. Now calls get routed from HG4000 to PBX on the UDP protocol. I changed the protocol to UDP on the HG4000 and also the trunk on PBX to state transport=udp. I place a call to HG4000. I gets routed to speech server through PBX. I answer the IVR Prompt. Speech server places teh consulation call through PBX. I receive the call. Speech server plays the IVR prompt. I enter the code. So far so good. As soon as I enter the code, My application does a transfer (asyncTransfer) and both th ephones hang up immediately.

     

    PBX log shows timeout. The FULL Log is attached. following is the section that shows timeout. 192.168.1.13:6060 is my trunk to speech server

     

     

    [6] 20110617102002: SIP TCP/TLS timeout on 192.168.1.13:6060, closing connection

    [5] 20110617102007: SIP Rx tcp:192.168.1.13:49341:

    REFER sip:0012012181411@192.168.1.13:49413;transport=tcp SIP/2.0

    FROM: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=84769DFAC6;tag=41d59776aa

    TO: <sip:0012012181411@192.168.1.12:5060;user=phone>;tag=44336

    CSEQ: 2 REFER

    CALL-ID: 8e255b39@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 192.168.1.13:49341;branch=z9hG4bK96a9afa1

    CONTACT: <sip:CommServer.creditfree.local:49341;transport=Tcp;maddr=192.168.1.13;ms-opaque=8e287a1fdc510a6b>;automata

    CONTENT-LENGTH: 0

    REFER-TO: <sip:12012181444@192.168.1.13:49413;transport=tcp;user=phone?REPLACES=0c17f648-a2b6-46ed-85a5-c1c0cdcbd336%3Bto-tag%3Ddaa53a4d54%3Bfrom-tag%3D9353450d8>

    REFERRED-BY: <sip:8768776075@192.168.1.13:6060;user=phone>

    USER-AGENT: RTCC/3.0.0.0

    Log.txt

  3. Supervised Transfer Not working

    in Microsoft Speech Server

    Posted

    Here is the corresponding PBX Log when Hg4000 transport = TCP. I hope this can be resolved either by getting PBX to work when Hypermedia transport=UDP or force PBX transport = TCP...

     

    INVITE sip:8768776075@192.168.1.13 SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-dc12fe7d8c7cfa32-1---d8754z-;rport

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.12:5060>

    To: <sip:8768776075@192.168.1.13>

    From: <sip:0012012181444@192.168.1.12:5060>;tag=ab429725

    Call-ID: MzU1ZjBkMjcwMzEzZmIyYzQ1ZGZjMDYxZTEzNTYzNGE.

    CSeq: 1 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub, 100rel, em

    User-Agent: HG4000/1.0

    Content-Length: 345

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4000 RTP/AVP 4 18 18 18 18 0 8 101

    a=rtpmap:4 G723/8000

    a=rtpmap:18 G729/8000

    a=rtpmap:18 G729a/8000

    a=rtpmap:18 G729b/8000

    a=rtpmap:18 G729ab/8000

    a=rtpmap:0 PCMU/8000

    a=rtpmap:8 PCMA/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110616172433: SIP Tx tcp:192.168.1.12:12330:

    SIP/2.0 100 Trying

    Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-dc12fe7d8c7cfa32-1---d8754z-;rport=12330

    From: <sip:0012012181444@192.168.1.12:5060>;tag=ab429725

    To: <sip:8768776075@192.168.1.13>;tag=bbfe18e7ec

    Call-ID: MzU1ZjBkMjcwMzEzZmIyYzQ1ZGZjMDYxZTEzNTYzNGE.

    CSeq: 1 INVITE

    Content-Length: 0

     

     

    [5] 20110616172433: Using <sip:0012012181444@192.168.1.12:5060;user=phone> as redirect source address

    [5] 20110616172433: SIP Tx tcp:192.168.1.13:6060:

    INVITE sip:8768776075@192.168.1.13:6060;user=phone SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:65150;branch=z9hG4bK-4a388cfccf9a8ae060fbe113e148ad53;rport

    From: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=31288

    To: <sip:8768776075@192.168.1.13:6060;user=phone>

    Call-ID: 99529a35@pbx

    CSeq: 1767 INVITE

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.13:65150;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.1.4009

    Diversion: <tel:42>;reason=unconditional;screen=no;privacy=off

    Related-Call-ID: MzU1ZjBkMjcwMzEzZmIyYzQ1ZGZjMDYxZTEzNTYzNGE.

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Type: application/sdp

    Content-Length: 327

     

    v=0

    o=- 17834 17834 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 36412 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=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

     

    [5] 20110616172433: SIP Rx tcp:192.168.1.13:6060:

    SIP/2.0 100 Trying

    FROM: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=31288

    TO: <sip:8768776075@192.168.1.13:6060;user=phone>

    CSEQ: 1767 INVITE

    CALL-ID: 99529a35@pbx

    VIA: SIP/2.0/TCP 192.168.1.13:65150;branch=z9hG4bK-4a388cfccf9a8ae060fbe113e148ad53;rport

    CONTENT-LENGTH: 0

     

     

    [5] 20110616172433: SIP Tx tcp:192.168.1.12:12330:

    SIP/2.0 183 Session Progress

    Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-dc12fe7d8c7cfa32-1---d8754z-;rport=12330

    From: <sip:0012012181444@192.168.1.12:5060>;tag=ab429725

    To: <sip:8768776075@192.168.1.13>;tag=bbfe18e7ec

    Call-ID: MzU1ZjBkMjcwMzEzZmIyYzQ1ZGZjMDYxZTEzNTYzNGE.

    CSeq: 1 INVITE

    Contact: <sip:8768776075@192.168.1.13:5060;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.1.4009

    Require: 100rel

    RSeq: 1

    Content-Type: application/sdp

    Content-Length: 253

     

    v=0

    o=- 28006 28006 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 54788 RTP/AVP 0 8 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

     

    [5] 20110616172433: SIP Rx tcp:192.168.1.13:6060:

    SIP/2.0 302 Moved Temporarily

    FROM: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=31288

    TO: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=1652901e79

    CSEQ: 1767 INVITE

    CALL-ID: 99529a35@pbx

    VIA: SIP/2.0/TCP 192.168.1.13:65150;branch=z9hG4bK-4a388cfccf9a8ae060fbe113e148ad53;rport

    CONTACT: <sip:8768776075@192.168.1.13:65122;user=phone;transport=Tcp;maddr=192.168.1.13;x-mss-call-id=99529a35%40pbx>

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0

     

     

    [7] 20110616172433: Call 99529a35@pbx: Clear last INVITE

    [5] 20110616172433: SIP Tx tcp:192.168.1.13:6060:

    ACK sip:8768776075@192.168.1.13:6060;user=phone SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:65150;branch=z9hG4bK-4a388cfccf9a8ae060fbe113e148ad53;rport

    From: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=31288

    To: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=1652901e79

    Call-ID: 99529a35@pbx

    CSeq: 1767 ACK

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.13:65150;transport=tcp>

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110616172433: Redirecting call

    [5] 20110616172433: SIP Tx tcp:192.168.1.13:65122:

    INVITE sip:8768776075@192.168.1.13:65122;user=phone;transport=Tcp;maddr=192.168.1.13;x-mss-call-id=99529a35%40pbx SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-9c613fdf39873449343bbb52519440be;rport

    From: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=31288

    To: <sip:8768776075@192.168.1.13:6060;user=phone>

    Call-ID: 99529a35@pbx

    CSeq: 1768 INVITE

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.13:65152;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.1.4009

    Diversion: <tel:42>;reason=unconditional;screen=no;privacy=off

    Related-Call-ID: MzU1ZjBkMjcwMzEzZmIyYzQ1ZGZjMDYxZTEzNTYzNGE.

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Type: application/sdp

    Content-Length: 327

     

    v=0

    o=- 17834 17834 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 36412 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=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

     

    [5] 20110616172433: SIP Rx tcp:192.168.1.13:65122:

    SIP/2.0 100 Trying

    FROM: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=31288

    TO: <sip:8768776075@192.168.1.13:6060;user=phone>

    CSEQ: 1768 INVITE

    CALL-ID: 99529a35@pbx

    VIA: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-9c613fdf39873449343bbb52519440be;rport

    CONTENT-LENGTH: 0

     

     

    [5] 20110616172433: SIP Rx tcp:192.168.1.13:65122:

    SIP/2.0 180 Ringing

    FROM: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=31288

    TO: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=834EFDD15A;tag=49062df6

    CSEQ: 1768 INVITE

    CALL-ID: 99529a35@pbx

    VIA: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-9c613fdf39873449343bbb52519440be;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0

     

     

    [5] 20110616172433: SIP Rx tcp:192.168.1.13:65122:

    SIP/2.0 200 OK

    FROM: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=31288

    TO: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=834EFDD15A;tag=49062df6

    CSEQ: 1768 INVITE

    CALL-ID: 99529a35@pbx

    VIA: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-9c613fdf39873449343bbb52519440be;rport

    CONTACT: <sip:CommServer.creditfree.local:65122;transport=Tcp;maddr=192.168.1.13>;automata

    CONTENT-LENGTH: 194

    CONTENT-TYPE: application/sdp

    ALLOW: UPDATE

    SERVER: RTCC/3.0.0.0

    ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify

     

    v=0

    o=- 0 0 IN IP4 192.168.1.13

    s=Microsoft Speech Server session

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 38400 RTP/AVP 0 8 101

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

     

    [7] 20110616172433: Call 99529a35@pbx: Clear last INVITE

    [5] 20110616172433: SIP Tx tcp:192.168.1.13:65122:

    ACK sip:CommServer.creditfree.local:65122;transport=Tcp;maddr=192.168.1.13 SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-ba37c71e815d8926de016156b10dbff6;rport

    From: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=31288

    To: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=49062df6;epid=834EFDD15A

    Call-ID: 99529a35@pbx

    CSeq: 1768 ACK

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.13:65152;transport=tcp>

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110616172433: SIP Tx tcp:192.168.1.12:12330:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-dc12fe7d8c7cfa32-1---d8754z-;rport=12330

    From: <sip:0012012181444@192.168.1.12:5060>;tag=ab429725

    To: <sip:8768776075@192.168.1.13>;tag=bbfe18e7ec

    Call-ID: MzU1ZjBkMjcwMzEzZmIyYzQ1ZGZjMDYxZTEzNTYzNGE.

    CSeq: 1 INVITE

    Contact: <sip:8768776075@192.168.1.13:5060;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.1.4009

    Content-Type: application/sdp

    Content-Length: 253

     

    v=0

    o=- 28006 28006 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 54788 RTP/AVP 0 8 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

     

    [5] 20110616172433: SIP Rx tcp:192.168.1.12:12330:

    ACK sip:8768776075@192.168.1.13:5060;transport=tcp SIP/2.0

    Via: SIP/2.0/TCP 192.168.0.2:5060;branch=z9hG4bK-d8754z-ebdfbe3d0bdb584f-1---d8754z-;rport

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.12:5060>

    To: <sip:8768776075@192.168.1.13>;tag=bbfe18e7ec

    From: <sip:0012012181444@192.168.1.12:5060>;tag=ab429725

    Call-ID: MzU1ZjBkMjcwMzEzZmIyYzQ1ZGZjMDYxZTEzNTYzNGE.

    CSeq: 1 ACK

    User-Agent: HG4000/1.0

    Content-Length: 0

     

     

    [5] 20110616172441: SIP Rx tcp:192.168.1.12:12330:

    BYE sip:8768776075@192.168.1.13:5060;transport=tcp SIP/2.0

    Via: SIP/2.0/TCP 192.168.0.2:5060;branch=z9hG4bK-d8754z-b83b342fda56d142-1---d8754z-;rport

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.12:5060>

    To: <sip:8768776075@192.168.1.13>;tag=bbfe18e7ec

    From: <sip:0012012181444@192.168.1.12:5060>;tag=ab429725

    Call-ID: MzU1ZjBkMjcwMzEzZmIyYzQ1ZGZjMDYxZTEzNTYzNGE.

    CSeq: 2 BYE

    User-Agent: HG4000/1.0

    Reason: SIP;description="ACK not received"

    Content-Length: 0

     

     

    [5] 20110616172441: SIP Tx tcp:192.168.1.12:12330:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 192.168.0.2:5060;branch=z9hG4bK-d8754z-b83b342fda56d142-1---d8754z-;rport=12330;received=192.168.1.12

    From: <sip:0012012181444@192.168.1.12:5060>;tag=ab429725

    To: <sip:8768776075@192.168.1.13>;tag=bbfe18e7ec

    Call-ID: MzU1ZjBkMjcwMzEzZmIyYzQ1ZGZjMDYxZTEzNTYzNGE.

    CSeq: 2 BYE

    Contact: <sip:8768776075@192.168.1.13:5060;transport=tcp>

    User-Agent: snom-PBX/2011-4.2.1.4009

    Content-Length: 0

     

     

    [5] 20110616172441: SIP Tx tcp:192.168.1.13:65122:

    BYE sip:CommServer.creditfree.local:65122;transport=Tcp;maddr=192.168.1.13 SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-897f7badaf3d7ad46c5b67c81342b878;rport

    From: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=31288

    To: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=49062df6

    Call-ID: 99529a35@pbx

    CSeq: 1769 BYE

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.13:65152;transport=tcp>

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110616172441: SIP Rx tcp:192.168.1.13:65122:

    SIP/2.0 200 OK

    FROM: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=31288

    TO: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=49062df6;epid=834EFDD15A

    CSEQ: 1769 BYE

    CALL-ID: 99529a35@pbx

    VIA: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-897f7badaf3d7ad46c5b67c81342b878;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0

     

     

    [7] 20110616172441: Call 99529a35@pbx: Clear last request

    [5] 20110616172441: BYE Response: Terminate 99529a35@pbx

    [5] 20110616172442: SIP Rx udp:192.168.1.13:41032:

    SUBSCRIBE sip:1000@192.168.1.13 SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:41032;branch=z9hG4bK-d8754z-eadee8af60c2b8cf-1---d8754z-;rport

    Max-Forwards: 70

    Contact: <sip:1000@192.168.1.13:41032>

    To: "1000"<sip:1000@192.168.1.13>

    From: "1000"<sip:1000@192.168.1.13>;tag=ba371dbd

    Call-ID: ZjE0YjMyMmVjYWNiOGZkZTkxOGIyZTE3ODhlYWY1YzU.

    CSeq: 1 SUBSCRIBE

    Expires: 300

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

    User-Agent: X-Lite 4 release 4.0 stamp 58832

    Event: message-summary

    Content-Length: 0

     

     

    [5] 20110616172442: SIP Tx udp:192.168.1.13:41032:

    SIP/2.0 404 Not Found

    Via: SIP/2.0/UDP 192.168.1.13:41032;branch=z9hG4bK-d8754z-eadee8af60c2b8cf-1---d8754z-;rport=41032

    From: "1000" <sip:1000@192.168.1.13>;tag=ba371dbd

    To: "1000" <sip:1000@192.168.1.13>;tag=ba7af3f391

    Call-ID: ZjE0YjMyMmVjYWNiOGZkZTkxOGIyZTE3ODhlYWY1YzU.

    CSeq: 1 SUBSCRIBE

    Content-Length: 0

     

     

    [5] 20110616172502: SIP Rx tcp:192.168.1.12:12330:

    INVITE sip:8768776075@192.168.1.13 SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-450bb10b9ca6da46-1---d8754z-;rport

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.12:5060>

    To: <sip:8768776075@192.168.1.13>

    From: <sip:0012012181444@192.168.1.12:5060>;tag=29823d4a

    Call-ID: YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

    CSeq: 1 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub, 100rel, em

    User-Agent: HG4000/1.0

    Content-Length: 345

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4000 RTP/AVP 4 18 18 18 18 0 8 101

    a=rtpmap:4 G723/8000

    a=rtpmap:18 G729/8000

    a=rtpmap:18 G729a/8000

    a=rtpmap:18 G729b/8000

    a=rtpmap:18 G729ab/8000

    a=rtpmap:0 PCMU/8000

    a=rtpmap:8 PCMA/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110616172502: SIP Tx tcp:192.168.1.12:12330:

    SIP/2.0 100 Trying

    Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-450bb10b9ca6da46-1---d8754z-;rport=12330

    From: <sip:0012012181444@192.168.1.12:5060>;tag=29823d4a

    To: <sip:8768776075@192.168.1.13>;tag=92de13d950

    Call-ID: YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

    CSeq: 1 INVITE

    Content-Length: 0

     

     

    [5] 20110616172502: Using <sip:0012012181444@192.168.1.12:5060;user=phone> as redirect source address

    [5] 20110616172502: SIP Tx tcp:192.168.1.13:6060:

    INVITE sip:8768776075@192.168.1.13:6060;user=phone SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:65150;branch=z9hG4bK-ddf022c07713811d4d214d87b8100b76;rport

    From: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

    To: <sip:8768776075@192.168.1.13:6060;user=phone>

    Call-ID: 5d084739@pbx

    CSeq: 16994 INVITE

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.13:65150;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.1.4009

    Diversion: <tel:42>;reason=unconditional;screen=no;privacy=off

    Related-Call-ID: YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Type: application/sdp

    Content-Length: 327

     

    v=0

    o=- 38841 38841 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 49044 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=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

     

    [5] 20110616172502: SIP Rx tcp:192.168.1.13:6060:

    SIP/2.0 100 Trying

    FROM: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

    TO: <sip:8768776075@192.168.1.13:6060;user=phone>

    CSEQ: 16994 INVITE

    CALL-ID: 5d084739@pbx

    VIA: SIP/2.0/TCP 192.168.1.13:65150;branch=z9hG4bK-ddf022c07713811d4d214d87b8100b76;rport

    CONTENT-LENGTH: 0

     

     

    [5] 20110616172502: SIP Tx tcp:192.168.1.12:12330:

    SIP/2.0 183 Session Progress

    Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-450bb10b9ca6da46-1---d8754z-;rport=12330

    From: <sip:0012012181444@192.168.1.12:5060>;tag=29823d4a

    To: <sip:8768776075@192.168.1.13>;tag=92de13d950

    Call-ID: YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

    CSeq: 1 INVITE

    Contact: <sip:8768776075@192.168.1.13:5060;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.1.4009

    Require: 100rel

    RSeq: 1

    Content-Type: application/sdp

    Content-Length: 253

     

    v=0

    o=- 57347 57347 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 53830 RTP/AVP 0 8 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

     

    [5] 20110616172502: SIP Rx tcp:192.168.1.13:6060:

    SIP/2.0 302 Moved Temporarily

    FROM: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

    TO: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=7a5d8de81d

    CSEQ: 16994 INVITE

    CALL-ID: 5d084739@pbx

    VIA: SIP/2.0/TCP 192.168.1.13:65150;branch=z9hG4bK-ddf022c07713811d4d214d87b8100b76;rport

    CONTACT: <sip:8768776075@192.168.1.13:65122;user=phone;transport=Tcp;maddr=192.168.1.13;x-mss-call-id=5d084739%40pbx>

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0

     

     

    [7] 20110616172502: Call 5d084739@pbx: Clear last INVITE

    [5] 20110616172502: SIP Tx tcp:192.168.1.13:6060:

    ACK sip:8768776075@192.168.1.13:6060;user=phone SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:65150;branch=z9hG4bK-ddf022c07713811d4d214d87b8100b76;rport

    From: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

    To: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=7a5d8de81d

    Call-ID: 5d084739@pbx

    CSeq: 16994 ACK

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.13:65150;transport=tcp>

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110616172502: Redirecting call

    [5] 20110616172502: SIP Tx tcp:192.168.1.13:65122:

    INVITE sip:8768776075@192.168.1.13:65122;user=phone;transport=Tcp;maddr=192.168.1.13;x-mss-call-id=5d084739%40pbx SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-25324b29831544a6365c1bd4b82aa8e5;rport

    From: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

    To: <sip:8768776075@192.168.1.13:6060;user=phone>

    Call-ID: 5d084739@pbx

    CSeq: 16995 INVITE

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.13:65152;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.1.4009

    Diversion: <tel:42>;reason=unconditional;screen=no;privacy=off

    Related-Call-ID: YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Type: application/sdp

    Content-Length: 327

     

    v=0

    o=- 38841 38841 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 49044 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=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

     

    [5] 20110616172502: SIP Rx tcp:192.168.1.13:65122:

    SIP/2.0 100 Trying

    FROM: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

    TO: <sip:8768776075@192.168.1.13:6060;user=phone>

    CSEQ: 16995 INVITE

    CALL-ID: 5d084739@pbx

    VIA: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-25324b29831544a6365c1bd4b82aa8e5;rport

    CONTENT-LENGTH: 0

     

     

    [5] 20110616172502: SIP Rx tcp:192.168.1.13:65122:

    SIP/2.0 180 Ringing

    FROM: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

    TO: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=834EFDD15A;tag=bb7e72d89

    CSEQ: 16995 INVITE

    CALL-ID: 5d084739@pbx

    VIA: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-25324b29831544a6365c1bd4b82aa8e5;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0

     

     

    [5] 20110616172502: SIP Rx tcp:192.168.1.13:65122:

    SIP/2.0 200 OK

    FROM: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

    TO: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=834EFDD15A;tag=bb7e72d89

    CSEQ: 16995 INVITE

    CALL-ID: 5d084739@pbx

    VIA: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-25324b29831544a6365c1bd4b82aa8e5;rport

    CONTACT: <sip:CommServer.creditfree.local:65122;transport=Tcp;maddr=192.168.1.13>;automata

    CONTENT-LENGTH: 194

    CONTENT-TYPE: application/sdp

    ALLOW: UPDATE

    SERVER: RTCC/3.0.0.0

    ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify

     

    v=0

    o=- 0 0 IN IP4 192.168.1.13

    s=Microsoft Speech Server session

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 50944 RTP/AVP 0 8 101

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

     

    [7] 20110616172502: Call 5d084739@pbx: Clear last INVITE

    [5] 20110616172502: SIP Tx tcp:192.168.1.13:65122:

    ACK sip:CommServer.creditfree.local:65122;transport=Tcp;maddr=192.168.1.13 SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-eee5763cf0213961f4bf7dd0f5495d8e;rport

    From: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

    To: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=bb7e72d89;epid=834EFDD15A

    Call-ID: 5d084739@pbx

    CSeq: 16995 ACK

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.13:65152;transport=tcp>

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110616172502: SIP Tx tcp:192.168.1.12:12330:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-450bb10b9ca6da46-1---d8754z-;rport=12330

    From: <sip:0012012181444@192.168.1.12:5060>;tag=29823d4a

    To: <sip:8768776075@192.168.1.13>;tag=92de13d950

    Call-ID: YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

    CSeq: 1 INVITE

    Contact: <sip:8768776075@192.168.1.13:5060;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.1.4009

    Content-Type: application/sdp

    Content-Length: 253

     

    v=0

    o=- 57347 57347 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 53830 RTP/AVP 0 8 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

     

    [5] 20110616172502: SIP Rx tcp:192.168.1.12:12330:

    ACK sip:8768776075@192.168.1.13:5060;transport=tcp SIP/2.0

    Via: SIP/2.0/TCP 192.168.0.2:5060;branch=z9hG4bK-d8754z-90e6572bc2e79711-1---d8754z-;rport

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.12:5060>

    To: <sip:8768776075@192.168.1.13>;tag=92de13d950

    From: <sip:0012012181444@192.168.1.12:5060>;tag=29823d4a

    Call-ID: YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

    CSeq: 1 ACK

    User-Agent: HG4000/1.0

    Content-Length: 0

     

     

    [5] 20110616172531: SIP Rx tcp:192.168.1.13:65122:

    INVITE sip:0012012181444@192.168.1.13:65152;transport=tcp SIP/2.0

    FROM: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=834EFDD15A;tag=bb7e72d89

    TO: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

    CSEQ: 1 INVITE

    CALL-ID: 5d084739@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 192.168.1.13:65122;branch=z9hG4bK3ffd794d

    CONTACT: <sip:CommServer.creditfree.local:65122;transport=Tcp;maddr=192.168.1.13;ms-opaque=e6d2caedfe360c75>;automata

    CONTENT-LENGTH: 206

    USER-AGENT: RTCC/3.0.0.0

    CONTENT-TYPE: application/sdp

     

    v=0

    o=- 0 0 IN IP4 192.168.1.13

    s=Microsoft Speech Server session

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 50944 RTP/AVP 0 8 101

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=sendonly

    a=ptime:20

     

    [5] 20110616172531: SIP Tx tcp:192.168.1.13:65122:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 192.168.1.13:65122;branch=z9hG4bK3ffd794d

    From: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=bb7e72d89;epid=834EFDD15A

    To: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

    Call-ID: 5d084739@pbx

    CSeq: 1 INVITE

    Contact: <sip:0012012181444@192.168.1.13:65152;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.1.4009

    Content-Type: application/sdp

    Content-Length: 265

     

    v=0

    o=- 38841 38841 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 49044 RTP/AVP 0 8 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=recvonly

     

    [5] 20110616172531: SIP Rx tcp:192.168.1.13:65122:

    ACK sip:0012012181444@192.168.1.13:65152;transport=tcp SIP/2.0

    FROM: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=834EFDD15A;tag=bb7e72d89

    TO: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

    CSEQ: 1 ACK

    CALL-ID: 5d084739@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 192.168.1.13:65122;branch=z9hG4bKc5459f2d

    CONTENT-LENGTH: 0

    USER-AGENT: RTCC/3.0.0.0

     

     

    [5] 20110616172531: SIP Rx tcp:192.168.1.13:65159:

    INVITE sip:13109644430@192.168.1.13:5060;transport=tcp SIP/2.0

    FROM: <sip:0012012181444@192.168.1.12:5060;transport=tcp>;epid=834EFDD15A;tag=d01782b7fb

    TO: <sip:13109644430@192.168.1.13:5060;transport=tcp>

    CSEQ: 2 INVITE

    CALL-ID: 9b5e977d-efb2-4bf6-99e7-b07c4e631f0c

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 192.168.1.13:65159;branch=z9hG4bKf27d5133

    CONTACT: <sip:CommServer.creditfree.local:65122;transport=Tcp;maddr=192.168.1.13;ms-opaque=e6d2caedfe360c75>;automata

    CONTENT-LENGTH: 336

    USER-AGENT: RTCC/3.0.0.0

    CONTENT-TYPE: application/sdp

    ALLOW: UPDATE

    ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify

     

    v=0

    o=- 0 0 IN IP4 192.168.1.13

    s=Microsoft Speech Server session

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 38400 RTP/AVP 114 115 4 0 8 97 101

    a=rtpmap:114 x-msrta/16000

    a=fmtp:114 bitrate=29000

    a=rtpmap:115 x-msrta/8000

    a=fmtp:115 bitrate=11800

    a=rtpmap:97 RED/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

     

    [5] 20110616172531: SIP Tx tcp:192.168.1.13:65159:

    SIP/2.0 100 Trying

    Via: SIP/2.0/TCP 192.168.1.13:65159;branch=z9hG4bKf27d5133

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp>;tag=d01782b7fb;epid=834EFDD15A

    To: <sip:13109644430@192.168.1.13:5060;transport=tcp>;tag=51fdee39e8

    Call-ID: 9b5e977d-efb2-4bf6-99e7-b07c4e631f0c

    CSeq: 2 INVITE

    Content-Length: 0

     

     

    [5] 20110616172531: Using <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone> as redirect source address

    [5] 20110616172531: SIP Tx tcp:192.168.1.12:5060:

    INVITE sip:13109644430@192.168.1.12;user=phone SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    To: <sip:13109644430@192.168.1.12;user=phone>

    Call-ID: 7e61c163@pbx

    CSeq: 19524 INVITE

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.13:65160;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.1.4009

    Diversion: <tel:42>;reason=unconditional;screen=no;privacy=off

    Related-Call-ID: 9b5e977d-efb2-4bf6-99e7-b07c4e631f0c

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Type: application/sdp

    Content-Length: 327

     

    v=0

    o=- 44371 44371 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 48640 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=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

     

    [5] 20110616172531: SIP Tx tcp:192.168.1.13:65159:

    SIP/2.0 183 Session Progress

    Via: SIP/2.0/TCP 192.168.1.13:65159;branch=z9hG4bKf27d5133

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp>;tag=d01782b7fb;epid=834EFDD15A

    To: <sip:13109644430@192.168.1.13:5060;transport=tcp>;tag=51fdee39e8

    Call-ID: 9b5e977d-efb2-4bf6-99e7-b07c4e631f0c

    CSeq: 2 INVITE

    Contact: <sip:13109644430@192.168.1.13:5060;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.1.4009

    Content-Type: application/sdp

    Content-Length: 263

     

    v=0

    o=- 8882 8882 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 32724 RTP/AVP 0 8 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

     

    [5] 20110616172532: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 183 Session Progress

    Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport=65160

    Contact: <sip:13109644430@192.168.1.12:5060;user=phone>

    To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    Call-ID: 7e61c163@pbx

    CSeq: 19524 INVITE

    Content-Type: application/sdp

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4010 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110616172539: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport=65160

    Contact: <sip:13109644430@192.168.1.12:5060;user=phone>

    To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    Call-ID: 7e61c163@pbx

    CSeq: 19524 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4010 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [7] 20110616172539: Call 7e61c163@pbx: Clear last INVITE

    [5] 20110616172539: SIP Tx udp:192.168.1.12:5060:

    ACK sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-37e2311adbf79cf72758680717621611;rport

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

    Call-ID: 7e61c163@pbx

    CSeq: 19524 ACK

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110616172539: SIP Tx tcp:192.168.1.13:65159:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 192.168.1.13:65159;branch=z9hG4bKf27d5133

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp>;tag=d01782b7fb;epid=834EFDD15A

    To: <sip:13109644430@192.168.1.13:5060;transport=tcp>;tag=51fdee39e8

    Call-ID: 9b5e977d-efb2-4bf6-99e7-b07c4e631f0c

    CSeq: 2 INVITE

    Contact: <sip:13109644430@192.168.1.13:5060;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.1.4009

    Content-Type: application/sdp

    Content-Length: 263

     

    v=0

    o=- 8882 8882 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 32724 RTP/AVP 0 8 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

     

    [5] 20110616172539: SIP Rx tcp:192.168.1.13:65159:

    ACK sip:13109644430@192.168.1.13:5060;transport=tcp SIP/2.0

    FROM: <sip:0012012181444@192.168.1.12:5060;transport=tcp>;epid=834EFDD15A;tag=d01782b7fb

    TO: <sip:13109644430@192.168.1.13:5060;transport=tcp>;tag=51fdee39e8

    CSEQ: 2 ACK

    CALL-ID: 9b5e977d-efb2-4bf6-99e7-b07c4e631f0c

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 192.168.1.13:65159;branch=z9hG4bKb96c31c1

    CONTENT-LENGTH: 0

    USER-AGENT: RTCC/3.0.0.0

     

     

    [5] 20110616172539: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport=65160

    Contact: <sip:13109644430@192.168.1.12:5060;user=phone>

    To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    Call-ID: 7e61c163@pbx

    CSeq: 19524 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4010 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110616172539: SIP Tx udp:192.168.1.12:5060:

    ACK sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-37e2311adbf79cf72758680717621611;rport

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

    Call-ID: 7e61c163@pbx

    CSeq: 19524 ACK

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110616172540: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport=65160

    Contact: <sip:13109644430@192.168.1.12:5060;user=phone>

    To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    Call-ID: 7e61c163@pbx

    CSeq: 19524 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4010 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110616172540: SIP Tx udp:192.168.1.12:5060:

    ACK sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-37e2311adbf79cf72758680717621611;rport

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

    Call-ID: 7e61c163@pbx

    CSeq: 19524 ACK

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110616172542: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport=65160

    Contact: <sip:13109644430@192.168.1.12:5060;user=phone>

    To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    Call-ID: 7e61c163@pbx

    CSeq: 19524 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4010 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110616172542: SIP Tx udp:192.168.1.12:5060:

    ACK sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-37e2311adbf79cf72758680717621611;rport

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

    Call-ID: 7e61c163@pbx

    CSeq: 19524 ACK

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110616172546: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport=65160

    Contact: <sip:13109644430@192.168.1.12:5060;user=phone>

    To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    Call-ID: 7e61c163@pbx

    CSeq: 19524 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4010 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110616172546: SIP Tx udp:192.168.1.12:5060:

    ACK sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-37e2311adbf79cf72758680717621611;rport

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

    Call-ID: 7e61c163@pbx

    CSeq: 19524 ACK

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110616172550: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport=65160

    Contact: <sip:13109644430@192.168.1.12:5060;user=phone>

    To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    Call-ID: 7e61c163@pbx

    CSeq: 19524 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4010 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110616172550: SIP Tx udp:192.168.1.12:5060:

    ACK sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-37e2311adbf79cf72758680717621611;rport

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

    Call-ID: 7e61c163@pbx

    CSeq: 19524 ACK

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110616172554: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport=65160

    Contact: <sip:13109644430@192.168.1.12:5060;user=phone>

    To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    Call-ID: 7e61c163@pbx

    CSeq: 19524 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4010 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110616172554: SIP Tx udp:192.168.1.12:5060:

    ACK sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-37e2311adbf79cf72758680717621611;rport

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

    Call-ID: 7e61c163@pbx

    CSeq: 19524 ACK

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110616172558: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport=65160

    Contact: <sip:13109644430@192.168.1.12:5060;user=phone>

    To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    Call-ID: 7e61c163@pbx

    CSeq: 19524 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4010 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110616172558: SIP Tx udp:192.168.1.12:5060:

    ACK sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-37e2311adbf79cf72758680717621611;rport

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

    Call-ID: 7e61c163@pbx

    CSeq: 19524 ACK

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [6] 20110616172602: SIP TCP/TLS timeout on 192.168.1.13:6060, closing connection

    [5] 20110616172602: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport=65160

    Contact: <sip:13109644430@192.168.1.12:5060;user=phone>

    To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    Call-ID: 7e61c163@pbx

    CSeq: 19524 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4010 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110616172602: SIP Tx udp:192.168.1.12:5060:

    ACK sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-37e2311adbf79cf72758680717621611;rport

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

    Call-ID: 7e61c163@pbx

    CSeq: 19524 ACK

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110616172606: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport=65160

    Contact: <sip:13109644430@192.168.1.12:5060;user=phone>

    To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    Call-ID: 7e61c163@pbx

    CSeq: 19524 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4010 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110616172606: SIP Tx udp:192.168.1.12:5060:

    ACK sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-37e2311adbf79cf72758680717621611;rport

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

    Call-ID: 7e61c163@pbx

    CSeq: 19524 ACK

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110616172610: SIP Rx tcp:192.168.1.12:12330:

    BYE sip:8768776075@192.168.1.13:5060;transport=tcp SIP/2.0

    Via: SIP/2.0/TCP 192.168.0.2:5060;branch=z9hG4bK-d8754z-48372c12e9252c06-1---d8754z-;rport

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.12:5060>

    To: <sip:8768776075@192.168.1.13>;tag=92de13d950

    From: <sip:0012012181444@192.168.1.12:5060>;tag=29823d4a

    Call-ID: YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

    CSeq: 2 BYE

    User-Agent: HG4000/1.0

    Reason: SIP;description="ACK not received"

    Content-Length: 0

     

     

    [5] 20110616172610: SIP Tx tcp:192.168.1.12:12330:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 192.168.0.2:5060;branch=z9hG4bK-d8754z-48372c12e9252c06-1---d8754z-;rport=12330;received=192.168.1.12

    From: <sip:0012012181444@192.168.1.12:5060>;tag=29823d4a

    To: <sip:8768776075@192.168.1.13>;tag=92de13d950

    Call-ID: YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

    CSeq: 2 BYE

    Contact: <sip:8768776075@192.168.1.13:5060;transport=tcp>

    User-Agent: snom-PBX/2011-4.2.1.4009

    Content-Length: 0

     

     

    [5] 20110616172610: SIP Tx tcp:192.168.1.13:65122:

    BYE sip:CommServer.creditfree.local:65122;transport=Tcp;maddr=192.168.1.13 SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-cd85a3105db01a4a2724cb991e7945f2;rport

    From: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

    To: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=bb7e72d89

    Call-ID: 5d084739@pbx

    CSeq: 16996 BYE

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.13:65152;transport=tcp>

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110616172610: SIP Rx tcp:192.168.1.13:65122:

    SIP/2.0 200 OK

    FROM: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

    TO: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=bb7e72d89;epid=834EFDD15A

    CSEQ: 16996 BYE

    CALL-ID: 5d084739@pbx

    VIA: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-cd85a3105db01a4a2724cb991e7945f2;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0

     

     

    [7] 20110616172610: Call 5d084739@pbx: Clear last request

    [5] 20110616172610: BYE Response: Terminate 5d084739@pbx

    [5] 20110616172610: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport=65160

    Contact: <sip:13109644430@192.168.1.12:5060;user=phone>

    To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    Call-ID: 7e61c163@pbx

    CSeq: 19524 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4010 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110616172610: SIP Tx udp:192.168.1.12:5060:

    ACK sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-37e2311adbf79cf72758680717621611;rport

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

    Call-ID: 7e61c163@pbx

    CSeq: 19524 ACK

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [6] 20110616172710: SIP TCP/TLS timeout on 192.168.1.13:65122, closing connection

    [6] 20110616172710: SIP TCP/TLS timeout on 192.168.1.12:5060, closing connection

    [5] 20110616172745: SIP Rx udp:192.168.1.13:41032:

    SUBSCRIBE sip:1000@192.168.1.13 SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:41032;branch=z9hG4bK-d8754z-f4f6dcd533171c22-1---d8754z-;rport

    Max-Forwards: 70

    Contact: <sip:1000@192.168.1.13:41032>

    To: "1000"<sip:1000@192.168.1.13>

    From: "1000"<sip:1000@192.168.1.13>;tag=565b18cf

    Call-ID: MDY2NDEzMDgzOTlmMTM1OWI0ZjNiOTFjNGE4MDZmMTA.

    CSeq: 1 SUBSCRIBE

    Expires: 300

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

    User-Agent: X-Lite 4 release 4.0 stamp 58832

    Event: message-summary

    Content-Length: 0

     

     

    [5] 20110616172745: SIP Tx udp:192.168.1.13:41032:

    SIP/2.0 404 Not Found

    Via: SIP/2.0/UDP 192.168.1.13:41032;branch=z9hG4bK-d8754z-f4f6dcd533171c22-1---d8754z-;rport=41032

    From: "1000" <sip:1000@192.168.1.13>;tag=565b18cf

    To: "1000" <sip:1000@192.168.1.13>;tag=badc876e36

    Call-ID: MDY2NDEzMDgzOTlmMTM1OWI0ZjNiOTFjNGE4MDZmMTA.

    CSeq: 1 SUBSCRIBE

    Content-Length: 0

     

     

    [5] 20110616172811: SIP Tx udp:192.168.1.12:5060:

    BYE sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8a685730b04cc2548150ba741c72b373;rport

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

    Call-ID: 7e61c163@pbx

    CSeq: 19525 BYE

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

    Reason: Preemption;cause=3;text="No Media"

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110616172811: SIP Tr udp:192.168.1.12:5060:

    BYE sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8a685730b04cc2548150ba741c72b373;rport

    From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

    To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

    Call-ID: 7e61c163@pbx

    CSeq: 19525 BYE

    Max-Forwards: 70

    Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

    Reason: Preemption;cause=3;text="No Media"

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110616172843: Last message repeated 10 times

    [7] 20110616172843: Call 7e61c163@pbx: Clear last request

    [5] 20110616172843: BYE Response: Terminate 7e61c163@pbx

    [5] 20110616172843: SIP Tx tcp:192.168.1.13:65159:

    BYE sip:CommServer.creditfree.local:65122;transport=Tcp;maddr=192.168.1.13;ms-opaque=e6d2caedfe360c75 SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:5060;branch=z9hG4bK-0e77f88bc248991b805201ccdbd1466a;rport

    From: <sip:13109644430@192.168.1.13:5060;transport=tcp>;tag=51fdee39e8

    To: <sip:0012012181444@192.168.1.12:5060;transport=tcp>;tag=d01782b7fb;epid=834EFDD15A

    Call-ID: 9b5e977d-efb2-4bf6-99e7-b07c4e631f0c

    CSeq: 21455 BYE

    Max-Forwards: 70

    Contact: <sip:13109644430@192.168.1.13:5060;transport=tcp>

    Content-Length: 0

     

     

    [5] 20110616172843: SIP Rx tcp:192.168.1.13:65159:

    SIP/2.0 200 OK

    FROM: <sip:13109644430@192.168.1.13:5060;transport=tcp>;tag=51fdee39e8

    TO: <sip:0012012181444@192.168.1.12:5060;transport=tcp>;tag=d01782b7fb;epid=834EFDD15A

    CSEQ: 21455 BYE

    CALL-ID: 9b5e977d-efb2-4bf6-99e7-b07c4e631f0c

    VIA: SIP/2.0/TCP 192.168.1.13:5060;branch=z9hG4bK-0e77f88bc248991b805201ccdbd1466a;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0

  4. Supervised Transfer Not working

    in Microsoft Speech Server

    Posted

    This the LOG for HG4000 when Transport=TCP. Please note that Call connects. I Do an IVR and Do a supervised transfer. The outgoing call connects too. I do an IVR and authenticate the called party. Then I do a Transfer. I cannot hear anything on either phones.

     

    HG 4000 when Transport = TCP

     

    [17/06-17:24:59.113] [debug] exProceedEvent AudioCodes event: EV_ENHANCED_BIT_STATUS

    [17/06-17:24:59.114] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:01.694] HMCServer received [ping] from 127.0.0.1:40150

    [17/06-17:25:01.695] processRequestLines: reply: "pong " for client: 13

    [17/06-17:25:04.243] * Received Packet: GenericReply /#90/@2b/x0,1/I3032/G

    [17/06-17:25:05.503] * Received Packet: Dialing /A21/I456/o0/H5/S1/h14/s0/n18768776075/N0012012181444/i1

    [17/06-17:25:05.505] updateReplyContext: no effect

    [17/06-17:25:05.506] packStr=/A21/I456/o0/H5/S1/h14/s0/n18768776075/N0012012181444/i1

    [17/06-17:25:05.507] strHW=5

    [17/06-17:25:05.507] strTS=1

    [17/06-17:25:05.508] No filters required for in address 21.1

    [17/06-17:25:05.509] getApplication for:21.1-VoIP

    [17/06-17:25:05.533] strCard=[21], nClientID=0

    [17/06-17:25:05.534] To VoIP Trigger:Dialing /A21/I456/o0/H5/S1/h14/s0/n18768776075/N0012012181444/i1

    [17/06-17:25:05.535] MGWConnThread sending: [Dialing /A21/I456/o0/H5/S1/h14/s0/n18768776075/N0012012181444/i1/#0]

    [17/06-17:25:05.541] [debug] ProcessLine received from hgs: [Dialing /A21/I456/o0/H5/S1/h14/s0/n18768776075/N0012012181444/i1/#0]

    [17/06-17:25:05.554] [debug] ProcessLine invoking Dialing with tid: 456 cid: 0 params: /A21/I456/o0/H5/S1/h14/s0/n18768776075/N0012012181444/i1/#0

    [17/06-17:25:05.555] [debug] CreateOutgoingSession CID: 0 TID: 456 cardAddres: 21 direction: 1 dstHW: 14 dstTS: 0 srcHW: 5 srcTS: 1

    [17/06-17:25:05.555] [debug] Init (/) start session active timer

    [17/06-17:25:05.556] [debug] ReserveLocalMediaResources (/456) m_MediaResourcesInUse Set 14.0

    [17/06-17:25:05.556] [debug] ReserveLocalMediaResources (/456) : LocalHwyTS 14:0 RemoteHwyTS 5:1

    [17/06-17:25:05.556] [debug] RegisterSession 1:0 -> /456

    [17/06-17:25:05.557] Received from MGW: [DialAck /A21/I456/x0,0/o1/#0]

    [17/06-17:25:05.558] Application:VoIP

    [17/06-17:25:05.559] Sending: [DialAck /A21/I456/x0,0/o1], Client ID:0

    [17/06-17:25:05.559] Real session ID [456]

    [17/06-17:25:05.602] [debug] MakeNetCall (/456) source: 0012012181444 destination: 18768776075

    [17/06-17:25:05.604] [debug] MakeNetCall (/456) best matching prefix for 18768776075 is 18768776075

    [17/06-17:25:05.605] [info] (/456) Making call to: 8768776075@192.168.1.13

    [17/06-17:25:05.605] [debug] GetLocalInfo (/456) GetLocalInfo received RTPAddr:192.168.0.3 wRTPPort:4000

    [17/06-17:25:05.606] [debug] SetState (/456) 1

    [17/06-17:25:05.606] [notice] call from 0012012181444 to 18768776075 dialing

    [17/06-17:25:05.607] [debug] makeCall MakeCall from: 0012012181444 to: 8768776075@192.168.1.13

    [17/06-17:25:05.608] [debug] ChangeContactAddress SetDefaultFrom report IP: 192.168.1.12

    [17/06-17:25:05.609] [debug] createSdpContents FindMediaReportIP(192.168.1.13)=192.168.1.12

    [17/06-17:25:05.609] [debug] createSdpContents addCodec(G723,8000)

    [17/06-17:25:05.610] [debug] createSdpContents addCodec(G729,8000)

    [17/06-17:25:05.610] [debug] createSdpContents addCodec(G729a,8000)

    [17/06-17:25:05.611] [debug] createSdpContents addCodec(G729b,8000)

    [17/06-17:25:05.611] [debug] createSdpContents addCodec(G729ab,8000)

    [17/06-17:25:05.612] [debug] createSdpContents addCodec(PCMU,8000)

    [17/06-17:25:05.619] [debug] createSdpContents addCodec(PCMA,8000)

    [17/06-17:25:05.619] [debug] createSdpContents addCodec(telephone-event,8000)

    [17/06-17:25:05.620] [debug] RegisterTokenForSession 1:0 -> UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456

    [17/06-17:25:05.620] [debug] SetState Set session state to:eNotConnected

    [17/06-17:25:05.621] [debug] makeCall Via:

    [17/06-17:25:05.621] [debug] makeCall Via: 192.168.1.12

    [17/06-17:25:05.622] [debug] makeCall From = [sip:0012012181444@192.168.1.12:5060]

    [17/06-17:25:05.624] [debug] makeCall Sending INVITE [uAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.]

    [17/06-17:25:05.625] [debug] onTrying UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

    [17/06-17:25:05.626] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

    [17/06-17:25:05.627] [debug] onNewSession UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.remoteIP: 192.168.1.13

    [17/06-17:25:05.628] [debug] GetLocalInfo (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) GetLocalInfo received RTPAddr:192.168.0.3 wRTPPort:4000

    [17/06-17:25:05.629] Received from MGW: [sysAlerting /A21/I456/x0,0/o1/#0]

    [17/06-17:25:05.631] Application:VoIP

    [17/06-17:25:05.632] Sending: [sysAlerting /A21/I456/x0,0/o1], Client ID:0

    [17/06-17:25:05.633] Real session ID [456]

    [17/06-17:25:05.683] [debug] OnCreateExternalRTPHandler created external rtp handler board: 192.168.0.3 rtp port: 4000 for token: UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

    [17/06-17:25:05.683] [debug] onProvisional UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

    [17/06-17:25:05.684] [debug] onEarlyMedia UAC - Starting media.

    [17/06-17:25:05.685] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

    [17/06-17:25:05.686] [debug] updateRemoteMediaAddr Updated remote media address to: 192.168.1.13

    [17/06-17:25:05.687] [debug] OnStartExternalRTPHandler start external rtp handler for token: UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

    [17/06-17:25:05.688] [debug] mediaAllocResources Creating channel with payload 0

    [17/06-17:25:05.689] [debug] setChannelparam setting channel parameters board: 0 bus: 1 timeslot: 0 payload: 0 mute DTMF: 0

    [17/06-17:25:05.690] [debug] setChannelparam using SIP/RFC2833, payload: 101

    [17/06-17:25:05.691] [debug] exOpenChannel Setting mapping: Channel=0 -> HW:TS=1.0

    [17/06-17:25:05.693] [debug] exOpenChannel Created channel 0

    [17/06-17:25:05.694] [debug] mediaAllocResources OK mediaAllocResources Bus#1 Bus#0 PayLoad=0

    [17/06-17:25:05.695] [debug] ~stopwatch mediaAllocResources: 690 usec

    [17/06-17:25:05.696] [debug] ~stopwatch openChannel: 742 usec

    [17/06-17:25:05.697] [debug] CreateChannel (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) created channel timeslot: 0 handle: 0

    [17/06-17:25:05.698] [debug] StartMediaStream (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) localMediaAddr:192.168.0.3 localMediaPort:4000 remoteMediaAddr:192.168.1.13 remoteMediaPort:53830 Payload:0

    [17/06-17:25:05.699] [debug] mediaActivateRTP_RTCPChannel acActivateRTP_RTCPChannel( IPPrec=0, nTOS=0, tx=0,rx=0,ChannelHandle 0 ) returned 0

    [17/06-17:25:05.700] [debug] ~stopwatch mediaActivateRTP_RTCPChannel: 276 usec

    [17/06-17:25:05.701] [debug] startMedia Started media, accepting call [uAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.]

    [17/06-17:25:05.714] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:05.725] [debug] onReadyToSend UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

    [17/06-17:25:05.726] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

    [17/06-17:25:05.726] [debug] onAnswer UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

    [17/06-17:25:05.727] Received from MGW: [Answering /A21/I456/x0,0/o1/#0]

    [17/06-17:25:05.728] Application:VoIP

    [17/06-17:25:05.729] Sending: [Answering /A21/I456/x0,0/o1], Client ID:0

    [17/06-17:25:05.730] Real session ID [456]

    [17/06-17:25:05.773] [debug] SetState (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) 2

    [17/06-17:25:05.774] [debug] StartConnectionTimer (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) Call from PBX, setting keepalive timer to 90 seconds

    [17/06-17:25:05.775] [debug] StartConnectionTimer (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) Started connection timer

    [17/06-17:25:05.776] [debug] onConnected UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

    [17/06-17:25:05.777] [debug] SetState Set session state to:eConnected

    [17/06-17:25:06.193] * Received Packet: ConnectAck /A21/I456/o0

    [17/06-17:25:06.194] updateReplyContext: no effect

    [17/06-17:25:06.194] Application:VoIP

    [17/06-17:25:06.195] MGWConnThread sending: [ConnectAck /A21/I456/o0/#0]

    [17/06-17:25:06.198] [debug] ProcessLine received from hgs: [ConnectAck /A21/I456/o0/#0]

    [17/06-17:25:06.199] [debug] ProcessLine invoking ConnectAck with tid: 456 cid: 0 params: /A21/I456/o0/#0

    [17/06-17:25:06.393] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:09.833] Sending: [ConnectionPing /AMG/I26/S1], Client ID:8

    [17/06-17:25:09.838] MGWConnThread sending: [ConnectionPing /AMG/I26/S1/#8]

    [17/06-17:25:09.841] [debug] ProcessLine received from hgs: [ConnectionPing /AMG/I26/S1/#8]

    [17/06-17:25:09.843] [debug] ProcessLine invoking ConnectionPing with tid: 26 cid: 8 params: /AMG/I26/S1/#8

    [17/06-17:25:09.844] Received from MGW: [ConnectionPong /I26/#8]

    [17/06-17:25:10.723] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:11.393] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:13.813] * Received Packet: FaultyChannelsInfo /A24/I1204/H5/h14/r31=12,32=13,33=14,34=15

    [17/06-17:25:13.814] updateReplyContext: no effect

    [17/06-17:25:13.816] Sending: [FaultyChannels /AMG/I87/r12,13,14,15], Client ID:8

    [17/06-17:25:13.817] MGWConnThread sending: [FaultyChannels /AMG/I87/r12,13,14,15/#8]

    [17/06-17:25:13.823] [debug] ProcessLine received from hgs: [FaultyChannels /AMG/I87/r12,13,14,15/#8]

    [17/06-17:25:13.824] [debug] ProcessLine invoking FaultyChannels with tid: 87 cid: 8 params: /AMG/I87/r12,13,14,15/#8

    [17/06-17:25:15.723] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:16.393] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:17.614] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 2 NumDigits: 1 HW: 1 TS: 0

    [17/06-17:25:17.615] [debug] OnDTMF OnDTMF notification: Digit=2, nHW=14, nTS.Type=0.1

    [17/06-17:25:17.963] [debug] exProceedEvent AudioCodes event: EV_BROKEN_RTP_CONNECTION

    [17/06-17:25:18.333] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 0 NumDigits: 1 HW: 1 TS: 0

    [17/06-17:25:18.335] [debug] OnDTMF OnDTMF notification: Digit=0, nHW=14, nTS.Type=0.1

    [17/06-17:25:19.054] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 1 NumDigits: 1 HW: 1 TS: 0

    [17/06-17:25:19.055] [debug] OnDTMF OnDTMF notification: Digit=1, nHW=14, nTS.Type=0.1

    [17/06-17:25:19.773] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 2 NumDigits: 1 HW: 1 TS: 0

    [17/06-17:25:19.775] [debug] OnDTMF OnDTMF notification: Digit=2, nHW=14, nTS.Type=0.1

    [17/06-17:25:20.484] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 1 NumDigits: 1 HW: 1 TS: 0

    [17/06-17:25:20.485] [debug] OnDTMF OnDTMF notification: Digit=1, nHW=14, nTS.Type=0.1

    [17/06-17:25:20.724] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:21.264] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 8 NumDigits: 1 HW: 1 TS: 0

    [17/06-17:25:21.265] [debug] OnDTMF OnDTMF notification: Digit=8, nHW=14, nTS.Type=0.1

    [17/06-17:25:21.393] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:21.934] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 1 NumDigits: 1 HW: 1 TS: 0

    [17/06-17:25:21.935] [debug] OnDTMF OnDTMF notification: Digit=1, nHW=14, nTS.Type=0.1

    [17/06-17:25:22.634] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 4 NumDigits: 1 HW: 1 TS: 0

    [17/06-17:25:22.635] [debug] OnDTMF OnDTMF notification: Digit=4, nHW=14, nTS.Type=0.1

    [17/06-17:25:23.324] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 1 NumDigits: 1 HW: 1 TS: 0

    [17/06-17:25:23.327] [debug] OnDTMF OnDTMF notification: Digit=1, nHW=14, nTS.Type=0.1

    [17/06-17:25:24.043] * Received Packet: GenericReply /#90/@2b/x0,1/I3033/G

    [17/06-17:25:24.074] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 1 NumDigits: 1 HW: 1 TS: 0

    [17/06-17:25:24.075] [debug] OnDTMF OnDTMF notification: Digit=1, nHW=14, nTS.Type=0.1

    [17/06-17:25:24.204] * Received Packet: SystemTimeEvent /A2b/x0,1/I3034/g11,6,16,5,17,15,56

    [17/06-17:25:24.205] Ignoring system time event

    [17/06-17:25:24.233] [debug] exProceedEvent AudioCodes event: EV_CONNECTION_ESTABLISHED

    [17/06-17:25:25.714] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:26.394] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:28.364] [debug] exProceedEvent AudioCodes event: EV_BROKEN_RTP_CONNECTION

    [17/06-17:25:30.004] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 1 NumDigits: 1 HW: 1 TS: 0

    [17/06-17:25:30.005] [debug] OnDTMF OnDTMF notification: Digit=1, nHW=14, nTS.Type=0.1

    [17/06-17:25:30.724] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:30.725] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 2 NumDigits: 1 HW: 1 TS: 0

    [17/06-17:25:30.725] [debug] OnDTMF OnDTMF notification: Digit=2, nHW=14, nTS.Type=0.1

    [17/06-17:25:31.394] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:31.464] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 3 NumDigits: 1 HW: 1 TS: 0

    [17/06-17:25:31.465] [debug] OnDTMF OnDTMF notification: Digit=3, nHW=14, nTS.Type=0.1

    [17/06-17:25:31.695] HMCServer received [ping] from 127.0.0.1:40150

    [17/06-17:25:31.695] processRequestLines: reply: "pong " for client: 13

    [17/06-17:25:32.174] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 4 NumDigits: 1 HW: 1 TS: 0

    [17/06-17:25:32.175] [debug] OnDTMF OnDTMF notification: Digit=4, nHW=14, nTS.Type=0.1

    [17/06-17:25:32.534] [debug] exProceedEvent AudioCodes event: EV_CONNECTION_ESTABLISHED

    [17/06-17:25:35.338] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

    [17/06-17:25:35.339] [debug] onNewSession UAS:7e61c163@pbxremoteIP: 192.168.1.13

    [17/06-17:25:35.340] [debug] SetState Set session state to:eNotConnected

    [17/06-17:25:35.340] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

    [17/06-17:25:35.340] [debug] updateRemoteMediaAddr Updated remote media address to: 192.168.1.13

    [17/06-17:25:35.341] [debug] onNewSession UAS:7e61c163@pbxm_TokenToSession.insert

    [17/06-17:25:35.342] [debug] CreateIncomingSession token: UAS:7e61c163@pbx remoteNumber: 0012012181444 localNumber: 13109644430 remoteIP: 192.168.1.13

    [17/06-17:25:35.342] [debug] Init (/) start session active timer

    [17/06-17:25:35.343] [debug] IsCallAllowed Number is not in BlockedDDIs list

    [17/06-17:25:35.344] [debug] IsCallAllowed source DDI:13109644430 allowed DDI:^*

    [17/06-17:25:35.345] [debug] FindLocalMediaResources (UAS:7e61c163@pbx/) LocalHwyTS: 14:1 RemoteHwyTS: 5:2 media resource: 14.1

    [17/06-17:25:35.345] [debug] RegisterSession 1:1 -> UAS:7e61c163@pbx/

    [17/06-17:25:35.346] Received from MGW: [Dialing /A21/x0,0/I2/o0/N"0012012181444"/H14/S1/h5/s2/n13109644430/#9]

    [17/06-17:25:35.347] packStr=/A21/x0,0/I2/o0/N"0012012181444"/H14/S1/h5/s2/n13109644430

    [17/06-17:25:35.347] strHW=14

    [17/06-17:25:35.348] strTS=1

    [17/06-17:25:35.348] no in filter for add9

    [17/06-17:25:35.348] getApplication for:MG.2-VoIP

    [17/06-17:25:35.349] strCard=[MG], nClientID=9

    [17/06-17:25:35.349] From VoIP Trigger:Dialing /A21/x0,0/I2/o0/N"0012012181444"/H14/S1/h5/s2/n13109644430

    [17/06-17:25:35.350] Sending: [Dialing /A21/x0,0/I2/o0/N"0012012181444"/H14/S1/h5/s2/n13109644430], Client ID:9

    [17/06-17:25:35.350] No filters required for out address 21.2

    [17/06-17:25:35.351] After dial filter: [Dialing /A21/x0,0/I2/o0/N"0012012181444"/H14/S1/h5/s2/n13109644430]

    [17/06-17:25:35.351] Real session ID [2]

    [17/06-17:25:35.393] [debug] SetState (UAS:7e61c163@pbx/2) 1

    [17/06-17:25:35.394] [notice] call from 192.168.1.13 to 13109644430 dialing

    [17/06-17:25:35.395] [debug] GetLocalInfo (UAS:7e61c163@pbx/2) GetLocalInfo received RTPAddr:192.168.0.3 wRTPPort:4010

    [17/06-17:25:35.396] [debug] OnCreateExternalRTPHandler created external rtp handler board: 192.168.0.3 rtp port: 4010 for token: UAS:7e61c163@pbx

    [17/06-17:25:35.397] [debug] onNewSession UAS:UAS:7e61c163@pbx Early Media - Send 183

    [17/06-17:25:35.398] [debug] onOffer UAS:7e61c163@pbx

    [17/06-17:25:35.399] [debug] provideOkWithSDP UAS: UAS:7e61c163@pbx

    [17/06-17:25:35.400] [debug] provideOkWithSDP UAS:offered media:audio|RTP/AVP|0

    [17/06-17:25:35.401] [debug] provideOkWithSDP UAS:findFirstMatchingCodecs for: Name=pcmu, payload=0

    [17/06-17:25:35.402] [debug] provideOkWithSDP UAS:get strRemoteMediaAddr from msg contents, addr=192.168.1.13

    [17/06-17:25:35.402] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

    [17/06-17:25:35.404] [debug] updateRemoteMediaAddr Updated remote media address to: 192.168.1.13

    [17/06-17:25:35.405] [debug] createSdpContents FindMediaReportIP(192.168.1.13)=192.168.1.12

    [17/06-17:25:35.406] [debug] createSdpContents addCodec(pcmu,8000)

    [17/06-17:25:35.409] [debug] createSdpContents addCodec(telephone-event,8000)

    [17/06-17:25:35.410] [debug] provideOkWithSDP provideAnswer

    [17/06-17:25:35.411] [debug] onOffer UAS:onOffer - Early media state - send 183. token=UAS:7e61c163@pbx

    [17/06-17:25:35.411] [debug] onReadyToSend UAS:7e61c163@pbx

    [17/06-17:25:35.412] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

    [17/06-17:25:35.414] [debug] OnStartExternalRTPHandler start external rtp handler for token: UAS:7e61c163@pbx

    [17/06-17:25:35.416] [debug] mediaAllocResources Creating channel with payload 0

    [17/06-17:25:35.417] [debug] setChannelparam setting channel parameters board: 0 bus: 1 timeslot: 1 payload: 0 mute DTMF: 0

    [17/06-17:25:35.418] [debug] setChannelparam using SIP/RFC2833, payload: 101

    [17/06-17:25:35.419] [debug] exOpenChannel Setting mapping: Channel=1 -> HW:TS=1.1

    [17/06-17:25:35.432] [debug] exOpenChannel Created channel 1

    [17/06-17:25:35.433] [debug] mediaAllocResources OK mediaAllocResources Bus#1 Bus#1 PayLoad=0

    [17/06-17:25:35.438] [debug] ~stopwatch mediaAllocResources: 719 usec

    [17/06-17:25:35.439] [debug] ~stopwatch openChannel: 772 usec

    [17/06-17:25:35.449] [debug] CreateChannel (UAS:7e61c163@pbx/2) created channel timeslot: 1 handle: 1

    [17/06-17:25:35.450] [debug] StartMediaStream (UAS:7e61c163@pbx/2) localMediaAddr:192.168.0.3 localMediaPort:4010 remoteMediaAddr:192.168.1.13 remoteMediaPort:48640 Payload:0

    [17/06-17:25:35.451] [debug] mediaActivateRTP_RTCPChannel acActivateRTP_RTCPChannel( IPPrec=0, nTOS=0, tx=0,rx=0,ChannelHandle 1 ) returned 0

    [17/06-17:25:35.452] [debug] ~stopwatch mediaActivateRTP_RTCPChannel: 288 usec

    [17/06-17:25:35.453] [debug] onOffer UAS:Started media, accepting call [uAS:7e61c163@pbx]

    [17/06-17:25:35.454] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:35.484] [debug] exProceedEvent AudioCodes event: EV_BROKEN_RTP_CONNECTION

    [17/06-17:25:35.514] * Received Packet: DialAck /A21/I2/o1/i1

    [17/06-17:25:35.514] updateReplyContext: no effect

    [17/06-17:25:35.515] ClientIdMGWSend strMsg=DialAck /A21/I2/o1/i1

    [17/06-17:25:35.515] Reply for MGW:DialAck /A21/I2/o1/i1

    [17/06-17:25:35.516] MGWConnThread sending: [DialAck /A21/I2/o1/i1/#9]

    [17/06-17:25:35.518] [debug] ProcessLine received from hgs: [DialAck /A21/I2/o1/i1/#9]

    [17/06-17:25:35.554] [debug] ProcessLine invoking DialAck with tid: 2 cid: 9 params: /A21/I2/o1/i1/#9

    [17/06-17:25:35.684] [debug] exProceedEvent AudioCodes event: EV_BROKEN_RTP_CONNECTION

    [17/06-17:25:35.724] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:36.394] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:36.395] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:40.434] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:40.724] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:40.725] [debug] exProceedEvent AudioCodes event: EV_CONNECTION_ESTABLISHED

    [17/06-17:25:41.394] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:41.395] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:42.414] * Received Packet: Answering /A21/I2/o1

    [17/06-17:25:42.415] updateReplyContext: no effect

    [17/06-17:25:42.416] ClientIdMGWSend strMsg=Answering /A21/I2/o1

    [17/06-17:25:42.417] Reply for MGW:Answering /A21/I2/o1

    [17/06-17:25:42.417] MGWConnThread sending: [Answering /A21/I2/o1/#9]

    [17/06-17:25:42.422] [debug] ProcessLine received from hgs: [Answering /A21/I2/o1/#9]

    [17/06-17:25:42.423] [debug] ProcessLine invoking Answering with tid: 2 cid: 9 params: /A21/I2/o1/#9

    [17/06-17:25:42.425] Received from MGW: [ConnectAck /A21/I2/x0,0/o0/#9]

    [17/06-17:25:42.427] ClientIdMGWSend strMsg=ConnectAck /A21/I2/x0,0/o0

    [17/06-17:25:42.428] Sending: [ConnectAck /A21/I2/x0,0/o0], Client ID:9

    [17/06-17:25:42.429] Real session ID [2]

    [17/06-17:25:42.473] [debug] SetState (UAS:7e61c163@pbx/2) 2

    [17/06-17:25:42.474] [debug] StartConnectionTimer (UAS:7e61c163@pbx/2) Call from Net, setting keepalive timer to 60 seconds

    [17/06-17:25:42.475] [debug] StartConnectionTimer (UAS:7e61c163@pbx/2) Started connection timer

    [17/06-17:25:42.476] [debug] sendAnswering UAS sendAnswering (UAS:7e61c163@pbx)

    [17/06-17:25:42.477] [debug] sendAnswering Sending accept

    [17/06-17:25:42.478] [debug] onReadyToSend UAS:7e61c163@pbx

    [17/06-17:25:42.479] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

    [17/06-17:25:42.480] [debug] onConnected UAS:7e61c163@pbx

    [17/06-17:25:42.481] [debug] SetState Set session state to:eConnected

    [17/06-17:25:42.494] [debug] exProceedEvent AudioCodes event: EV_CONNECTION_ESTABLISHED

    [17/06-17:25:42.925] [debug] onReadyToSend UAS:7e61c163@pbx

    [17/06-17:25:42.926] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

    [17/06-17:25:43.914] * Received Packet: GenericReply /#90/@2b/x0,1/I3035/G

    [17/06-17:25:43.935] [debug] onReadyToSend UAS:7e61c163@pbx

    [17/06-17:25:43.936] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

    [17/06-17:25:45.434] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:45.724] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:45.945] [debug] onReadyToSend UAS:7e61c163@pbx

    [17/06-17:25:45.946] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

    [17/06-17:25:46.395] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:46.396] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:49.955] [debug] onReadyToSend UAS:7e61c163@pbx

    [17/06-17:25:49.956] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

    [17/06-17:25:50.434] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:50.715] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:51.394] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:51.396] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:52.424] [debug] exProceedEvent AudioCodes event: EV_BROKEN_RTP_CONNECTION

    [17/06-17:25:53.635] [debug] exProceedEvent EV_DIGIT: handle: 1 Digit: 3 NumDigits: 1 HW: 1 TS: 1

    [17/06-17:25:53.636] [debug] OnDTMF OnDTMF notification: Digit=3, nHW=14, nTS.Type=1.1

    [17/06-17:25:53.944] [debug] exProceedEvent EV_DIGIT: handle: 1 Digit: 4 NumDigits: 1 HW: 1 TS: 1

    [17/06-17:25:53.946] [debug] OnDTMF OnDTMF notification: Digit=4, nHW=14, nTS.Type=1.1

    [17/06-17:25:53.965] [debug] onReadyToSend UAS:7e61c163@pbx

    [17/06-17:25:53.966] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

    [17/06-17:25:55.434] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:55.724] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:56.395] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:56.396] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:25:57.975] [debug] onReadyToSend UAS:7e61c163@pbx

    [17/06-17:25:57.976] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

    [17/06-17:25:59.854] Sending: [ConnectionPing /AMG/I27/S1], Client ID:8

    [17/06-17:25:59.855] MGWConnThread sending: [ConnectionPing /AMG/I27/S1/#8]

    [17/06-17:25:59.858] [debug] ProcessLine received from hgs: [ConnectionPing /AMG/I27/S1/#8]

    [17/06-17:25:59.858] [debug] ProcessLine invoking ConnectionPing with tid: 27 cid: 8 params: /AMG/I27/S1/#8

    [17/06-17:25:59.859] Received from MGW: [ConnectionPong /I27/#8]

    [17/06-17:26:00.435] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:26:00.435] [debug] exProceedEvent AudioCodes event: EV_CONNECTION_ESTABLISHED

    [17/06-17:26:00.725] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:26:01.395] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:26:01.396] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:26:01.685] HMCServer received [ping] from 127.0.0.1:40150

    [17/06-17:26:01.686] processRequestLines: reply: "pong " for client: 13

    [17/06-17:26:01.985] [debug] onReadyToSend UAS:7e61c163@pbx

    [17/06-17:26:01.986] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

    [17/06-17:26:03.734] * Received Packet: GenericReply /#90/@2b/x0,1/I3036/G

    [17/06-17:26:05.094] * Received Packet: FaultyChannelsInfo /A25/I1414/H5/h14/r41=16,42=17,43=18,44=19

    [17/06-17:26:05.095] updateReplyContext: no effect

    [17/06-17:26:05.096] Sending: [FaultyChannels /AMG/I87/r16,17,18,19], Client ID:8

    [17/06-17:26:05.098] MGWConnThread sending: [FaultyChannels /AMG/I87/r16,17,18,19/#8]

    [17/06-17:26:05.102] [debug] ProcessLine received from hgs: [FaultyChannels /AMG/I87/r16,17,18,19/#8]

    [17/06-17:26:05.103] [debug] ProcessLine invoking FaultyChannels with tid: 87 cid: 8 params: /AMG/I87/r16,17,18,19/#8

    [17/06-17:26:05.434] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:26:05.545] [debug] OnSessionActiveTimeout (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) state: 2

    [17/06-17:26:05.546] [info] (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) timeslot: 0 handle: 0 call duration: 59 seconds

    [17/06-17:26:05.725] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:26:05.825] * Received Packet: FaultyChannelsInfo /A23/I909/H5/h14/r21=8,22=9,23=10,24=11

    [17/06-17:26:05.827] updateReplyContext: no effect

    [17/06-17:26:05.829] Sending: [FaultyChannels /AMG/I87/r8,9,10,11], Client ID:8

    [17/06-17:26:05.830] MGWConnThread sending: [FaultyChannels /AMG/I87/r8,9,10,11/#8]

    [17/06-17:26:05.833] [debug] ProcessLine received from hgs: [FaultyChannels /AMG/I87/r8,9,10,11/#8]

    [17/06-17:26:05.834] [debug] ProcessLine invoking FaultyChannels with tid: 87 cid: 8 params: /AMG/I87/r8,9,10,11/#8

    [17/06-17:26:05.995] [debug] onReadyToSend UAS:7e61c163@pbx

    [17/06-17:26:05.996] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

    [17/06-17:26:06.395] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:26:06.396] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:26:07.774] * Received Packet: KeepAlive /A21/I456/o0

    [17/06-17:26:07.775] updateReplyContext: no effect

    [17/06-17:26:07.776] Application:VoIP

    [17/06-17:26:07.777] MGWConnThread sending: [KeepAlive /A21/I456/o0/#0]

    [17/06-17:26:07.780] [debug] ProcessLine received from hgs: [KeepAlive /A21/I456/o0/#0]

    [17/06-17:26:07.782] [debug] ProcessLine invoking KeepAlive with tid: 456 cid: 0 params: /A21/I456/o0/#0

    [17/06-17:26:07.783] Received from MGW: [KeepAlive /A21/I456/x0,0/o1/#0]

    [17/06-17:26:07.784] Application:VoIP

    [17/06-17:26:07.785] Sending: [KeepAlive /A21/I456/x0,0/o1], Client ID:0

    [17/06-17:26:07.785] Real session ID [456]

    [17/06-17:26:10.005] [debug] onReadyToSend UAS:7e61c163@pbx

    [17/06-17:26:10.006] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

    [17/06-17:26:10.435] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:26:10.725] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:26:11.395] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:26:11.395] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

    [17/06-17:26:13.675] * Received Packet: HangingUp /A21/I456/o0/R10

    [17/06-17:26:13.675] updateReplyContext: no effect

    [17/06-17:26:13.676] Application:VoIP

    [17/06-17:26:13.677] ID2App removing ID:456

    [17/06-17:26:13.678] MGWConnThread sending: [HangingUp /A21/I456/o0/R10/#0]

    [17/06-17:26:13.682] [debug] ProcessLine received from hgs: [HangingUp /A21/I456/o0/R10/#0]

    [17/06-17:26:13.683] [debug] ProcessLine invoking HangingUp with tid: 456 cid: 0 params: /A21/I456/o0/R10/#0

    [17/06-17:26:13.683] [debug] Hangup (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456)

    [17/06-17:26:13.684] [notice] call from 0012012181444 to 8768776075 hangup

    [17/06-17:26:13.685] [debug] CloseAudioChannel (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) deactivate rtp channel: 0

    [17/06-17:26:13.685] [debug] ~stopwatch mediaDeactivateRTP_RTCPChannel: 177 usec

    [17/06-17:26:13.686] [debug] CloseAudioChannel (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) closing channel timeslot: 0 handle: 0

    [17/06-17:26:13.686] [debug] mediaCloseResources Closing channel 0

    [17/06-17:26:13.687] [debug] ~stopwatch mediaCloseResources: 173 usec

    [17/06-17:26:13.687] [debug] ~stopwatch closeChannel: 282 usec

    [17/06-17:26:13.688] [debug] Hangup (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) waiting for call statistics...

    [17/06-17:26:13.697] [debug] exProceedEvent received acEV_RTCP_CLOSE

    [17/06-17:26:13.698] [debug] OnCloseRTP (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456)

    [17/06-17:26:13.699] [debug] UnregisterSession 1:0 -> UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456

    [17/06-17:26:13.699] [debug] Close (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456)

    [17/06-17:26:13.700] [debug] Close (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) closing session timeslot:0 handle: -1

    [17/06-17:26:13.701] [debug] CloseResources (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456)

    [17/06-17:26:13.707] [debug] CloseAudioChannel (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) channel timeslot: 0 already closed

    [17/06-17:26:13.708] [debug] CloseResources (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) remove media resoruce: 14.0

    [17/06-17:26:13.709] [debug] DisconnectEndpoints (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456)

    [17/06-17:26:13.710] [debug] ClearNetCall (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) Cause Code: 10 Converted Cause Code: 10

    [17/06-17:26:13.711] [debug] UpdateStatsAndCDR (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456)

    [17/06-17:26:13.712] [info] VoIP CDR: 4,2011-06-17 T 17:25:05,2011-06-17 T 17:25:05,0012012181444,8768776075,2011-06-17 T 17:25:05,2011-06-17 T 17:26:13,67,192.168.1.13,192.168.1.13,53830,1,0,1,1,3403,935,0,10

    [17/06-17:26:13.713] [debug] ~tMGWSession (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456)

    [17/06-17:26:13.715] [debug] clearCall ClearCall [uAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.]

    [17/06-17:26:13.716] [debug] setHangupReason m_nHangupQ931: 10 m_nHangupSIP: 480

    [17/06-17:26:13.716] [debug] endCall Q931 reason: 10 SIP reason: 480

    [17/06-17:26:13.717] [debug] onReadyToSend UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

    [17/06-17:26:13.718] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

    [17/06-17:26:13.719] [debug] onTerminated UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.] reason[1]

    [17/06-17:26:13.737] [debug] callTeardown telephony Disconnected

    [17/06-17:26:13.740] [debug] removeSession Call Id [uAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.]: m_tokenToSession.erase

    [17/06-17:26:13.741] [debug] ~tSessionInfo Call id [uAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.]: Calling ~tSessionInfo

    [17/06-17:26:14.016] [debug] onReadyToSend UAS:7e61c163@pbx

    [17/06-17:26:14.017] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

    [17/06-17:26:14.426] [debug] onAckNotReceived UAS:

    [17/06-17:26:14.428] [debug] callTeardown network Disconnected

    [17/06-17:26:14.429] [debug] OnRemoteNetDisconnect reason: 10 for token: UAS:7e61c163@pbx

    [17/06-17:26:14.429] [debug] Hangup (UAS:7e61c163@pbx/2)

    [17/06-17:26:14.430] [notice] call from 0012012181444 to 13109644430 hangup

    [17/06-17:26:14.431] [debug] CloseAudioChannel (UAS:7e61c163@pbx/2) deactivate rtp channel: 1

    [17/06-17:26:14.432] [debug] ~stopwatch mediaDeactivateRTP_RTCPChannel: 181 usec

    [17/06-17:26:14.433] [debug] CloseAudioChannel (UAS:7e61c163@pbx/2) closing channel timeslot: 1 handle: 1

    [17/06-17:26:14.433] [debug] mediaCloseResources Closing channel 1

    [17/06-17:26:14.437] [debug] ~stopwatch mediaCloseResources: 185 usec

    [17/06-17:26:14.438] [debug] ~stopwatch closeChannel: 293 usec

    [17/06-17:26:14.439] [debug] Hangup (UAS:7e61c163@pbx/2) waiting for call statistics...

    [17/06-17:26:14.440] [debug] removeSession Call Id [uAS:7e61c163@pbx]: m_tokenToSession.erase

    [17/06-17:26:14.441] [debug] ~tSessionInfo Call id [uAS:7e61c163@pbx]: Calling ~tSessionInfo

    [17/06-17:26:14.443] [debug] exProceedEvent received acEV_RTCP_CLOSE

    [17/06-17:26:14.444] [debug] OnCloseRTP (UAS:7e61c163@pbx/2)

    [17/06-17:26:14.445] [debug] UnregisterSession 1:1 -> UAS:7e61c163@pbx/2

    [17/06-17:26:14.446] [debug] Close (UAS:7e61c163@pbx/2)

    [17/06-17:26:14.446] [debug] Close (UAS:7e61c163@pbx/2) closing session timeslot:1 handle: -1

    [17/06-17:26:14.447] [debug] CloseResources (UAS:7e61c163@pbx/2)

    [17/06-17:26:14.448] [debug] CloseAudioChannel (UAS:7e61c163@pbx/2) channel timeslot: 1 already closed

    [17/06-17:26:14.449] [debug] CloseResources (UAS:7e61c163@pbx/2) remove media resoruce: 14.1

    [17/06-17:26:14.450] [debug] DisconnectEndpoints (UAS:7e61c163@pbx/2)

    [17/06-17:26:14.451] [debug] SendHangupToCard Cause Code =10 (SendHangupToCard)

    [17/06-17:26:14.452] [debug] SendHangupToCard Converted Cause Code=10

    [17/06-17:26:14.453] Received from MGW: [HangingUp /A21/I2/x0,0/o0/R10/#9]

    [17/06-17:26:14.455] ClientIdMGWSend strMsg=HangingUp /A21/I2/x0,0/o0/R10

    [17/06-17:26:14.456] Sending: [HangingUp /A21/I2/x0,0/o0/R10], Client ID:9

    [17/06-17:26:14.457] Real session ID [2]

    [17/06-17:26:14.504] [debug] UpdateStatsAndCDR (UAS:7e61c163@pbx/2)

    [17/06-17:26:14.505] [info] VoIP CDR: 5,2011-06-17 T 17:25:35,2011-06-17 T 17:25:35,0012012181444,13109644430,2011-06-17 T 17:25:42,2011-06-17 T 17:26:14,32,192.168.1.13,192.168.1.13,48640,2,0,0,0,1954,484,0,10

    [17/06-17:26:14.506] [debug] ~tMGWSession (UAS:7e61c163@pbx/2)

    [17/06-17:26:14.775] * Received Packet: ClearAck /A21/I2/o1

    [17/06-17:26:14.775] updateReplyContext: no effect

    [17/06-17:26:14.776] ClientIdMGWSend strMsg=ClearAck /A21/I2/o1

    [17/06-17:26:14.776] ID2App removing ID:2

    [17/06-17:26:14.777] Reply for MGW:ClearAck /A21/I2/o1

    [17/06-17:26:14.777] MGWConnThread sending: [ClearAck /A21/I2/o1/#9]

    [17/06-17:26:14.780] [debug] ProcessLine received from hgs: [ClearAck /A21/I2/o1/#9]

  5. Supervised Transfer Not working

    in Microsoft Speech Server

    Posted

    yes HG 4000 Supports both UDP and TCP. There is an option to set the protocol. If I set it to UDP, the incoming call in the first leg does not work. PBX just drops it. I dont even see it in the logs (even if I specify UDP on the trunk as transport). The moment I change the transport to TCP on HG 4000 , PBX sees the call and it works.

     

    Is there any way We can force PBX to send TCP back to HG 4000?

     

    Thanks,

     

    Sanjeev.

     

    I again tried to set HG4000 Transport = UDP. I also changed the Hypermedia Trunk in PBX to

    sip:192.168.1.12:5060;transport=udp

     

    The Call does not connect when I call. It look slike HG4000 i strying to make a call. Please see HG4000 Log below. But I see nothing in PBX (192.68.1.13). Should I configure something differetnly in PBX for UDP to work? I will also post a log of HG 4000 where the incoming and outgoing call are working but transfer doe not take place.

     

    HG 4000 Log when Transport = UDP

     

     

    [17/06-16:54:52.820] [debug] MakeNetCall (/452) source: 0012012181444 destination: 18768776075

    [17/06-16:54:52.821] [debug] MakeNetCall (/452) best matching prefix for 18768776075 is 18768776075

    [17/06-16:54:52.821] [info] (/452) Making call to: 8768776075@192.168.1.13

    [17/06-16:54:52.822] [debug] GetLocalInfo (/452) GetLocalInfo received RTPAddr:192.168.0.3 wRTPPort:4000

    [17/06-16:54:52.822] [debug] SetState (/452) 1

    [17/06-16:54:52.823] [notice] call from 0012012181444 to 18768776075 dialing

    [17/06-16:54:52.824] [debug] makeCall MakeCall from: 0012012181444 to: 8768776075@192.168.1.13

    [17/06-16:54:52.825] [debug] ChangeContactAddress SetDefaultFrom report IP: 192.168.1.12

    [17/06-16:54:52.825] [debug] createSdpContents FindMediaReportIP(192.168.1.13)=192.168.1.12

    [17/06-16:54:52.826] [debug] createSdpContents addCodec(G723,8000)

    [17/06-16:54:52.826] [debug] createSdpContents addCodec(G729,8000)

    [17/06-16:54:52.827] [debug] createSdpContents addCodec(G729a,8000)

    [17/06-16:54:52.827] [debug] createSdpContents addCodec(G729b,8000)

    [17/06-16:54:52.827] [debug] createSdpContents addCodec(G729ab,8000)

    [17/06-16:54:52.831] [debug] createSdpContents addCodec(PCMU,8000)

    [17/06-16:54:52.832] [debug] createSdpContents addCodec(PCMA,8000)

    [17/06-16:54:52.833] [debug] createSdpContents addCodec(telephone-event,8000)

    [17/06-16:54:52.834] [debug] RegisterTokenForSession 1:0 -> UAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ./452

    [17/06-16:54:52.835] [debug] SetState Set session state to:eNotConnected

    [17/06-16:54:52.836] [debug] makeCall Via:

    [17/06-16:54:52.837] [debug] makeCall Via: 192.168.1.12

    [17/06-16:54:52.838] [debug] makeCall From = [sip:0012012181444@192.168.1.12:5060]

    [17/06-16:54:52.839] [debug] makeCall Sending INVITE [uAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ.]

    [17/06-16:54:54.462] HMCServer received [MGWResStatus /I124/AMG] from 192.168.1.14:62261

    [17/06-16:54:54.463] Sending: [MGWResStatus /I124/AMG], Client ID:12

    [17/06-16:54:54.463] MGWConnThread sending: [MGWResStatus /I124/AMG/#12]

    [17/06-16:54:54.467] [debug] ProcessLine received from hgs: [MGWResStatus /I124/AMG/#12]

    [17/06-16:54:54.468] [debug] ProcessLine invoking MGWResStatus with tid: 124 cid: 12 params: /I124/AMG/#12

    [17/06-16:54:54.469] Received from MGW: [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected/#12]

    [17/06-16:54:54.471] m_server.sendToNetwork:MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected client:12 app:0

    [17/06-16:54:54.472] Adding message [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected] to eventQueue for Client 12 Application ID 0

    [17/06-16:54:54.473] HMCServer.signalSendChannels: sending to specific client

    [17/06-16:54:56.460] HMCServer received [MGWResStatus /I124/AMG] from 192.168.1.14:62261

    [17/06-16:54:56.461] Sending: [MGWResStatus /I124/AMG], Client ID:12

    [17/06-16:54:56.461] MGWConnThread sending: [MGWResStatus /I124/AMG/#12]

    [17/06-16:54:56.465] [debug] ProcessLine received from hgs: [MGWResStatus /I124/AMG/#12]

    [17/06-16:54:56.466] [debug] ProcessLine invoking MGWResStatus with tid: 124 cid: 12 params: /I124/AMG/#12

    [17/06-16:54:56.467] Received from MGW: [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected/#12]

    [17/06-16:54:56.469] m_server.sendToNetwork:MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected client:12 app:0

    [17/06-16:54:56.470] Adding message [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected] to eventQueue for Client 12 Application ID 0

    [17/06-16:54:56.471] HMCServer.signalSendChannels: sending to specific client

    [17/06-16:54:58.461] HMCServer received [MGWResStatus /I124/AMG] from 192.168.1.14:62261

    [17/06-16:54:58.462] Sending: [MGWResStatus /I124/AMG], Client ID:12

    [17/06-16:54:58.462] MGWConnThread sending: [MGWResStatus /I124/AMG/#12]

    [17/06-16:54:58.466] [debug] ProcessLine received from hgs: [MGWResStatus /I124/AMG/#12]

    [17/06-16:54:58.467] [debug] ProcessLine invoking MGWResStatus with tid: 124 cid: 12 params: /I124/AMG/#12

    [17/06-16:54:58.468] Received from MGW: [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected/#12]

    [17/06-16:54:58.470] m_server.sendToNetwork:MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected client:12 app:0

    [17/06-16:54:58.471] Adding message [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected] to eventQueue for Client 12 Application ID 0

    [17/06-16:54:58.472] HMCServer.signalSendChannels: sending to specific client

    [17/06-16:54:58.891] * Received Packet: GenericReply /#90/@2b/x0,1/I2912/G

    [17/06-16:55:00.471] HMCServer received [MGWResStatus /I124/AMG] from 192.168.1.14:62261

    [17/06-16:55:00.471] Sending: [MGWResStatus /I124/AMG], Client ID:12

    [17/06-16:55:00.472] MGWConnThread sending: [MGWResStatus /I124/AMG/#12]

    [17/06-16:55:00.476] [debug] ProcessLine received from hgs: [MGWResStatus /I124/AMG/#12]

    [17/06-16:55:00.477] [debug] ProcessLine invoking MGWResStatus with tid: 124 cid: 12 params: /I124/AMG/#12

    [17/06-16:55:00.478] Received from MGW: [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected/#12]

    [17/06-16:55:00.479] m_server.sendToNetwork:MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected client:12 app:0

    [17/06-16:55:00.481] Adding message [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected] to eventQueue for Client 12 Application ID 0

    [17/06-16:55:00.482] HMCServer.signalSendChannels: sending to specific client

    [17/06-16:55:02.462] HMCServer received [MGWResStatus /I124/AMG] from 192.168.1.14:62261

    [17/06-16:55:02.463] Sending: [MGWResStatus /I124/AMG], Client ID:12

    [17/06-16:55:02.463] MGWConnThread sending: [MGWResStatus /I124/AMG/#12]

    [17/06-16:55:02.467] [debug] ProcessLine received from hgs: [MGWResStatus /I124/AMG/#12]

    [17/06-16:55:02.468] [debug] ProcessLine invoking MGWResStatus with tid: 124 cid: 12 params: /I124/AMG/#12

    [17/06-16:55:02.469] Received from MGW: [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected/#12]

    [17/06-16:55:02.472] m_server.sendToNetwork:MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected client:12 app:0

    [17/06-16:55:02.473] Adding message [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected] to eventQueue for Client 12 Application ID 0

    [17/06-16:55:02.475] HMCServer.signalSendChannels: sending to specific client

    [17/06-16:55:04.460] HMCServer received [MGWResStatus /I124/AMG/S] from 192.168.1.14:62261

    [17/06-16:55:04.461] Sending: [MGWResStatus /I124/AMG/S], Client ID:12

    [17/06-16:55:04.462] MGWConnThread sending: [MGWResStatus /I124/AMG/S/#12]

    [17/06-16:55:04.466] [debug] ProcessLine received from hgs: [MGWResStatus /I124/AMG/S/#12]

    [17/06-16:55:04.467] [debug] ProcessLine invoking MGWResStatus with tid: 124 cid: 12 params: /I124/AMG/S/#12

    [17/06-16:55:04.468] Received from MGW: [MGWResStatusReply /I124/s0|0,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected/#12]

    [17/06-16:55:04.470] m_server.sendToNetwork:MGWResStatusReply /I124/s0|0,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected client:12 app:0

    [17/06-16:55:04.471] Adding message [MGWResStatusReply /I124/s0|0,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected] to eventQueue for Client 12 Application ID 0

    [17/06-16:55:04.473] HMCServer.signalSendChannels: sending to specific client

    [17/06-16:55:06.463] HMCServer received [MGWResStatus /I124/AMG] from 192.168.1.14:62261

    [17/06-16:55:06.464] Sending: [MGWResStatus /I124/AMG], Client ID:12

    [17/06-16:55:06.465] MGWConnThread sending: [MGWResStatus /I124/AMG/#12]

    [17/06-16:55:06.474] [debug] ProcessLine received from hgs: [MGWResStatus /I124/AMG/#12]

    [17/06-16:55:06.475] [debug] ProcessLine invoking MGWResStatus with tid: 124 cid: 12 params: /I124/AMG/#12

    [17/06-16:55:06.475] Received from MGW: [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected/#12]

    [17/06-16:55:06.477] m_server.sendToNetwork:MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected client:12 app:0

    [17/06-16:55:06.478] Adding message [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected] to eventQueue for Client 12 Application ID 0

    [17/06-16:55:06.479] HMCServer.signalSendChannels: sending to specific client

    [17/06-16:55:08.459] HMCServer received [MGWResStatus /I124/AMG] from 192.168.1.14:62261

    [17/06-16:55:08.460] Sending: [MGWResStatus /I124/AMG], Client ID:12

    [17/06-16:55:08.460] MGWConnThread sending: [MGWResStatus /I124/AMG/#12]

    [17/06-16:55:08.464] [debug] ProcessLine received from hgs: [MGWResStatus /I124/AMG/#12]

    [17/06-16:55:08.492] [debug] ProcessLine invoking MGWResStatus with tid: 124 cid: 12 params: /I124/AMG/#12

    [17/06-16:55:08.493] Received from MGW: [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected/#12]

    [17/06-16:55:08.494] m_server.sendToNetwork:MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected client:12 app:0

    [17/06-16:55:08.495] Adding message [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected] to eventQueue for Client 12 Application ID 0

    [17/06-16:55:08.496] HMCServer.signalSendChannels: sending to specific client

    [17/06-16:55:10.462] HMCServer received [MGWResStatus /I124/AMG] from 192.168.1.14:62261

    [17/06-16:55:10.463] Sending: [MGWResStatus /I124/AMG], Client ID:12

    [17/06-16:55:10.464] MGWConnThread sending: [MGWResStatus /I124/AMG/#12]

    [17/06-16:55:10.468] [debug] ProcessLine received from hgs: [MGWResStatus /I124/AMG/#12]

    [17/06-16:55:10.469] [debug] ProcessLine invoking MGWResStatus with tid: 124 cid: 12 params: /I124/AMG/#12

    [17/06-16:55:10.469] Received from MGW: [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected/#12]

    [17/06-16:55:10.470] m_server.sendToNetwork:MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected client:12 app:0

    [17/06-16:55:10.471] Adding message [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected] to eventQueue for Client 12 Application ID 0

    [17/06-16:55:10.472] HMCServer.signalSendChannels: sending to specific client

    [17/06-16:55:12.461] HMCServer received [MGWResStatus /I124/AMG] from 192.168.1.14:62261

    [17/06-16:55:12.462] Sending: [MGWResStatus /I124/AMG], Client ID:12

    [17/06-16:55:12.462] MGWConnThread sending: [MGWResStatus /I124/AMG/#12]

    [17/06-16:55:12.466] [debug] ProcessLine received from hgs: [MGWResStatus /I124/AMG/#12]

    [17/06-16:55:12.467] [debug] ProcessLine invoking MGWResStatus with tid: 124 cid: 12 params: /I124/AMG/#12

    [17/06-16:55:12.467] Received from MGW: [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected/#12]

    [17/06-16:55:12.468] m_server.sendToNetwork:MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected client:12 app:0

    [17/06-16:55:12.469] Adding message [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected] to eventQueue for Client 12 Application ID 0

    [17/06-16:55:12.471] HMCServer.signalSendChannels: sending to specific client

    [17/06-16:55:14.051] * Received Packet: HangingUp /A21/I452/o0/R10

    [17/06-16:55:14.052] updateReplyContext: no effect

    [17/06-16:55:14.053] Application:VoIP

    [17/06-16:55:14.054] ID2App removing ID:452

    [17/06-16:55:14.055] MGWConnThread sending: [HangingUp /A21/I452/o0/R10/#0]

    [17/06-16:55:14.059] [debug] ProcessLine received from hgs: [HangingUp /A21/I452/o0/R10/#0]

    [17/06-16:55:14.061] [debug] ProcessLine invoking HangingUp with tid: 452 cid: 0 params: /A21/I452/o0/R10/#0

    [17/06-16:55:14.061] [debug] OnPhoneHangup (UAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ./452) media was not started - unregistering

    [17/06-16:55:14.062] [debug] UnregisterSession 1:0 -> UAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ./452

    [17/06-16:55:14.063] [debug] Close (UAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ./452)

    [17/06-16:55:14.063] [debug] Close (UAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ./452) closing session timeslot:0 handle: -1

    [17/06-16:55:14.064] [debug] CloseResources (UAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ./452)

    [17/06-16:55:14.064] [debug] CloseAudioChannel (UAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ./452) channel timeslot: 0 already closed

    [17/06-16:55:14.065] [debug] CloseResources (UAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ./452) remove media resoruce: 14.0

    [17/06-16:55:14.065] [debug] DisconnectEndpoints (UAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ./452)

    [17/06-16:55:14.066] [debug] ClearNetCall (UAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ./452) Cause Code: 10 Converted Cause Code: 10

    [17/06-16:55:14.066] [debug] UpdateStatsAndCDR (UAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ./452)

    [17/06-16:55:14.067] [info] VoIP CDR: 1,2011-06-17 T 16:54:52,2011-06-17 T 16:54:52,0012012181444,8768776075,,2011-06-17 T 16:55:14,,,,,1,0,1,1,0,0,0,10

    [17/06-16:55:14.068] [debug] ~tMGWSession (UAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ./452)

    [17/06-16:55:14.068] [debug] clearCall ClearCall [uAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ.]

    [17/06-16:55:14.069] [debug] setHangupReason m_nHangupQ931: 10 m_nHangupSIP: 480

    [17/06-16:55:14.069] [debug] removeSession Call Id [uAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ.]: m_tokenToSession.erase

    [17/06-16:55:14.070] [debug] ~tSessionInfo Call id [uAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ.]: Calling ~tSessionInfo

    [17/06-16:55:14.512] HMCServer received [MGWResStatus /I124/AMG] from 192.168.1.14:62261

    [17/06-16:55:14.513] Sending: [MGWResStatus /I124/AMG], Client ID:12

    [17/06-16:55:14.514] MGWConnThread sending: [MGWResStatus /I124/AMG/#12]

    [17/06-16:55:14.518] [debug] ProcessLine received from hgs: [MGWResStatus /I124/AMG/#12]

    [17/06-16:55:14.519] [debug] ProcessLine invoking MGWResStatus with tid: 124 cid: 12 params: /I124/AMG/#12

    [17/06-16:55:14.520] Received

  6. Supervised Transfer Not working

    in Microsoft Speech Server

    Posted

    Does the HG4000 support UDP? This is because the 200 Ok comes on TCP transport layer and contains a contact that is implies UDP. So the PBX sends it on UDP. Because the gateway repeats the 200 Ok, it seems that the ACK does not make it and this transport layer problem could be the reason.

     

    yes HG 4000 Supports both UDP and TCP. There is an option to set the protocol. If I set it to UDP, the incoming call in the first leg does not work. PBX just drops it. I dont even see it in the logs (even if I specify UDP on the trunk as transport). The moment I change the transport to TCP on HG 4000 , PBX sees the call and it works.

     

    Is there any way We can force PBX to send TCP back to HG 4000?

     

    Thanks,

     

    Sanjeev.

  7. Supervised Transfer Not working

    in Microsoft Speech Server

    Posted

    You might be a "victim" of the 8 seconds TCP disconnect issue when the TCP/TLS connection did not register. if you send a pivate message to pbx_support and indicate what OS you have, your problem might go away already.

     

     

    I had problems with TCP Timeout during outbound call because speech server did not authenticate and call timed out after 8 secs and PBXnSIP support team helped by providing the latest .exe files that solved the tcp timeout issue. So the Outbound leg is working fine. Now I just need to transfer the call and I will complete this project (http://doctoroncalljamaica.com) . Please see the logs below.

     

    However, I again see TCP Timeout in the logs...

  8. Supervised Transfer Not working

    in Microsoft Speech Server

    Posted

    Am I missing something? there are no comments here....

    FYI

    I also changed the "Assume Calls from" in the Speech server trunk to 42 to match the "Asuume call comes from" in the Hypermedia Trunk. I hoped that it might help if teh two legs of the call originate from same extension. But it still does not work.

     

    Sorry. I will check teh HG 4000 Logs and let you know.

  9. Supervised Transfer Not working

    in Microsoft Speech Server

    Posted

    Looking at the log, the below messages are being repeated many times. So it looks like the ACK sent by PBX is never received by HG4000/1.0 (or somehow it is dropping it). Can you verify why HG4000/1.0 is re-transmitting the 200 OK message even after the ACK was sent by PBX?

     

    [5] 20110615020821: SIP Rx tcp:192.168.1.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557
Contact: <sip:12012181444@192.168.1.12:5060;user=phone>
To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131
From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005
Call-ID: 1df6b336@pbx
CSeq: 19159 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER
Content-Type: application/sdp
Supported: replaces, norefersub
User-Agent: HG4000/1.0
Content-Length: 189

v=0
o=HG4000 0 0 IN IP4 192.168.1.12
s=HG4000-Session
c=IN IP4 192.168.1.12
t=0 0
m=audio 4010 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

[5] 20110615020821: SIP Tx udp:192.168.1.12:5060:
ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-9e70d3b58c580ffc8526933219ca6651;rport
From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005
To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131
Call-ID: 1df6b336@pbx
CSeq: 19159 ACK
Max-Forwards: 70
Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>
[b]P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>[/b]Content-Length: 0

     

    Am I missing something? there are no comments here....

    FYI

    I also changed the "Assume Calls from" in the Speech server trunk to 42 to match the "Asuume call comes from" in the Hypermedia Trunk. I hoped that it might help if teh two legs of the call originate from same extension. But it still does not work.

  10. Supervised Transfer Not working

    in Microsoft Speech Server

    Posted

    I have the following set up

     

    CellPhone <-> Hypermedia Gateway <-> PBXnSIP <-> Microsoft SpeechServer

     

    I am trying to implement the supervised transfer. The Incoming call works fine. I hear the IVR Prompt. Speech server makes an outbound call (consultation call) to another phone. This works Ok too I hear the IVR Prompt and enter my response. However, after the I enter the response on teh consultation call, and the transfer occurs, I do not hear anything on either phones. after sometime the outbound call hangs up on the phone. However, PBXnSIP still shows teh call as active.

     

    I had problems with TCP Timeout during outbound call because speech server did not authenticate and call timed out after 8 secs and PBXnSIP support team helped by providing the latest .exe files that solved the tcp timeout issue. So the Outbound leg is working fine. Now I just need to transfer the call and I will complete this project (http://doctoroncalljamaica.com) . Please see the logs below. I again see TCP Timeout in the logs...

     

     

     

    INVITE sip:8768776075@192.168.1.13 SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-1d4212783c10c531-1---d8754z-;rport

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.12:5060>

    To: <sip:8768776075@192.168.1.13>

    From: <sip:0012018881440@192.168.1.12:5060>;tag=b853b310

    Call-ID: OWNjNWEyMDJhYTY5ZDcwYWU2NGVlMTQyNGM3MjkxZTA.

    CSeq: 1 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub, 100rel, em

    User-Agent: HG4000/1.0

    Content-Length: 345

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4000 RTP/AVP 4 18 18 18 18 0 8 101

    a=rtpmap:4 G723/8000

    a=rtpmap:18 G729/8000

    a=rtpmap:18 G729a/8000

    a=rtpmap:18 G729b/8000

    a=rtpmap:18 G729ab/8000

    a=rtpmap:0 PCMU/8000

    a=rtpmap:8 PCMA/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110615020610: SIP Tx tcp:192.168.1.12:9224:

    SIP/2.0 100 Trying

    Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-1d4212783c10c531-1---d8754z-;rport=9224

    From: <sip:0012018881440@192.168.1.12:5060>;tag=b853b310

    To: <sip:8768776075@192.168.1.13>;tag=b90bf2c68a

    Call-ID: OWNjNWEyMDJhYTY5ZDcwYWU2NGVlMTQyNGM3MjkxZTA.

    CSeq: 1 INVITE

    Content-Length: 0

     

     

    [5] 20110615020610: Using <sip:0012018881440@192.168.1.12:5060;user=phone> as redirect source address

    [5] 20110615020610: SIP Tx tcp:192.168.1.13:6060:

    INVITE sip:8768776075@192.168.1.13:6060;user=phone SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:52543;branch=z9hG4bK-885b2b907e6916aef67ac0402280350b;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=33746

    To: <sip:8768776075@192.168.1.13:6060;user=phone>

    Call-ID: 7226e2d7@pbx

    CSeq: 16835 INVITE

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:52543;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.1.4009

    Diversion: <tel:42>;reason=unconditional;screen=no;privacy=off

    Related-Call-ID: OWNjNWEyMDJhYTY5ZDcwYWU2NGVlMTQyNGM3MjkxZTA.

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Type: application/sdp

    Content-Length: 327

     

    v=0

    o=- 32718 32718 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 10920 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=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

     

    [5] 20110615020610: SIP Rx tcp:192.168.1.13:6060:

    SIP/2.0 100 Trying

    FROM: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=33746

    TO: <sip:8768776075@192.168.1.13:6060;user=phone>

    CSEQ: 16835 INVITE

    CALL-ID: 7226e2d7@pbx

    VIA: SIP/2.0/TCP 192.168.1.13:52543;branch=z9hG4bK-885b2b907e6916aef67ac0402280350b;rport

    CONTENT-LENGTH: 0

     

     

    [5] 20110615020610: SIP Tx tcp:192.168.1.12:9224:

    SIP/2.0 183 Session Progress

    Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-1d4212783c10c531-1---d8754z-;rport=9224

    From: <sip:0012018881440@192.168.1.12:5060>;tag=b853b310

    To: <sip:8768776075@192.168.1.13>;tag=b90bf2c68a

    Call-ID: OWNjNWEyMDJhYTY5ZDcwYWU2NGVlMTQyNGM3MjkxZTA.

    CSeq: 1 INVITE

    Contact: <sip:8768776075@192.168.1.13:5060;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.1.4009

    Require: 100rel

    RSeq: 1

    Content-Type: application/sdp

    Content-Length: 251

     

    v=0

    o=- 1422 1422 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 50780 RTP/AVP 0 8 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

     

    [5] 20110615020610: SIP Rx tcp:192.168.1.13:6060:

    SIP/2.0 302 Moved Temporarily

    FROM: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=33746

    TO: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=9d5d30897e

    CSEQ: 16835 INVITE

    CALL-ID: 7226e2d7@pbx

    VIA: SIP/2.0/TCP 192.168.1.13:52543;branch=z9hG4bK-885b2b907e6916aef67ac0402280350b;rport

    CONTACT: <sip:8768776075@192.168.1.13:52479;user=phone;transport=Tcp;maddr=192.168.1.13;x-mss-call-id=7226e2d7%40pbx>

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0

     

     

    [7] 20110615020610: Call 7226e2d7@pbx: Clear last INVITE

    [5] 20110615020610: SIP Tx tcp:192.168.1.13:6060:

    ACK sip:8768776075@192.168.1.13:6060;user=phone SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:52543;branch=z9hG4bK-885b2b907e6916aef67ac0402280350b;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=33746

    To: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=9d5d30897e

    Call-ID: 7226e2d7@pbx

    CSeq: 16835 ACK

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:52543;transport=tcp>

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110615020610: Redirecting call

    [5] 20110615020610: SIP Tx tcp:192.168.1.13:52479:

    INVITE sip:8768776075@192.168.1.13:52479;user=phone;transport=Tcp;maddr=192.168.1.13;x-mss-call-id=7226e2d7%40pbx SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-9947fb33db4d71b414e25acf3deba21d;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=33746

    To: <sip:8768776075@192.168.1.13:6060;user=phone>

    Call-ID: 7226e2d7@pbx

    CSeq: 16836 INVITE

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:52544;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.1.4009

    Diversion: <tel:42>;reason=unconditional;screen=no;privacy=off

    Related-Call-ID: OWNjNWEyMDJhYTY5ZDcwYWU2NGVlMTQyNGM3MjkxZTA.

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Type: application/sdp

    Content-Length: 327

     

    v=0

    o=- 32718 32718 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 10920 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=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

     

    [5] 20110615020610: SIP Rx tcp:192.168.1.13:52479:

    SIP/2.0 100 Trying

    FROM: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=33746

    TO: <sip:8768776075@192.168.1.13:6060;user=phone>

    CSEQ: 16836 INVITE

    CALL-ID: 7226e2d7@pbx

    VIA: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-9947fb33db4d71b414e25acf3deba21d;rport

    CONTENT-LENGTH: 0

     

     

    [5] 20110615020610: SIP Rx tcp:192.168.1.13:52479:

    SIP/2.0 180 Ringing

    FROM: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=33746

    TO: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=8E0ADA2D20;tag=5b60ff999

    CSEQ: 16836 INVITE

    CALL-ID: 7226e2d7@pbx

    VIA: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-9947fb33db4d71b414e25acf3deba21d;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0

     

     

    [5] 20110615020610: SIP Rx tcp:192.168.1.13:52479:

    SIP/2.0 200 OK

    FROM: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=33746

    TO: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=8E0ADA2D20;tag=5b60ff999

    CSEQ: 16836 INVITE

    CALL-ID: 7226e2d7@pbx

    VIA: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-9947fb33db4d71b414e25acf3deba21d;rport

    CONTACT: <sip:CommServer.creditfree.local:52479;transport=Tcp;maddr=192.168.1.13>;automata

    CONTENT-LENGTH: 194

    CONTENT-TYPE: application/sdp

    ALLOW: UPDATE

    SERVER: RTCC/3.0.0.0

    ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify

     

    v=0

    o=- 0 0 IN IP4 192.168.1.13

    s=Microsoft Speech Server session

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 13440 RTP/AVP 0 8 101

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

     

    [7] 20110615020610: Call 7226e2d7@pbx: Clear last INVITE

    [5] 20110615020610: SIP Tx tcp:192.168.1.13:52479:

    ACK sip:CommServer.creditfree.local:52479;transport=Tcp;maddr=192.168.1.13 SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-cc94dd24be9fd5e8c08ff1830236a068;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=33746

    To: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=5b60ff999;epid=8E0ADA2D20

    Call-ID: 7226e2d7@pbx

    CSeq: 16836 ACK

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:52544;transport=tcp>

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110615020610: SIP Tx tcp:192.168.1.12:9224:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-1d4212783c10c531-1---d8754z-;rport=9224

    From: <sip:0012018881440@192.168.1.12:5060>;tag=b853b310

    To: <sip:8768776075@192.168.1.13>;tag=b90bf2c68a

    Call-ID: OWNjNWEyMDJhYTY5ZDcwYWU2NGVlMTQyNGM3MjkxZTA.

    CSeq: 1 INVITE

    Contact: <sip:8768776075@192.168.1.13:5060;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.1.4009

    Content-Type: application/sdp

    Content-Length: 251

     

    v=0

    o=- 1422 1422 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 50780 RTP/AVP 0 8 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

     

    [5] 20110615020612: SIP Tx tcp:192.168.1.12:9224:

    BYE sip:0012018881440@192.168.1.12:5060 SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:5060;branch=z9hG4bK-ba981f3f1fe00badc3844a63077c2c5d;rport

    From: <sip:8768776075@192.168.1.13>;tag=b90bf2c68a

    To: <sip:0012018881440@192.168.1.12:5060>;tag=b853b310

    Call-ID: OWNjNWEyMDJhYTY5ZDcwYWU2NGVlMTQyNGM3MjkxZTA.

    CSeq: 21381 BYE

    Max-Forwards: 70

    Contact: <sip:8768776075@192.168.1.13:5060;transport=tcp>

    Content-Length: 0

     

     

    [5] 20110615020612: SIP Tx tcp:192.168.1.13:52479:

    BYE sip:CommServer.creditfree.local:52479;transport=Tcp;maddr=192.168.1.13 SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-9d38c8c5d436718a6eb0a76aec6f17a9;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=33746

    To: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=5b60ff999

    Call-ID: 7226e2d7@pbx

    CSeq: 16837 BYE

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:52544;transport=tcp>

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110615020612: SIP Rx tcp:192.168.1.13:52479:

    SIP/2.0 200 OK

    FROM: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=33746

    TO: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=5b60ff999;epid=8E0ADA2D20

    CSEQ: 16837 BYE

    CALL-ID: 7226e2d7@pbx

    VIA: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-9d38c8c5d436718a6eb0a76aec6f17a9;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0

     

     

    [7] 20110615020612: Call 7226e2d7@pbx: Clear last request

    [5] 20110615020612: BYE Response: Terminate 7226e2d7@pbx

    [5] 20110615020612: SIP Rx tcp:192.168.1.12:9224:

    SIP/2.0 481 Call/Transaction Does Not Exist

    Via: SIP/2.0/TCP 192.168.1.13:5060;branch=z9hG4bK-ba981f3f1fe00badc3844a63077c2c5d;rport=5060

    To: <sip:0012018881440@192.168.1.12:5060>;tag=b853b310

    From: <sip:8768776075@192.168.1.13>;tag=b90bf2c68a

    Call-ID: OWNjNWEyMDJhYTY5ZDcwYWU2NGVlMTQyNGM3MjkxZTA.

    CSeq: 21381 BYE

    Accept-Language: en

    Content-Length: 0

     

     

    [7] 20110615020612: Call OWNjNWEyMDJhYTY5ZDcwYWU2NGVlMTQyNGM3MjkxZTA.: Clear last request

    [5] 20110615020612: BYE Response: Terminate OWNjNWEyMDJhYTY5ZDcwYWU2NGVlMTQyNGM3MjkxZTA.

    [5] 20110615020705: SIP Rx tcp:192.168.1.12:9224:

    INVITE sip:8768776075@192.168.1.13 SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-b5ca0f04f101ea54-1---d8754z-;rport

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.12:5060>

    To: <sip:8768776075@192.168.1.13>

    From: <sip:0012018881440@192.168.1.12:5060>;tag=97716957

    Call-ID: ZjI2ZmQwNjAwYmNlMzhhY2FlNTJhYTQxNWRmODMxN2Y.

    CSeq: 1 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub, 100rel, em

    User-Agent: HG4000/1.0

    Content-Length: 345

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4000 RTP/AVP 4 18 18 18 18 0 8 101

    a=rtpmap:4 G723/8000

    a=rtpmap:18 G729/8000

    a=rtpmap:18 G729a/8000

    a=rtpmap:18 G729b/8000

    a=rtpmap:18 G729ab/8000

    a=rtpmap:0 PCMU/8000

    a=rtpmap:8 PCMA/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110615020705: SIP Tx tcp:192.168.1.12:9224:

    SIP/2.0 100 Trying

    Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-b5ca0f04f101ea54-1---d8754z-;rport=9224

    From: <sip:0012018881440@192.168.1.12:5060>;tag=97716957

    To: <sip:8768776075@192.168.1.13>;tag=2af52cac74

    Call-ID: ZjI2ZmQwNjAwYmNlMzhhY2FlNTJhYTQxNWRmODMxN2Y.

    CSeq: 1 INVITE

    Content-Length: 0

     

     

    [5] 20110615020705: Using <sip:0012018881440@192.168.1.12:5060;user=phone> as redirect source address

    [5] 20110615020705: SIP Tx tcp:192.168.1.13:6060:

    INVITE sip:8768776075@192.168.1.13:6060;user=phone SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:52543;branch=z9hG4bK-63b9477d8a581c99013ba28baf356258;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

    To: <sip:8768776075@192.168.1.13:6060;user=phone>

    Call-ID: 6e790856@pbx

    CSeq: 858 INVITE

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:52543;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.1.4009

    Diversion: <tel:42>;reason=unconditional;screen=no;privacy=off

    Related-Call-ID: ZjI2ZmQwNjAwYmNlMzhhY2FlNTJhYTQxNWRmODMxN2Y.

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Type: application/sdp

    Content-Length: 327

     

    v=0

    o=- 23513 23513 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 30302 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=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

     

    [5] 20110615020705: SIP Rx tcp:192.168.1.13:6060:

    SIP/2.0 100 Trying

    FROM: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

    TO: <sip:8768776075@192.168.1.13:6060;user=phone>

    CSEQ: 858 INVITE

    CALL-ID: 6e790856@pbx

    VIA: SIP/2.0/TCP 192.168.1.13:52543;branch=z9hG4bK-63b9477d8a581c99013ba28baf356258;rport

    CONTENT-LENGTH: 0

     

     

    [5] 20110615020705: SIP Tx tcp:192.168.1.12:9224:

    SIP/2.0 183 Session Progress

    Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-b5ca0f04f101ea54-1---d8754z-;rport=9224

    From: <sip:0012018881440@192.168.1.12:5060>;tag=97716957

    To: <sip:8768776075@192.168.1.13>;tag=2af52cac74

    Call-ID: ZjI2ZmQwNjAwYmNlMzhhY2FlNTJhYTQxNWRmODMxN2Y.

    CSeq: 1 INVITE

    Contact: <sip:8768776075@192.168.1.13:5060;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.1.4009

    Require: 100rel

    RSeq: 1

    Content-Type: application/sdp

    Content-Length: 251

     

    v=0

    o=- 3356 3356 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 52784 RTP/AVP 0 8 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

     

    [5] 20110615020705: SIP Rx tcp:192.168.1.13:6060:

    SIP/2.0 302 Moved Temporarily

    FROM: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

    TO: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=49bbd880fd

    CSEQ: 858 INVITE

    CALL-ID: 6e790856@pbx

    VIA: SIP/2.0/TCP 192.168.1.13:52543;branch=z9hG4bK-63b9477d8a581c99013ba28baf356258;rport

    CONTACT: <sip:8768776075@192.168.1.13:52479;user=phone;transport=Tcp;maddr=192.168.1.13;x-mss-call-id=6e790856%40pbx>

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0

     

     

    [7] 20110615020705: Call 6e790856@pbx: Clear last INVITE

    [5] 20110615020705: SIP Tx tcp:192.168.1.13:6060:

    ACK sip:8768776075@192.168.1.13:6060;user=phone SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:52543;branch=z9hG4bK-63b9477d8a581c99013ba28baf356258;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

    To: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=49bbd880fd

    Call-ID: 6e790856@pbx

    CSeq: 858 ACK

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:52543;transport=tcp>

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110615020705: Redirecting call

    [5] 20110615020705: SIP Tx tcp:192.168.1.13:52479:

    INVITE sip:8768776075@192.168.1.13:52479;user=phone;transport=Tcp;maddr=192.168.1.13;x-mss-call-id=6e790856%40pbx SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-140e26839d6c2ca053d84501bf7bb611;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

    To: <sip:8768776075@192.168.1.13:6060;user=phone>

    Call-ID: 6e790856@pbx

    CSeq: 859 INVITE

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:52544;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.1.4009

    Diversion: <tel:42>;reason=unconditional;screen=no;privacy=off

    Related-Call-ID: ZjI2ZmQwNjAwYmNlMzhhY2FlNTJhYTQxNWRmODMxN2Y.

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Type: application/sdp

    Content-Length: 327

     

    v=0

    o=- 23513 23513 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 30302 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=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

     

    [5] 20110615020705: SIP Rx tcp:192.168.1.13:52479:

    SIP/2.0 100 Trying

    FROM: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

    TO: <sip:8768776075@192.168.1.13:6060;user=phone>

    CSEQ: 859 INVITE

    CALL-ID: 6e790856@pbx

    VIA: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-140e26839d6c2ca053d84501bf7bb611;rport

    CONTENT-LENGTH: 0

     

     

    [5] 20110615020705: SIP Rx tcp:192.168.1.13:52479:

    SIP/2.0 180 Ringing

    FROM: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

    TO: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=8E0ADA2D20;tag=649944c4f

    CSEQ: 859 INVITE

    CALL-ID: 6e790856@pbx

    VIA: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-140e26839d6c2ca053d84501bf7bb611;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0

     

     

    [5] 20110615020705: SIP Rx tcp:192.168.1.13:52479:

    SIP/2.0 200 OK

    FROM: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

    TO: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=8E0ADA2D20;tag=649944c4f

    CSEQ: 859 INVITE

    CALL-ID: 6e790856@pbx

    VIA: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-140e26839d6c2ca053d84501bf7bb611;rport

    CONTACT: <sip:CommServer.creditfree.local:52479;transport=Tcp;maddr=192.168.1.13>;automata

    CONTENT-LENGTH: 194

    CONTENT-TYPE: application/sdp

    ALLOW: UPDATE

    SERVER: RTCC/3.0.0.0

    ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify

     

    v=0

    o=- 0 0 IN IP4 192.168.1.13

    s=Microsoft Speech Server session

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 35840 RTP/AVP 0 8 101

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

     

    [7] 20110615020705: Call 6e790856@pbx: Clear last INVITE

    [5] 20110615020705: SIP Tx tcp:192.168.1.13:52479:

    ACK sip:CommServer.creditfree.local:52479;transport=Tcp;maddr=192.168.1.13 SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-8dad98c585973e9425c80587931750c9;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

    To: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=649944c4f;epid=8E0ADA2D20

    Call-ID: 6e790856@pbx

    CSeq: 859 ACK

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:52544;transport=tcp>

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110615020705: SIP Tx tcp:192.168.1.12:9224:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-b5ca0f04f101ea54-1---d8754z-;rport=9224

    From: <sip:0012018881440@192.168.1.12:5060>;tag=97716957

    To: <sip:8768776075@192.168.1.13>;tag=2af52cac74

    Call-ID: ZjI2ZmQwNjAwYmNlMzhhY2FlNTJhYTQxNWRmODMxN2Y.

    CSeq: 1 INVITE

    Contact: <sip:8768776075@192.168.1.13:5060;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.1.4009

    Content-Type: application/sdp

    Content-Length: 251

     

    v=0

    o=- 3356 3356 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 52784 RTP/AVP 0 8 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

     

    [5] 20110615020705: SIP Rx tcp:192.168.1.12:9224:

    ACK sip:8768776075@192.168.1.13:5060;transport=tcp SIP/2.0

    Via: SIP/2.0/TCP 192.168.0.2:5060;branch=z9hG4bK-d8754z-7c1251351bd6f75e-1---d8754z-;rport

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.12:5060>

    To: <sip:8768776075@192.168.1.13>;tag=2af52cac74

    From: <sip:0012018881440@192.168.1.12:5060>;tag=97716957

    Call-ID: ZjI2ZmQwNjAwYmNlMzhhY2FlNTJhYTQxNWRmODMxN2Y.

    CSeq: 1 ACK

    User-Agent: HG4000/1.0

    Content-Length: 0

     

     

    [5] 20110615020737: SIP Rx tcp:192.168.1.13:52479:

    INVITE sip:0012018881440@192.168.1.13:52544;transport=tcp SIP/2.0

    FROM: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=8E0ADA2D20;tag=649944c4f

    TO: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

    CSEQ: 1 INVITE

    CALL-ID: 6e790856@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 192.168.1.13:52479;branch=z9hG4bK1b7861a6

    CONTACT: <sip:CommServer.creditfree.local:52479;transport=Tcp;maddr=192.168.1.13;ms-opaque=5cf12f79b09db613>;automata

    CONTENT-LENGTH: 206

    USER-AGENT: RTCC/3.0.0.0

    CONTENT-TYPE: application/sdp

     

    v=0

    o=- 0 0 IN IP4 192.168.1.13

    s=Microsoft Speech Server session

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 35840 RTP/AVP 0 8 101

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=sendonly

    a=ptime:20

     

    [5] 20110615020737: SIP Tx tcp:192.168.1.13:52479:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 192.168.1.13:52479;branch=z9hG4bK1b7861a6

    From: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=649944c4f;epid=8E0ADA2D20

    To: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

    Call-ID: 6e790856@pbx

    CSeq: 1 INVITE

    Contact: <sip:0012018881440@192.168.1.13:52544;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.1.4009

    Content-Type: application/sdp

    Content-Length: 265

     

    v=0

    o=- 23513 23513 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 30302 RTP/AVP 0 8 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=recvonly

     

    [5] 20110615020737: SIP Rx tcp:192.168.1.13:52479:

    ACK sip:0012018881440@192.168.1.13:52544;transport=tcp SIP/2.0

    FROM: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=8E0ADA2D20;tag=649944c4f

    TO: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

    CSEQ: 1 ACK

    CALL-ID: 6e790856@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 192.168.1.13:52479;branch=z9hG4bK4ac3478c

    CONTENT-LENGTH: 0

    USER-AGENT: RTCC/3.0.0.0

     

     

    [5] 20110615020737: SIP Rx tcp:192.168.1.13:52556:

    INVITE sip:12012181444@192.168.1.13:5060;transport=tcp SIP/2.0

    FROM: <sip:0012018881440@192.168.1.12:5060>;epid=8E0ADA2D20;tag=5454b672e0

    TO: <sip:12012181444@192.168.1.13:5060;transport=tcp>

    CSEQ: 2 INVITE

    CALL-ID: 4d74728f-46c9-4a48-9268-c6b892a4ecc8

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 192.168.1.13:52556;branch=z9hG4bKcd18343

    CONTACT: <sip:CommServer.creditfree.local:52479;transport=Tcp;maddr=192.168.1.13;ms-opaque=5cf12f79b09db613>;automata

    CONTENT-LENGTH: 336

    USER-AGENT: RTCC/3.0.0.0

    CONTENT-TYPE: application/sdp

    ALLOW: UPDATE

    ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify

     

    v=0

    o=- 0 0 IN IP4 192.168.1.13

    s=Microsoft Speech Server session

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 13440 RTP/AVP 114 115 4 0 8 97 101

    a=rtpmap:114 x-msrta/16000

    a=fmtp:114 bitrate=29000

    a=rtpmap:115 x-msrta/8000

    a=fmtp:115 bitrate=11800

    a=rtpmap:97 RED/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

     

    [5] 20110615020737: SIP Tx tcp:192.168.1.13:52556:

    SIP/2.0 100 Trying

    Via: SIP/2.0/TCP 192.168.1.13:52556;branch=z9hG4bKcd18343

    From: <sip:0012018881440@192.168.1.12:5060>;tag=5454b672e0;epid=8E0ADA2D20

    To: <sip:12012181444@192.168.1.13:5060;transport=tcp>;tag=9936c19f3f

    Call-ID: 4d74728f-46c9-4a48-9268-c6b892a4ecc8

    CSeq: 2 INVITE

    Content-Length: 0

     

     

    [5] 20110615020737: Using <sip:0012018881440@192.168.1.12:5060;user=phone> as redirect source address

    [5] 20110615020737: SIP Tx tcp:192.168.1.12:5060:

    INVITE sip:12012181444@192.168.1.12;user=phone SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    To: <sip:12012181444@192.168.1.12;user=phone>

    Call-ID: 1df6b336@pbx

    CSeq: 19159 INVITE

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:52557;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.1.4009

    Diversion: <tel:45>;reason=unconditional;screen=no;privacy=off

    Related-Call-ID: 4d74728f-46c9-4a48-9268-c6b892a4ecc8

    P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

    Content-Type: application/sdp

    Content-Length: 327

     

    v=0

    o=- 28916 28916 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 38056 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=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

     

    [5] 20110615020737: SIP Tx tcp:192.168.1.13:52556:

    SIP/2.0 183 Session Progress

    Via: SIP/2.0/TCP 192.168.1.13:52556;branch=z9hG4bKcd18343

    From: <sip:0012018881440@192.168.1.12:5060>;tag=5454b672e0;epid=8E0ADA2D20

    To: <sip:12012181444@192.168.1.13:5060;transport=tcp>;tag=9936c19f3f

    Call-ID: 4d74728f-46c9-4a48-9268-c6b892a4ecc8

    CSeq: 2 INVITE

    Contact: <sip:12012181444@192.168.1.13:5060;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.1.4009

    Content-Type: application/sdp

    Content-Length: 263

     

    v=0

    o=- 6605 6605 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 10552 RTP/AVP 0 8 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

     

    [5] 20110615020737: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 183 Session Progress

    Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557

    Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    Call-ID: 1df6b336@pbx

    CSeq: 19159 INVITE

    Content-Type: application/sdp

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4010 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110615020742: Did not receive ACK, disconnecting call OWNjNWEyMDJhYTY5ZDcwYWU2NGVlMTQyNGM3MjkxZTA.

    [5] 20110615020750: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557

    Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    Call-ID: 1df6b336@pbx

    CSeq: 19159 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4010 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [7] 20110615020750: Call 1df6b336@pbx: Clear last INVITE

    [5] 20110615020750: SIP Tx udp:192.168.1.12:5060:

    ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-9e70d3b58c580ffc8526933219ca6651;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    Call-ID: 1df6b336@pbx

    CSeq: 19159 ACK

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110615020750: SIP Tx tcp:192.168.1.13:52556:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 192.168.1.13:52556;branch=z9hG4bKcd18343

    From: <sip:0012018881440@192.168.1.12:5060>;tag=5454b672e0;epid=8E0ADA2D20

    To: <sip:12012181444@192.168.1.13:5060;transport=tcp>;tag=9936c19f3f

    Call-ID: 4d74728f-46c9-4a48-9268-c6b892a4ecc8

    CSeq: 2 INVITE

    Contact: <sip:12012181444@192.168.1.13:5060;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.1.4009

    Content-Type: application/sdp

    Content-Length: 263

     

    v=0

    o=- 6605 6605 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 10552 RTP/AVP 0 8 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

     

    [5] 20110615020750: SIP Rx tcp:192.168.1.13:52556:

    ACK sip:12012181444@192.168.1.13:5060;transport=tcp SIP/2.0

    FROM: <sip:0012018881440@192.168.1.12:5060>;epid=8E0ADA2D20;tag=5454b672e0

    TO: <sip:12012181444@192.168.1.13:5060;transport=tcp>;tag=9936c19f3f

    CSEQ: 2 ACK

    CALL-ID: 4d74728f-46c9-4a48-9268-c6b892a4ecc8

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 192.168.1.13:52556;branch=z9hG4bKd3a69aa1

    CONTENT-LENGTH: 0

    USER-AGENT: RTCC/3.0.0.0

     

     

    [5] 20110615020750: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557

    Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    Call-ID: 1df6b336@pbx

    CSeq: 19159 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4010 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110615020750: SIP Tx udp:192.168.1.12:5060:

    ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-9e70d3b58c580ffc8526933219ca6651;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    Call-ID: 1df6b336@pbx

    CSeq: 19159 ACK

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110615020751: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557

    Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    Call-ID: 1df6b336@pbx

    CSeq: 19159 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4010 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110615020751: SIP Tx udp:192.168.1.12:5060:

    ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-9e70d3b58c580ffc8526933219ca6651;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    Call-ID: 1df6b336@pbx

    CSeq: 19159 ACK

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110615020753: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557

    Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    Call-ID: 1df6b336@pbx

    CSeq: 19159 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4010 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110615020753: SIP Tx udp:192.168.1.12:5060:

    ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-9e70d3b58c580ffc8526933219ca6651;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    Call-ID: 1df6b336@pbx

    CSeq: 19159 ACK

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110615020757: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557

    Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    Call-ID: 1df6b336@pbx

    CSeq: 19159 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4010 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110615020757: SIP Tx udp:192.168.1.12:5060:

    ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-9e70d3b58c580ffc8526933219ca6651;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    Call-ID: 1df6b336@pbx

    CSeq: 19159 ACK

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110615020801: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557

    Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    Call-ID: 1df6b336@pbx

    CSeq: 19159 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4010 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110615020801: SIP Tx udp:192.168.1.12:5060:

    ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-9e70d3b58c580ffc8526933219ca6651;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    Call-ID: 1df6b336@pbx

    CSeq: 19159 ACK

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [6] 20110615020805: SIP TCP/TLS timeout on 192.168.1.13:6060, closing connection

    [5] 20110615020805: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557

    Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    Call-ID: 1df6b336@pbx

    CSeq: 19159 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4010 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110615020805: SIP Tx udp:192.168.1.12:5060:

    ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-9e70d3b58c580ffc8526933219ca6651;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    Call-ID: 1df6b336@pbx

    CSeq: 19159 ACK

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110615020809: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557

    Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    Call-ID: 1df6b336@pbx

    CSeq: 19159 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4010 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110615020809: SIP Tx udp:192.168.1.12:5060:

    ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-9e70d3b58c580ffc8526933219ca6651;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    Call-ID: 1df6b336@pbx

    CSeq: 19159 ACK

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110615020813: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557

    Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    Call-ID: 1df6b336@pbx

    CSeq: 19159 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4010 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110615020813: SIP Tx udp:192.168.1.12:5060:

    ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-9e70d3b58c580ffc8526933219ca6651;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    Call-ID: 1df6b336@pbx

    CSeq: 19159 ACK

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110615020817: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557

    Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    Call-ID: 1df6b336@pbx

    CSeq: 19159 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4010 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110615020817: SIP Tx udp:192.168.1.12:5060:

    ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-9e70d3b58c580ffc8526933219ca6651;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    Call-ID: 1df6b336@pbx

    CSeq: 19159 ACK

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110615020821: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557

    Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    Call-ID: 1df6b336@pbx

    CSeq: 19159 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4010 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [5] 20110615020821: SIP Tx udp:192.168.1.12:5060:

    ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-9e70d3b58c580ffc8526933219ca6651;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    Call-ID: 1df6b336@pbx

    CSeq: 19159 ACK

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110615020828: SIP Rx tcp:192.168.1.12:9224:

    BYE sip:8768776075@192.168.1.13:5060;transport=tcp SIP/2.0

    Via: SIP/2.0/TCP 192.168.0.2:5060;branch=z9hG4bK-d8754z-6c29ed1dac6bad0e-1---d8754z-;rport

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.12:5060>

    To: <sip:8768776075@192.168.1.13>;tag=2af52cac74

    From: <sip:0012018881440@192.168.1.12:5060>;tag=97716957

    Call-ID: ZjI2ZmQwNjAwYmNlMzhhY2FlNTJhYTQxNWRmODMxN2Y.

    CSeq: 2 BYE

    User-Agent: HG4000/1.0

    Reason: SIP;description="ACK not received"

    Content-Length: 0

     

     

    [5] 20110615020828: SIP Tx tcp:192.168.1.12:9224:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 192.168.0.2:5060;branch=z9hG4bK-d8754z-6c29ed1dac6bad0e-1---d8754z-;rport=9224;received=192.168.1.12

    From: <sip:0012018881440@192.168.1.12:5060>;tag=97716957

    To: <sip:8768776075@192.168.1.13>;tag=2af52cac74

    Call-ID: ZjI2ZmQwNjAwYmNlMzhhY2FlNTJhYTQxNWRmODMxN2Y.

    CSeq: 2 BYE

    Contact: <sip:8768776075@192.168.1.13:5060;transport=tcp>

    User-Agent: snom-PBX/2011-4.2.1.4009

    Content-Length: 0

     

     

    [5] 20110615020828: SIP Tx tcp:192.168.1.13:52479:

    BYE sip:CommServer.creditfree.local:52479;transport=Tcp;maddr=192.168.1.13 SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-8fee055997a827e58c90fb6c7e9671e2;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

    To: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=649944c4f

    Call-ID: 6e790856@pbx

    CSeq: 860 BYE

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:52544;transport=tcp>

    P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110615020828: SIP Rx tcp:192.168.1.13:52479:

    SIP/2.0 200 OK

    FROM: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

    TO: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=649944c4f;epid=8E0ADA2D20

    CSEQ: 860 BYE

    CALL-ID: 6e790856@pbx

    VIA: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-8fee055997a827e58c90fb6c7e9671e2;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0

     

     

    [7] 20110615020828: Call 6e790856@pbx: Clear last request

    [5] 20110615020828: BYE Response: Terminate 6e790856@pbx

    [5] 20110615020837: SIP Rx udp:192.168.1.13:41032:

    SUBSCRIBE sip:1000@192.168.1.13 SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:41032;branch=z9hG4bK-d8754z-b34c6224c862d35c-1---d8754z-;rport

    Max-Forwards: 70

    Contact: <sip:1000@192.168.1.13:41032>

    To: "1000"<sip:1000@192.168.1.13>

    From: "1000"<sip:1000@192.168.1.13>;tag=161e10eb

    Call-ID: OGQxMWQwZmFmYjQwMjUxZWU4M2NmNzkyNDQ5ZjgwZmQ.

    CSeq: 1 SUBSCRIBE

    Expires: 300

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

    User-Agent: X-Lite 4 release 4.0 stamp 58832

    Event: message-summary

    Content-Length: 0

     

     

    [5] 20110615020837: SIP Tx udp:192.168.1.13:41032:

    SIP/2.0 404 Not Found

    Via: SIP/2.0/UDP 192.168.1.13:41032;branch=z9hG4bK-d8754z-b34c6224c862d35c-1---d8754z-;rport=41032

    From: "1000" <sip:1000@192.168.1.13>;tag=161e10eb

    To: "1000" <sip:1000@192.168.1.13>;tag=e2eaf47150

    Call-ID: OGQxMWQwZmFmYjQwMjUxZWU4M2NmNzkyNDQ5ZjgwZmQ.

    CSeq: 1 SUBSCRIBE

    Content-Length: 0

     

     

    [5] 20110615020913: SIP Tx tcp:192.168.1.13:52556:

    BYE sip:CommServer.creditfree.local:52479;transport=Tcp;maddr=192.168.1.13;ms-opaque=5cf12f79b09db613 SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:5060;branch=z9hG4bK-fec7a8bfdfa5cbe6c65c284973f55bc3;rport

    From: <sip:12012181444@192.168.1.13:5060;transport=tcp>;tag=9936c19f3f

    To: <sip:0012018881440@192.168.1.12:5060>;tag=5454b672e0;epid=8E0ADA2D20

    Call-ID: 4d74728f-46c9-4a48-9268-c6b892a4ecc8

    CSeq: 31135 BYE

    Max-Forwards: 70

    Contact: <sip:12012181444@192.168.1.13:5060;transport=tcp>

    Content-Length: 0

     

     

    [5] 20110615020913: SIP Tx udp:192.168.1.12:5060:

    BYE sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-c31f3d66d57f59090580890239bf9e84;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    Call-ID: 1df6b336@pbx

    CSeq: 19160 BYE

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110615020913: SIP Rx tcp:192.168.1.13:52556:

    SIP/2.0 200 OK

    FROM: <sip:12012181444@192.168.1.13:5060;transport=tcp>;tag=9936c19f3f

    TO: <sip:0012018881440@192.168.1.12:5060>;tag=5454b672e0;epid=8E0ADA2D20

    CSEQ: 31135 BYE

    CALL-ID: 4d74728f-46c9-4a48-9268-c6b892a4ecc8

    VIA: SIP/2.0/TCP 192.168.1.13:5060;branch=z9hG4bK-fec7a8bfdfa5cbe6c65c284973f55bc3;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0

     

     

    [7] 20110615020913: Call 4d74728f-46c9-4a48-9268-c6b892a4ecc8: Clear last request

    [5] 20110615020913: BYE Response: Terminate 4d74728f-46c9-4a48-9268-c6b892a4ecc8

    [5] 20110615020913: SIP Tr udp:192.168.1.12:5060:

    BYE sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-c31f3d66d57f59090580890239bf9e84;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    Call-ID: 1df6b336@pbx

    CSeq: 19160 BYE

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110615020921: Last message repeated 4 times

    [6] 20110615020921: SIP TCP/TLS timeout on 192.168.1.12:5060, closing connection

    [5] 20110615020924: SIP Tr udp:192.168.1.12:5060:

    BYE sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-c31f3d66d57f59090580890239bf9e84;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    Call-ID: 1df6b336@pbx

    CSeq: 19160 BYE

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [6] 20110615020928: SIP TCP/TLS timeout on 192.168.1.13:52479, closing connection

    [5] 20110615020928: SIP Tr udp:192.168.1.12:5060:

    BYE sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-c31f3d66d57f59090580890239bf9e84;rport

    From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

    Call-ID: 1df6b336@pbx

    CSeq: 19160 BYE

    Max-Forwards: 70

    Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [5] 20110615020945: Last message repeated 5 times

    [7] 20110615020945: Call 1df6b336@pbx: Clear last request

    [5] 20110615020945: BYE Response: Terminate 1df6b336@pbx

  12. speech server 2007 - snom on - outbound call - authenticatin isue

    in Trunk Setup

    Posted

    A few months ago, we found out that the PBX closes a connection that has not been authenticated with a successful registration, after 8 seconds (this is to keep hackers away). We extended the logic so that also successful INVITE requests also keep the connection alive. It seems like you dont have the latest version (where did you get the link from), maybe just private message pbx_support, indicate your OS and then we'll send you the link with the latest build.

     

    actually I got a link from SNOM One when I registered and I downloaded that Link. My OS is windows 2008 server (64 bit). Can you please send the link to teh latest build.

     

    Thanks

     

    Sanjeev.

  16. speech server 2007 - snom on - outbound call - authenticatin isue

    in Trunk Setup

    Posted

    I Just blanked the Password for Extension 45 and the Authentication problem went away. however, Now I have a different problem. When I place an outgoing call, the phone rings but I cannot hear the Speech server prompt. It is silent. I have included the partial log that shows sections where I think the problem lies. Also The etire log is provided. As soon as I trigger the call from speech server, I check teh log and it shows "ession in progress". The phone rings and I pick up the call. I hear nothing. In the log the TCP/TLS connection times out... and later the transport changes to UDP and it is no longer tcp. It then complains that "[8] 2011/06/10 17:20:21: Call 37a2fda3@pbx: Response does not correspond to open request "

     

    any idea how to resolve this? BTW My gateway Hypermedia HG4000.

     

    Partial Log[9] 2011/06/10 17:20:06: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 183 Session Progress

    Via: SIP/2.0/TCP 192.168.1.13:50230;branch=z9hG4bK-ce635a49a2a692a1bfee46cad8bc6e27;rport=50230

    Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

    From: "Forty Five"<sip:45@pbx.company.com;user=phone>;tag=31523

    Call-ID: 37a2fda3@pbx

    CSeq: 27729 INVITE

    Content-Type: application/sdp

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4000 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [6] 2011/06/10 17:20:14: SIP TCP/TLS timeout on 192.168.1.13:50228, closing connection

    [8] 2011/06/10 17:20:14: Release SIP thread 324

    [9] 2011/06/10 17:20:21: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:50230;branch=z9hG4bK-ce635a49a2a692a1bfee46cad8bc6e27;rport=50230

    Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

    From: "Forty Five"<sip:45@pbx.company.com;user=phone>;tag=31523

    Call-ID: 37a2fda3@pbx

    CSeq: 27729 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4000 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [7] 2011/06/10 17:20:21: Call 37a2fda3@pbx: Clear last INVITE

    [9] 2011/06/10 17:20:21: Resolve 902: url sip:12012181444@192.168.1.12:5060;user=phone

    [9] 2011/06/10 17:20:21: Resolve 902: udp 192.168.1.12 5060

    [9] 2011/06/10 17:20:21: SIP Tx udp:192.168.1.12:5060:

    ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-ca096733c6c524b041e5e2755d954e74;rport

    From: "Forty Five" <sip:45@pbx.company.com;user=phone>;tag=31523

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

    Call-ID: 37a2fda3@pbx

    CSeq: 27729 ACK

    Max-Forwards: 70

    Contact: <sip:45@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [9] 2011/06/10 17:20:21: Resolve 903: tcp 192.168.1.13 50228

    [6] 2011/06/10 17:20:21: Response to 192.168.1.13:50228 must be sent over existing connection

    [9] 2011/06/10 17:20:21: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:50230;branch=z9hG4bK-ce635a49a2a692a1bfee46cad8bc6e27;rport=50230

    Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

    From: "Forty Five"<sip:45@pbx.company.com;user=phone>;tag=31523

    Call-ID: 37a2fda3@pbx

    CSeq: 27729 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4000 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [8] 2011/06/10 17:20:21: Call 37a2fda3@pbx: Response does not correspond to open request

    [9] 2011/06/10 17:20:21: Resolve 904: url sip:12012181444@192.168.1.12:5060;user=phone

    [9] 2011/06/10 17:20:21: Resolve 904: udp 192.168.1.12 5060

    [9] 2011/06/10 17:20:21: SIP Tx udp:192.168.1.12:5060:

    ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-ca096733c6c524b041e5e2755d954e74;rport

    From: "Forty Five" <sip:45@pbx.company.com;user=phone>;tag=31523

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

    Call-ID: 37a2fda3@pbx

    CSeq: 27729 ACK

    Max-Forwards: 70

    Contact: <sip:45@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

    Content-Length: 0

     

     

     

     

     

    [8] 2011/06/10 17:20:06: Received SIP connection 324 from 192.168.1.13:50228

    [9] 2011/06/10 17:20:06: SIP Rx tcp:192.168.1.13:50228:

    INVITE sip:12012181444@192.168.1.13:5060;user=phone SIP/2.0

    FROM: <sip:45@CommServer.creditfree.local:50682;user=phone>;epid=67717EB21D;tag=9751c85997

    TO: <sip:12012181444@192.168.1.13:5060;user=phone>

    CSEQ: 9 INVITE

    CALL-ID: b8384679-1360-46f3-9c8a-74235a9f71f9

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 192.168.1.13:50228;branch=z9hG4bK202785c

    CONTACT: <sip:CommServer.creditfree.local:50682;transport=Tcp;maddr=192.168.1.13;ms-opaque=f2f2b3b4e0c93efb>;automata

    CONTENT-LENGTH: 336

    USER-AGENT: RTCC/3.0.0.0

    CONTENT-TYPE: application/sdp

    ALLOW: UPDATE

    ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify

     

    v=0

    o=- 0 0 IN IP4 192.168.1.13

    s=Microsoft Speech Server session

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 64000 RTP/AVP 114 115 4 0 8 97 101

    a=rtpmap:114 x-msrta/16000

    a=fmtp:114 bitrate=29000

    a=rtpmap:115 x-msrta/8000

    a=fmtp:115 bitrate=11800

    a=rtpmap:97 RED/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

     

    [9] 2011/06/10 17:20:06: SIP Tx tcp:192.168.1.13:50228:

    SIP/2.0 100 Trying

    Via: SIP/2.0/TCP 192.168.1.13:50228;branch=z9hG4bK202785c

    From: <sip:45@CommServer.creditfree.local:50682;user=phone>;tag=9751c85997;epid=67717EB21D

    To: <sip:12012181444@192.168.1.13:5060;user=phone>;tag=6762cade95

    Call-ID: b8384679-1360-46f3-9c8a-74235a9f71f9

    CSeq: 9 INVITE

    Content-Length: 0

     

     

    [8] 2011/06/10 17:20:06: Set the To domain based on From user 45@pbx.company.com

    [9] 2011/06/10 17:20:06: Resolve 900: url sip:192.168.1.12:5060;transport=tcp

    [9] 2011/06/10 17:20:06: Resolve 900: a tcp 192.168.1.12 5060

    [9] 2011/06/10 17:20:06: Resolve 900: tcp 192.168.1.12 5060

    [8] 2011/06/10 17:20:06: Received SIP connection 325 from 192.168.1.12:5060

    [9] 2011/06/10 17:20:06: SIP Tx tcp:192.168.1.12:5060:

    INVITE sip:12012181444@192.168.1.12;user=phone SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.13:50230;branch=z9hG4bK-ce635a49a2a692a1bfee46cad8bc6e27;rport

    From: "Forty Five" <sip:45@pbx.company.com;user=phone>;tag=31523

    To: <sip:12012181444@192.168.1.12;user=phone>

    Call-ID: 37a2fda3@pbx

    CSeq: 27729 INVITE

    Max-Forwards: 70

    Contact: <sip:45@192.168.1.13:50230;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/4.2.0.3950

    P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

    Content-Type: application/sdp

    Content-Length: 327

     

    v=0

    o=- 63976 63976 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 31730 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=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

     

    [9] 2011/06/10 17:20:06: SIP Tx tcp:192.168.1.13:50228:

    SIP/2.0 183 Session Progress

    Via: SIP/2.0/TCP 192.168.1.13:50228;branch=z9hG4bK202785c

    From: <sip:45@CommServer.creditfree.local:50682;user=phone>;tag=9751c85997;epid=67717EB21D

    To: <sip:12012181444@192.168.1.13:5060;user=phone>;tag=6762cade95

    Call-ID: b8384679-1360-46f3-9c8a-74235a9f71f9

    CSeq: 9 INVITE

    Contact: <sip:45@192.168.1.13:5060;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/4.2.0.3950

    Content-Type: application/sdp

    Content-Length: 264

     

    v=0

    o=- 57454 57454 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 6566 RTP/AVP 0 8 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

     

    [9] 2011/06/10 17:20:06: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 183 Session Progress

    Via: SIP/2.0/TCP 192.168.1.13:50230;branch=z9hG4bK-ce635a49a2a692a1bfee46cad8bc6e27;rport=50230

    Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

    From: "Forty Five"<sip:45@pbx.company.com;user=phone>;tag=31523

    Call-ID: 37a2fda3@pbx

    CSeq: 27729 INVITE

    Content-Type: application/sdp

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4000 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [6] 2011/06/10 17:20:14: SIP TCP/TLS timeout on 192.168.1.13:50228, closing connection

    [8] 2011/06/10 17:20:14: Release SIP thread 324

    [9] 2011/06/10 17:20:21: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:50230;branch=z9hG4bK-ce635a49a2a692a1bfee46cad8bc6e27;rport=50230

    Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

    From: "Forty Five"<sip:45@pbx.company.com;user=phone>;tag=31523

    Call-ID: 37a2fda3@pbx

    CSeq: 27729 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4000 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [7] 2011/06/10 17:20:21: Call 37a2fda3@pbx: Clear last INVITE

    [9] 2011/06/10 17:20:21: Resolve 902: url sip:12012181444@192.168.1.12:5060;user=phone

    [9] 2011/06/10 17:20:21: Resolve 902: udp 192.168.1.12 5060

    [9] 2011/06/10 17:20:21: SIP Tx udp:192.168.1.12:5060:

    ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-ca096733c6c524b041e5e2755d954e74;rport

    From: "Forty Five" <sip:45@pbx.company.com;user=phone>;tag=31523

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

    Call-ID: 37a2fda3@pbx

    CSeq: 27729 ACK

    Max-Forwards: 70

    Contact: <sip:45@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [9] 2011/06/10 17:20:21: Resolve 903: tcp 192.168.1.13 50228

    [6] 2011/06/10 17:20:21: Response to 192.168.1.13:50228 must be sent over existing connection

    [9] 2011/06/10 17:20:21: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:50230;branch=z9hG4bK-ce635a49a2a692a1bfee46cad8bc6e27;rport=50230

    Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

    From: "Forty Five"<sip:45@pbx.company.com;user=phone>;tag=31523

    Call-ID: 37a2fda3@pbx

    CSeq: 27729 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4000 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [8] 2011/06/10 17:20:21: Call 37a2fda3@pbx: Response does not correspond to open request

    [9] 2011/06/10 17:20:21: Resolve 904: url sip:12012181444@192.168.1.12:5060;user=phone

    [9] 2011/06/10 17:20:21: Resolve 904: udp 192.168.1.12 5060

    [9] 2011/06/10 17:20:21: SIP Tx udp:192.168.1.12:5060:

    ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-ca096733c6c524b041e5e2755d954e74;rport

    From: "Forty Five" <sip:45@pbx.company.com;user=phone>;tag=31523

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

    Call-ID: 37a2fda3@pbx

    CSeq: 27729 ACK

    Max-Forwards: 70

    Contact: <sip:45@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [9] 2011/06/10 17:20:22: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:50230;branch=z9hG4bK-ce635a49a2a692a1bfee46cad8bc6e27;rport=50230

    Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

    From: "Forty Five"<sip:45@pbx.company.com;user=phone>;tag=31523

    Call-ID: 37a2fda3@pbx

    CSeq: 27729 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4000 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [8] 2011/06/10 17:20:22: Call 37a2fda3@pbx: Response does not correspond to open request

    [9] 2011/06/10 17:20:22: Resolve 905: url sip:12012181444@192.168.1.12:5060;user=phone

    [9] 2011/06/10 17:20:22: Resolve 905: udp 192.168.1.12 5060

    [9] 2011/06/10 17:20:22: SIP Tx udp:192.168.1.12:5060:

    ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-ca096733c6c524b041e5e2755d954e74;rport

    From: "Forty Five" <sip:45@pbx.company.com;user=phone>;tag=31523

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

    Call-ID: 37a2fda3@pbx

    CSeq: 27729 ACK

    Max-Forwards: 70

    Contact: <sip:45@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

    Content-Length: 0

     

     

    [9] 2011/06/10 17:20:24: SIP Rx tcp:192.168.1.12:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 192.168.1.13:50230;branch=z9hG4bK-ce635a49a2a692a1bfee46cad8bc6e27;rport=50230

    Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

    From: "Forty Five"<sip:45@pbx.company.com;user=phone>;tag=31523

    Call-ID: 37a2fda3@pbx

    CSeq: 27729 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

    Content-Type: application/sdp

    Supported: replaces, norefersub

    User-Agent: HG4000/1.0

    Content-Length: 189

     

    v=0

    o=HG4000 0 0 IN IP4 192.168.1.12

    s=HG4000-Session

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 4000 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [8] 2011/06/10 17:20:24: Call 37a2fda3@pbx: Response does not correspond to open request

    [9] 2011/06/10 17:20:24: Resolve 906: url sip:12012181444@192.168.1.12:5060;user=phone

    [9] 2011/06/10 17:20:24: Resolve 906: udp 192.168.1.12 5060

    [9] 2011/06/10 17:20:24: SIP Tx udp:192.168.1.12:5060:

    ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-ca096733c6c524b041e5e2755d954e74;rport

    From: "Forty Five" <sip:45@pbx.company.com;user=phone>;tag=31523

    To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

    Call-ID: 37a2fda3@pbx

    CSeq: 27729 ACK

    Max-Forwards: 70

    Contact: <sip:45@192.168.1.13:5060;transport=udp>

    P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

    Content-Length: 0

  17. speech server 2007 - snom on - outbound call - authenticatin isue

    in Trunk Setup

    Posted

    I am trying to place an outgoing call to PSTN from Microsoft Speech server 2007 through SNOM One PBX. But SNOM One is asking for authentication

     

     

    My set up is as follows

     

    1. speech server - Created a trusted peer to snom one pbx at localhost:5060 (speech server is listening on 6060)

    2. pbxnsip - created a trunk (type - sip proxy) to speech server - Assume call comes from - extension 42.

    3. PBXnSIP - created a trunk (type - sip gateway) to Hypermedia Gateway.

    4. PBXnSIP - Created a dialplan "Hypermedia-Dialplan" and set Trunk for the dial plan to Hypermedia (trunk created above)

    5. PBXnSIP - For extension 42 , dial plan is set to hyprmedia-dialplan

    6. When I place an outgoing call from speech server, I can see that the sip request is reachig the hypermedia gateway. But Speech server is complaining as follows

     

    An error occurred during call transfer: Microsoft.SpeechServer.SipPeerException: A SIP request has failed. The current operation is 'Opening'. The session state is 'Connecting'. The remote participant is 'sip:2012181444@192.168.1.13:5060;user=phone'. The response code was '401'. The response text was 'Authentication Required'. ---> SupportedAuthenticationProtocols=None

    Realm=

    FailureReason=None

    ErrorCode=0

    ResponseCode=401 ResponseText=Authentication Required

    Microsoft.Rtc.Signaling.AuthenticationException: Peer to peer endpoint does not support authentication.

    at Microsoft.Rtc.Signaling.SipAsyncResult.ThrowIfFailed()

    at Microsoft.Rtc.Signaling.Helper.EndAsyncOperation[T](Object owner, IAsyncResult asyncResult)

    at Microsoft.Rtc.Signaling.Helper.EndAsyncOperation[T](Object owner, IAsyncResult asyncResult, String operationId)

    at Microsoft.Rtc.Signaling.SignalingSession.EndEnter(SipInviteAsyncResultWrapper asyncWrapper)

    at Microsoft.SpeechServer.Core.TelephonySessionOutbound.ParticipateCallback(IAsyncResult result)

     

    5. In Snomone log I see the following...

     

     

    [8] 2011/06/10 10:38:51: Received SIP connection 255 from 192.168.1.13:53040

    [9] 2011/06/10 10:38:51: SIP Rx tcp:192.168.1.13:53040:

    INVITE sip:2012181444@192.168.1.13:5060;user=phone SIP/2.0

    FROM: <sip:45@CommServer.creditfree.local:50681;user=phone>;epid=457BE1388F;tag=8b9960a8ed

    TO: <sip:2012181444@192.168.1.13:5060;user=phone>

    CSEQ: 1 INVITE

    CALL-ID: 79644d90-3c04-4839-98a7-729c9b9a6b93

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 192.168.1.13:53040;branch=z9hG4bK641e3c71

    CONTACT: <sip:CommServer.creditfree.local:50681;transport=Tcp;maddr=192.168.1.13;ms-opaque=b2dfee4fee6e4834>;automata

    CONTENT-LENGTH: 335

    USER-AGENT: RTCC/3.0.0.0

    CONTENT-TYPE: application/sdp

    ALLOW: UPDATE

    ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify

     

    v=0

    o=- 0 0 IN IP4 192.168.1.13

    s=Microsoft Speech Server session

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 6274 RTP/AVP 114 115 4 0 8 97 101

    a=rtpmap:114 x-msrta/16000

    a=fmtp:114 bitrate=29000

    a=rtpmap:115 x-msrta/8000

    a=fmtp:115 bitrate=11800

    a=rtpmap:97 RED/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

     

    [9] 2011/06/10 10:38:51: SIP Tx tcp:192.168.1.13:53040:

    SIP/2.0 100 Trying

    Via: SIP/2.0/TCP 192.168.1.13:53040;branch=z9hG4bK641e3c71

    From: <sip:45@CommServer.creditfree.local:50681;user=phone>;tag=8b9960a8ed;epid=457BE1388F

    To: <sip:2012181444@192.168.1.13:5060;user=phone>;tag=ea32e7f2fd

    Call-ID: 79644d90-3c04-4839-98a7-729c9b9a6b93

    CSeq: 1 INVITE

    Content-Length: 0

     

     

    [9] 2011/06/10 10:38:51: SIP Tx tcp:192.168.1.13:53040:

    SIP/2.0 401 Authentication RequiredVia: SIP/2.0/TCP 192.168.1.13:53040;branch=z9hG4bK641e3c71

    From: <sip:45@CommServer.creditfree.local:50681;user=phone>;tag=8b9960a8ed;epid=457BE1388F

    To: <sip:2012181444@192.168.1.13:5060;user=phone>;tag=ea32e7f2fd

    Call-ID: 79644d90-3c04-4839-98a7-729c9b9a6b93

    CSeq: 1 INVITE

    User-Agent: snom-PBX/4.2.0.3950

    WWW-Authenticate: Digest realm="commserver.creditfree.local",nonce="d2e7e6915156a03a81283494f153136a",domain="sip:2012181444@192.168.1.13:5060;user=phone",algorithm=MD5

    Content-Length: 0

  18. MSS07 Outbound Calls Works But...

    in Microsoft Speech Server

    Posted

    My issue is that for all outbound calls, SnomeOne PBX is requiring authentication.Any idea how to fix this?

     

    My set up is as follows

     

    1. speech server - Created a trusted peer to snomone pbx at localhost:5060 (speech server is listening on 6060)

    2. pbxnsip - craeted a trunk (type - sip proxy) to speech server - Assume call comes from - extension 42.

    3. 42 extension dial plan is set to hyprmedia, and the trunk for the dial plan is set to hypermedia gateway.

    4. When I place an outgoing call from speech server, I can sed that the sip request is reachig the gateway. But Speech server is complaining as follows

     

    An error occurred during call transfer: Microsoft.SpeechServer.SipPeerException: A SIP request has failed. The current operation is 'Opening'. The session state is 'Connecting'. The remote participant is 'sip:2012181444@192.168.1.13:5060;user=phone'. The response code was '401'. The response text was 'Authentication Required'. ---> SupportedAuthenticationProtocols=None

    Realm=

    FailureReason=None

    ErrorCode=0

    ResponseCode=401 ResponseText=Authentication Required

    Microsoft.Rtc.Signaling.AuthenticationException: Peer to peer endpoint does not support authentication.

    at Microsoft.Rtc.Signaling.SipAsyncResult.ThrowIfFailed()

    at Microsoft.Rtc.Signaling.Helper.EndAsyncOperation[T](Object owner, IAsyncResult asyncResult)

    at Microsoft.Rtc.Signaling.Helper.EndAsyncOperation[T](Object owner, IAsyncResult asyncResult, String operationId)

    at Microsoft.Rtc.Signaling.SignalingSession.EndEnter(SipInviteAsyncResultWrapper asyncWrapper)

    at Microsoft.SpeechServer.Core.TelephonySessionOutbound.ParticipateCallback(IAsyncResult result)

     

    5. In Snomone log I see the following...

     

     

    [8] 2011/06/10 10:38:51: Received SIP connection 255 from 192.168.1.13:53040

    [9] 2011/06/10 10:38:51: SIP Rx tcp:192.168.1.13:53040:

    INVITE sip:2012181444@192.168.1.13:5060;user=phone SIP/2.0

    FROM: <sip:45@CommServer.creditfree.local:50681;user=phone>;epid=457BE1388F;tag=8b9960a8ed

    TO: <sip:2012181444@192.168.1.13:5060;user=phone>

    CSEQ: 1 INVITE

    CALL-ID: 79644d90-3c04-4839-98a7-729c9b9a6b93

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 192.168.1.13:53040;branch=z9hG4bK641e3c71

    CONTACT: <sip:CommServer.creditfree.local:50681;transport=Tcp;maddr=192.168.1.13;ms-opaque=b2dfee4fee6e4834>;automata

    CONTENT-LENGTH: 335

    USER-AGENT: RTCC/3.0.0.0

    CONTENT-TYPE: application/sdp

    ALLOW: UPDATE

    ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify

     

    v=0

    o=- 0 0 IN IP4 192.168.1.13

    s=Microsoft Speech Server session

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 6274 RTP/AVP 114 115 4 0 8 97 101

    a=rtpmap:114 x-msrta/16000

    a=fmtp:114 bitrate=29000

    a=rtpmap:115 x-msrta/8000

    a=fmtp:115 bitrate=11800

    a=rtpmap:97 RED/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

     

    [9] 2011/06/10 10:38:51: SIP Tx tcp:192.168.1.13:53040:

    SIP/2.0 100 Trying

    Via: SIP/2.0/TCP 192.168.1.13:53040;branch=z9hG4bK641e3c71

    From: <sip:45@CommServer.creditfree.local:50681;user=phone>;tag=8b9960a8ed;epid=457BE1388F

    To: <sip:2012181444@192.168.1.13:5060;user=phone>;tag=ea32e7f2fd

    Call-ID: 79644d90-3c04-4839-98a7-729c9b9a6b93

    CSeq: 1 INVITE

    Content-Length: 0

     

     

    [9] 2011/06/10 10:38:51: SIP Tx tcp:192.168.1.13:53040:

    SIP/2.0 401 Authentication Required

    Via: SIP/2.0/TCP 192.168.1.13:53040;branch=z9hG4bK641e3c71

    From: <sip:45@CommServer.creditfree.local:50681;user=phone>;tag=8b9960a8ed;epid=457BE1388F

    To: <sip:2012181444@192.168.1.13:5060;user=phone>;tag=ea32e7f2fd

    Call-ID: 79644d90-3c04-4839-98a7-729c9b9a6b93

    CSeq: 1 INVITE

    User-Agent: snom-PBX/4.2.0.3950

    WWW-Authenticate: Digest realm="commserver.creditfree.local",nonce="d2e7e6915156a03a81283494f153136a",domain="sip:2012181444@192.168.1.13:5060;user=phone",algorithm=MD5

    Content-Length: 0

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