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joso

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  1. voicemail times out

    in General Setup

    Posted

    I only have one firewall and ip at the moment. I have the log file, not sure if it will help:

     

    Also, what does this mean "SIP TCP/TLS timeout on 192.168.1.2:42354, closing connection "

     

    5] 2008/09/18 15:11:12: SIP port accept from 192.168.1.2:42393

    [7] 2008/09/18 15:11:21: SIP Rx udp:192.168.1.6:5060:

    INVITE sip:301@192.168.1.5;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.6;branch=z9hG4bKac2019376203

    Max-Forwards: 70

    From: "Subnet" <sip:4032708885@192.168.1.6>;tag=1c2019371919

    To: <sip:301@192.168.1.5;user=phone>

    Call-ID: 201937152741200005929@192.168.1.6

    CSeq: 1 INVITE

    Contact: <sip:4032708885@192.168.1.6>

    Supported: em,100rel,timer,replaces,path,early-session,resource-priority

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

    User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.5.20A.027.004

    Content-Type: application/sdp

    Content-Disposition: session

    Content-Length: 287

     

    v=0

    o=AudiocodesGW 2019365073 2019364953 IN IP4 192.168.1.6

    s=Phone-Call

    c=IN IP4 192.168.1.6

    t=0 0

    m=audio 6000 RTP/AVP 8 0 101

    a=rtpmap:8 PCMA/8000

    a=rtpmap:0 PCMU/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

    a=ptime:20

    a=sendrecv

    a=rtcp:6001 IN IP4 192.168.1.6

     

    [7] 2008/09/18 15:11:21: UDP: Opening socket on port 50166

    [7] 2008/09/18 15:11:21: UDP: Opening socket on port 50167

    [5] 2008/09/18 15:11:21: Identify trunk (domain name match) 2

    [7] 2008/09/18 15:11:21: SIP Tx udp:192.168.1.6:5060:

    SIP/2.0 100 Trying

    Via: SIP/2.0/UDP 192.168.1.6;branch=z9hG4bKac2019376203

    From: "Subnet" <sip:4032708885@192.168.1.6>;tag=1c2019371919

    To: <sip:301@192.168.1.5;user=phone>;tag=ebaf4451bd

    Call-ID: 201937152741200005929@192.168.1.6

    CSeq: 1 INVITE

    Content-Length: 0

     

     

    [7] 2008/09/18 15:11:21: Set packet length to 20

    [6] 2008/09/18 15:11:21: Sending RTP for 201937152741200005929@192.168.1.6#ebaf4451bd to 192.168.1.6:6000

    [5] 2008/09/18 15:11:21: Trunk AudioCodes sends call to 301

    [7] 2008/09/18 15:11:21: Set packet length to 20

    [7] 2008/09/18 15:11:21: SIP Tx udp:192.168.1.6:5060:

    SIP/2.0 200 Ok

    Via: SIP/2.0/UDP 192.168.1.6;branch=z9hG4bKac2019376203

    From: "Subnet" <sip:4032708885@192.168.1.6>;tag=1c2019371919

    To: <sip:301@192.168.1.5;user=phone>;tag=ebaf4451bd

    Call-ID: 201937152741200005929@192.168.1.6

    CSeq: 1 INVITE

    Contact: <sip:josog@192.168.1.5:5060;transport=udp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/3.0.0.2998

    Content-Type: application/sdp

    Content-Length: 226

     

    v=0

    o=- 62207 62207 IN IP4 192.168.1.5

    s=-

    c=IN IP4 192.168.1.5

    t=0 0

    m=audio 50166 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=sendrecv

     

    [7] 2008/09/18 15:11:21: Last message repeated 2 times

    [7] 2008/09/18 15:11:21: SIP Rx udp:192.168.1.6:5060:

    ACK sip:josog@192.168.1.5:5060;transport=udp SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.6;branch=z9hG4bKac2019562213

    Max-Forwards: 70

    From: "Subnet" <sip:4032708885@192.168.1.6>;tag=1c2019371919

    To: <sip:301@192.168.1.5;user=phone>;tag=ebaf4451bd

    Call-ID: 201937152741200005929@192.168.1.6

    CSeq: 1 ACK

    Contact: <sip:4032708885@192.168.1.6>

    Supported: em,timer,replaces,path,early-session,resource-priority

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

    User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.5.20A.027.004

    Content-Length: 0

     

     

    [7] 2008/09/18 15:11:29: Last message repeated 2 times

    [6] 2008/09/18 15:11:29: Received DTMF 3

    [6] 2008/09/18 15:11:29: Received DTMF 2

    [6] 2008/09/18 15:11:29: Received DTMF 5

    [7] 2008/09/18 15:11:32: Calling extension 325

    [7] 2008/09/18 15:11:32: UDP: Opening socket on port 60722

    [7] 2008/09/18 15:11:32: UDP: Opening socket on port 60723

    [5] 2008/09/18 15:11:32: Dialplan jngconsulting: Match 4037105450@localhost to <sip:4037105450@192.168.1.6;user=phone> on trunk AudioCodes

    [5] 2008/09/18 15:11:32: Using "Subnet" <sip:4032708885@localhost;user=phone> as redirect from

    [5] 2008/09/18 15:11:32: Charge user 325 for redirecting calls

    [7] 2008/09/18 15:11:32: SIP Tx tcp:192.168.1.4:5060:

    INVITE sip:+4036707140325@ssi-ocsmed.jngconsulting.com;transport=tcp SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.5:4638;branch=z9hG4bK-433167005897384ba08f53293f0ca7ca;rport

    From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=4673

    To: "Joso Grivicic" <sip:325@localhost>

    Call-ID: 82331c2d@pbx

    CSeq: 12820 INVITE

    Max-Forwards: 70

    Contact: <sip:325@192.168.1.5:4638;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/3.0.0.2998

    Alert-Info: <http://127.0.0.1/Bellcore-dr3>

    Content-Type: application/sdp

    Content-Length: 335

     

    v=0

    o=- 60462 60462 IN IP4 192.168.1.5

    s=-

    c=IN IP4 192.168.1.5

    t=0 0

    m=audio 60722 RTP/AVP 0 8 9 18 2 3 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:9 g722/8000

    a=rtpmap:18 g729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:2 g726-32/8000

    a=rtpmap:3 gsm/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=sendrecv

     

    [7] 2008/09/18 15:11:32: UDP: Opening socket on port 55700

    [7] 2008/09/18 15:11:32: UDP: Opening socket on port 55701

    [7] 2008/09/18 15:11:32: SIP Tx udp:192.168.1.22:5060:

    INVITE sip:325@192.168.1.22 SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-09918fb1a67eb8dd875c7cbbf513f878;rport

    From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=62862

    To: "Joso Grivicic" <sip:325@localhost>

    Call-ID: b016bc51@pbx

    CSeq: 4524 INVITE

    Max-Forwards: 70

    Contact: <sip:325@192.168.1.5:5060;transport=udp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/3.0.0.2998

    Alert-Info: <http://127.0.0.1/Bellcore-dr3>

    Content-Type: application/sdp

    Content-Length: 335

     

    v=0

    o=- 23309 23309 IN IP4 192.168.1.5

    s=-

    c=IN IP4 192.168.1.5

    t=0 0

    m=audio 55700 RTP/AVP 0 8 9 18 2 3 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:9 g722/8000

    a=rtpmap:18 g729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:2 g726-32/8000

    a=rtpmap:3 gsm/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=sendrecv

     

    [7] 2008/09/18 15:11:32: UDP: Opening socket on port 51064

    [7] 2008/09/18 15:11:32: UDP: Opening socket on port 51065

    [7] 2008/09/18 15:11:32: SIP Tx udp:192.168.1.6:5060:

    INVITE sip:4037105450@192.168.1.6;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-1faedd9e7819005bb387250f371f51f2;rport

    From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=65392

    To: <sip:4037105450@192.168.1.6;user=phone>

    Call-ID: 5f33b786@pbx

    CSeq: 14343 INVITE

    Max-Forwards: 70

    Contact: <sip:josog@192.168.1.5:5060;transport=udp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/3.0.0.2998

    P-Asserted-Identity: <sip:josog@192.168.1.6;user=phone>

    Content-Type: application/sdp

    Content-Length: 335

     

    v=0

    o=- 44121 44121 IN IP4 192.168.1.5

    s=-

    c=IN IP4 192.168.1.5

    t=0 0

    m=audio 51064 RTP/AVP 0 8 9 18 2 3 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:9 g722/8000

    a=rtpmap:18 g729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:2 g726-32/8000

    a=rtpmap:3 gsm/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=sendrecv

     

    [7] 2008/09/18 15:11:32: SIP Rx tcp:192.168.1.4:5060:

    SIP/2.0 100 Trying

    FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=4673

    TO: "Joso Grivicic"<sip:325@localhost>

    CSEQ: 12820 INVITE

    CALL-ID: 82331c2d@pbx

    VIA: SIP/2.0/TCP 192.168.1.5:4638;branch=z9hG4bK-433167005897384ba08f53293f0ca7ca;rport

    CONTENT-LENGTH: 0

     

     

    [7] 2008/09/18 15:11:32: SIP Rx udp:192.168.1.6:5060:

    SIP/2.0 404 Not Found

    Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-1faedd9e7819005bb387250f371f51f2;rport

    From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=65392

    To: <sip:4037105450@192.168.1.6;user=phone>;tag=1c2046102598

    Call-ID: 5f33b786@pbx

    CSeq: 14343 INVITE

    Supported: em,timer,replaces,path,resource-priority

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

    Server: Audiocodes-Sip-Gateway-MP-118 FXO/v.5.20A.027.004

    Reason: Q.850 ;cause=3

    Content-Length: 0

     

     

    [7] 2008/09/18 15:11:32: Call 5f33b786@pbx#65392: Clear last INVITE

    [7] 2008/09/18 15:11:32: SIP Tx udp:192.168.1.6:5060:

    ACK sip:4037105450@192.168.1.6;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-1faedd9e7819005bb387250f371f51f2;rport

    From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=65392

    To: <sip:4037105450@192.168.1.6;user=phone>;tag=1c2046102598

    Call-ID: 5f33b786@pbx

    CSeq: 14343 ACK

    Max-Forwards: 70

    Contact: <sip:josog@192.168.1.5:5060;transport=udp>

    P-Asserted-Identity: <sip:josog@192.168.1.6;user=phone>

    Content-Length: 0

     

     

    [5] 2008/09/18 15:11:32: INVITE Response: Terminate 5f33b786@pbx

    [7] 2008/09/18 15:11:32: Other Ports: 3

    [7] 2008/09/18 15:11:32: Call Port: 201937152741200005929@192.168.1.6#ebaf4451bd

    [7] 2008/09/18 15:11:32: Call Port: 82331c2d@pbx#4673

    [7] 2008/09/18 15:11:32: Call Port: b016bc51@pbx#62862

    [7] 2008/09/18 15:11:32: SIP Rx udp:192.168.1.22:5060:

    SIP/2.0 100 Trying

    Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-09918fb1a67eb8dd875c7cbbf513f878;rport

    From:"Subnet" <sip:4032708885@localhost;user=phone>;tag=62862

    To:"Joso Grivicic" <sip:325@localhost>;tag=48d2379a-220-6359b15a

    CSeq:4524 INVITE

    User-Agent:Mitel-5220-SIP-Phone 03.00.00.01 08000F178504

    Call-ID:b016bc51@pbx

    Content-Length:0

     

     

    [7] 2008/09/18 15:11:32: SIP Rx udp:192.168.1.22:5060:

    SIP/2.0 180 Ringing

    Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-09918fb1a67eb8dd875c7cbbf513f878;rport

    From:"Subnet" <sip:4032708885@localhost;user=phone>;tag=62862

    To:"Joso Grivicic" <sip:325@localhost>;tag=48d2379a-220-6359b15a

    CSeq:4524 INVITE

    User-Agent:Mitel-5220-SIP-Phone 03.00.00.01 08000F178504

    Call-ID:b016bc51@pbx

    Content-Length:0

     

     

    [7] 2008/09/18 15:11:32: SIP Rx tcp:192.168.1.4:5060:

    SIP/2.0 183 Session Progress

    FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=4673

    TO: Joso Grivicic<sip:325@localhost>;epid=0FDC8A2A83;tag=49601a2ec

    CSEQ: 12820 INVITE

    CALL-ID: 82331c2d@pbx

    VIA: SIP/2.0/TCP 192.168.1.5:4638;branch=z9hG4bK-433167005897384ba08f53293f0ca7ca;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0 MediationServer

     

     

    [7] 2008/09/18 15:11:32: SIP Rx tcp:192.168.1.4:5060:

    SIP/2.0 180 Ringing

    FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=4673

    TO: Joso Grivicic<sip:325@localhost>;epid=0FDC8A2A83;tag=49601a2ec

    CSEQ: 12820 INVITE

    CALL-ID: 82331c2d@pbx

    VIA: SIP/2.0/TCP 192.168.1.5:4638;branch=z9hG4bK-433167005897384ba08f53293f0ca7ca;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0 MediationServer

     

     

    [7] 2008/09/18 15:11:42: SIP Tx tcp:192.168.1.4:5060:

    CANCEL sip:+4036707140325@ssi-ocsmed.jngconsulting.com;transport=tcp SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.5:4638;branch=z9hG4bK-433167005897384ba08f53293f0ca7ca;rport

    From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=4673

    To: "Joso Grivicic" <sip:325@localhost>

    Call-ID: 82331c2d@pbx

    CSeq: 12820 CANCEL

    Max-Forwards: 70

    Content-Length: 0

     

     

    [6] 2008/09/18 15:11:42: Redirecting to external voicemail account 325 destination sip:7325@localhost

    [5] 2008/09/18 15:11:42: Dialplan jngconsulting: Match 7325@localhost to <sip:325@192.168.1.2;user=phone> on trunk Exchange

    [5] 2008/09/18 15:11:42: Using "Subnet" <sip:4032708885@localhost;user=phone> as redirect from

    [7] 2008/09/18 15:11:42: SIP Tx udp:192.168.1.22:5060:

    CANCEL sip:325@192.168.1.22 SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-09918fb1a67eb8dd875c7cbbf513f878;rport

    From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=62862

    To: "Joso Grivicic" <sip:325@localhost>

    Call-ID: b016bc51@pbx

    CSeq: 4524 CANCEL

    Max-Forwards: 70

    Content-Length: 0

     

     

    [5] 2008/09/18 15:11:42: Charge user 325 for redirecting calls

    [7] 2008/09/18 15:11:42: UDP: Opening socket on port 57840

    [7] 2008/09/18 15:11:42: UDP: Opening socket on port 57841

    [7] 2008/09/18 15:11:42: SIP Tx tcp:192.168.1.2:5060:

    INVITE sip:325@192.168.1.2;user=phone SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.5:4649;branch=z9hG4bK-abcbde6b048db506b99c96a6ac260f49;rport

    From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=54625

    To: <sip:325@192.168.1.2;user=phone>

    Call-ID: 26ed22a4@pbx

    CSeq: 19637 INVITE

    Max-Forwards: 70

    Contact: <sip:4032708885@192.168.1.5:4649;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/3.0.0.2998

    Diversion: <tel:325>;reason=no-answer;screen=no;privacy=off

    P-Asserted-Identity: "Joso Grivicic" <sip:325@192.168.1.2;user=phone>

    Content-Type: application/sdp

    Content-Length: 335

     

    v=0

    o=- 11803 11803 IN IP4 192.168.1.5

    s=-

    c=IN IP4 192.168.1.5

    t=0 0

    m=audio 57840 RTP/AVP 0 8 9 18 2 3 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:9 g722/8000

    a=rtpmap:18 g729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:2 g726-32/8000

    a=rtpmap:3 gsm/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=sendrecv

     

    [7] 2008/09/18 15:11:42: SIP Rx tcp:192.168.1.2:5060:

    SIP/2.0 100 Trying

    FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=54625

    TO: <sip:325@192.168.1.2;user=phone>

    CSEQ: 19637 INVITE

    CALL-ID: 26ed22a4@pbx

    VIA: SIP/2.0/TCP 192.168.1.5:4649;branch=z9hG4bK-abcbde6b048db506b99c96a6ac260f49;rport

    CONTENT-LENGTH: 0

     

     

    [7] 2008/09/18 15:11:42: SIP Rx udp:192.168.1.22:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-09918fb1a67eb8dd875c7cbbf513f878;rport

    From:"Subnet" <sip:4032708885@localhost;user=phone>;tag=62862

    To:"Joso Grivicic" <sip:325@localhost>

    CSeq:4524 CANCEL

    User-Agent:Mitel-5220-SIP-Phone 03.00.00.01 08000F178504

    Call-ID:b016bc51@pbx

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

    Content-Length:0

     

     

    [7] 2008/09/18 15:11:42: Call b016bc51@pbx#62862: Clear last request

    [7] 2008/09/18 15:11:42: SIP Rx udp:192.168.1.22:5060:

    SIP/2.0 487 Request Cancelled

    Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-09918fb1a67eb8dd875c7cbbf513f878;rport

    From:"Subnet" <sip:4032708885@localhost;user=phone>;tag=62862

    To:"Joso Grivicic" <sip:325@localhost>;tag=48d2379a-220-6359b15a

    CSeq:4524 INVITE

    User-Agent:Mitel-5220-SIP-Phone 03.00.00.01 08000F178504

    Call-ID:b016bc51@pbx

    Content-Length:0

     

     

    [7] 2008/09/18 15:11:42: Call b016bc51@pbx#62862: Clear last INVITE

    [7] 2008/09/18 15:11:42: SIP Tx udp:192.168.1.22:5060:

    ACK sip:325@192.168.1.22 SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-09918fb1a67eb8dd875c7cbbf513f878;rport

    From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=62862

    To: "Joso Grivicic" <sip:325@localhost>;tag=48d2379a-220-6359b15a

    Call-ID: b016bc51@pbx

    CSeq: 4524 ACK

    Max-Forwards: 70

    Contact: <sip:325@192.168.1.5:5060;transport=udp>

    Content-Length: 0

     

     

    [5] 2008/09/18 15:11:42: INVITE Response: Terminate b016bc51@pbx

    [7] 2008/09/18 15:11:42: Other Ports: 3

    [7] 2008/09/18 15:11:42: Call Port: 201937152741200005929@192.168.1.6#ebaf4451bd

    [7] 2008/09/18 15:11:42: Call Port: 26ed22a4@pbx#54625

    [7] 2008/09/18 15:11:42: Call Port: 82331c2d@pbx#4673

    [7] 2008/09/18 15:11:42: SIP Rx tcp:192.168.1.4:5060:

    SIP/2.0 487 Request Terminated

    FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=4673

    TO: Joso Grivicic<sip:325@localhost>;epid=0FDC8A2A83;tag=49601a2ec

    CSEQ: 12820 INVITE

    CALL-ID: 82331c2d@pbx

    VIA: SIP/2.0/TCP 192.168.1.5:4638;branch=z9hG4bK-433167005897384ba08f53293f0ca7ca;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0 MediationServer

     

     

    [7] 2008/09/18 15:11:42: Call 82331c2d@pbx#4673: Clear last INVITE

    [7] 2008/09/18 15:11:42: SIP Tx tcp:192.168.1.4:5060:

    ACK sip:+4036707140325@ssi-ocsmed.jngconsulting.com;transport=tcp SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.5:4638;branch=z9hG4bK-433167005897384ba08f53293f0ca7ca;rport

    From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=4673

    To: "Joso Grivicic" <sip:325@localhost>;tag=49601a2ec

    Call-ID: 82331c2d@pbx

    CSeq: 12820 ACK

    Max-Forwards: 70

    Contact: <sip:325@192.168.1.5:4638;transport=tcp>

    Content-Length: 0

     

     

    [5] 2008/09/18 15:11:42: INVITE Response: Terminate 82331c2d@pbx

    [7] 2008/09/18 15:11:42: Other Ports: 2

    [7] 2008/09/18 15:11:42: Call Port: 201937152741200005929@192.168.1.6#ebaf4451bd

    [7] 2008/09/18 15:11:42: Call Port: 26ed22a4@pbx#54625

    [7] 2008/09/18 15:11:42: SIP Rx tcp:192.168.1.4:5060:

    SIP/2.0 200 OK

    FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=4673

    TO: Joso Grivicic<sip:325@localhost>;epid=0FDC8A2A83;tag=bd82658dfc

    CSEQ: 12820 CANCEL

    CALL-ID: 82331c2d@pbx

    VIA: SIP/2.0/TCP 192.168.1.5:4638;branch=z9hG4bK-433167005897384ba08f53293f0ca7ca;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0 MediationServer

     

     

    [7] 2008/09/18 15:11:42: SIP Rx tcp:192.168.1.2:5060:

    SIP/2.0 302 Moved Temporarily

    FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=54625

    TO: <sip:325@192.168.1.2;user=phone>;tag=26226ab74c

    CSEQ: 19637 INVITE

    CALL-ID: 26ed22a4@pbx

    VIA: SIP/2.0/TCP 192.168.1.5:4649;branch=z9hG4bK-abcbde6b048db506b99c96a6ac260f49;rport

    CONTACT: <sip:325@192.168.1.2:5065;user=phone;transport=TCP>

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0

    Diversion: <tel:325>;reason=no-answer;screen=no;privacy=off

     

     

    [7] 2008/09/18 15:11:42: Call 26ed22a4@pbx#54625: Clear last INVITE

    [7] 2008/09/18 15:11:42: SIP Tx tcp:192.168.1.2:5060:

    ACK sip:325@192.168.1.2;user=phone SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.5:4649;branch=z9hG4bK-abcbde6b048db506b99c96a6ac260f49;rport

    From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=54625

    To: <sip:325@192.168.1.2;user=phone>;tag=26226ab74c

    Call-ID: 26ed22a4@pbx

    CSeq: 19637 ACK

    Max-Forwards: 70

    Contact: <sip:4032708885@192.168.1.5:4649;transport=tcp>

    P-Asserted-Identity: "Joso Grivicic" <sip:325@192.168.1.2;user=phone>

    Content-Length: 0

     

     

    [5] 2008/09/18 15:11:42: Redirecting call

    [7] 2008/09/18 15:11:42: SIP Tx tcp:192.168.1.2:5065:

    INVITE sip:325@192.168.1.2:5065;user=phone;transport=TCP SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.5:4650;branch=z9hG4bK-9ba3285162ecc8be7a001b93c1904328;rport

    From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=54625

    To: <sip:325@192.168.1.2;user=phone>

    Call-ID: 26ed22a4@pbx

    CSeq: 19638 INVITE

    Max-Forwards: 70

    Contact: <sip:4032708885@192.168.1.5:4650;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/3.0.0.2998

    Diversion: <tel:325>;reason=no-answer;screen=no;privacy=off

    P-Asserted-Identity: "Joso Grivicic" <sip:325@192.168.1.2;user=phone>

    Content-Type: application/sdp

    Content-Length: 335

     

    v=0

    o=- 11803 11803 IN IP4 192.168.1.5

    s=-

    c=IN IP4 192.168.1.5

    t=0 0

    m=audio 57840 RTP/AVP 0 8 9 18 2 3 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:9 g722/8000

    a=rtpmap:18 g729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:2 g726-32/8000

    a=rtpmap:3 gsm/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=sendrecv

     

    [7] 2008/09/18 15:11:42: SIP Rx tcp:192.168.1.2:5065:

    SIP/2.0 100 Trying

    FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=54625

    TO: <sip:325@192.168.1.2;user=phone>

    CSEQ: 19638 INVITE

    CALL-ID: 26ed22a4@pbx

    VIA: SIP/2.0/TCP 192.168.1.5:4650;branch=z9hG4bK-9ba3285162ecc8be7a001b93c1904328;rport

    CONTENT-LENGTH: 0

     

     

    [7] 2008/09/18 15:11:42: SIP Rx tcp:192.168.1.2:5065:

    SIP/2.0 180 Ringing

    FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=54625

    TO: <sip:325@192.168.1.2;user=phone>;epid=7EF0970BA2;tag=6b14a423e3

    CSEQ: 19638 INVITE

    CALL-ID: 26ed22a4@pbx

    VIA: SIP/2.0/TCP 192.168.1.5:4650;branch=z9hG4bK-9ba3285162ecc8be7a001b93c1904328;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0

     

     

    [7] 2008/09/18 15:11:42: SIP Rx tcp:192.168.1.2:5065:

    SIP/2.0 200 OK

    FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=54625

    TO: <sip:325@192.168.1.2;user=phone>;epid=7EF0970BA2;tag=6b14a423e3

    CSEQ: 19638 INVITE

    CALL-ID: 26ed22a4@pbx

    VIA: SIP/2.0/TCP 192.168.1.5:4650;branch=z9hG4bK-9ba3285162ecc8be7a001b93c1904328;rport

    CONTACT: <sip:SSI-TESTDC.jngconsulting.com:5065;transport=Tcp;maddr=192.168.1.2>;automata

    CONTENT-LENGTH: 192

    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 192.168.1.2

    s=Microsoft Exchange Speech Engine

    c=IN IP4 192.168.1.2

    t=0 0

    m=audio 6272 RTP/AVP 0 8 101

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

     

    [7] 2008/09/18 15:11:42: Call 26ed22a4@pbx#54625: Clear last INVITE

    [7] 2008/09/18 15:11:42: Set packet length to 20

    [6] 2008/09/18 15:11:42: Sending RTP for 26ed22a4@pbx#54625 to 192.168.1.2:6272

    [7] 2008/09/18 15:11:42: SIP Tx tcp:192.168.1.2:5065:

    ACK sip:SSI-TESTDC.jngconsulting.com:5065;transport=Tcp;maddr=192.168.1.2 SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.5:4650;branch=z9hG4bK-9869b4983d009ddb2438347e95d0ccb0;rport

    From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=54625

    To: <sip:325@192.168.1.2;user=phone>;tag=6b14a423e3

    Call-ID: 26ed22a4@pbx

    CSeq: 19638 ACK

    Max-Forwards: 70

    Contact: <sip:4032708885@192.168.1.5:4650;transport=tcp>

    P-Asserted-Identity: "Joso Grivicic" <sip:325@192.168.1.2;user=phone>

    Content-Length: 0

     

     

    [7] 2008/09/18 15:11:42: Determine pass-through mode after receiving response

    [7] 2008/09/18 15:11:42: 26ed22a4@pbx#54625: RTP pass-through mode

    [7] 2008/09/18 15:11:42: 201937152741200005929@192.168.1.6#ebaf4451bd: RTP pass-through mode

    [7] 2008/09/18 15:12:02: SIP Rx udp:192.168.1.6:5060:

    BYE sip:josog@192.168.1.5:5060;transport=udp SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.6;branch=z9hG4bKac2122852159

    Max-Forwards: 70

    From: "Subnet" <sip:4032708885@192.168.1.6>;tag=1c2019371919

    To: <sip:301@192.168.1.5;user=phone>;tag=ebaf4451bd

    Call-ID: 201937152741200005929@192.168.1.6

    CSeq: 2 BYE

    Supported: em,timer,replaces,path,early-session,resource-priority

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

    User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.5.20A.027.004

    Reason: Q.850 ;cause=31 ;text="RTP Broken Connection"

    Content-Length: 0

     

     

    [7] 2008/09/18 15:12:02: SIP Tx udp:192.168.1.6:5060:

    SIP/2.0 200 Ok

    Via: SIP/2.0/UDP 192.168.1.6;branch=z9hG4bKac2122852159

    From: "Subnet" <sip:4032708885@192.168.1.6>;tag=1c2019371919

    To: <sip:301@192.168.1.5;user=phone>;tag=ebaf4451bd

    Call-ID: 201937152741200005929@192.168.1.6

    CSeq: 2 BYE

    Contact: <sip:josog@192.168.1.5:5060;transport=udp>

    User-Agent: pbxnsip-PBX/3.0.0.2998

    RTP-RxStat: Dur=41,Pkt=2064,Oct=350952,Underun=0

    RTP-TxStat: Dur=41,Pkt=1545,Oct=265740

    Content-Length: 0

     

     

    [7] 2008/09/18 15:12:02: 26ed22a4@pbx#54625: Media-aware pass-through mode

    [7] 2008/09/18 15:12:02: Other Ports: 1

    [7] 2008/09/18 15:12:02: Call Port: 26ed22a4@pbx#54625

    [7] 2008/09/18 15:12:02: SIP Tx tcp:192.168.1.2:5065:

    BYE sip:SSI-TESTDC.jngconsulting.com:5065;transport=Tcp;maddr=192.168.1.2 SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.5:4650;branch=z9hG4bK-b02d89319eec09d9258589db2e1c8e79;rport

    From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=54625

    To: <sip:325@192.168.1.2;user=phone>;tag=6b14a423e3

    Call-ID: 26ed22a4@pbx

    CSeq: 19639 BYE

    Max-Forwards: 70

    Contact: <sip:4032708885@192.168.1.5:4650;transport=tcp>

    RTP-RxStat: Dur=21,Pkt=495,Oct=85140,Underun=0

    RTP-TxStat: Dur=20,Pkt=1017,Oct=174924

    P-Asserted-Identity: "Joso Grivicic" <sip:325@192.168.1.2;user=phone>

    Content-Length: 0

     

     

    [7] 2008/09/18 15:12:02: SIP Rx tcp:192.168.1.2:5065:

    SIP/2.0 200 OK

    FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=54625

    TO: <sip:325@192.168.1.2;user=phone>;tag=6b14a423e3;epid=7EF0970BA2

    CSEQ: 19639 BYE

    CALL-ID: 26ed22a4@pbx

    VIA: SIP/2.0/TCP 192.168.1.5:4650;branch=z9hG4bK-b02d89319eec09d9258589db2e1c8e79;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0

     

     

    [7] 2008/09/18 15:12:02: Call 26ed22a4@pbx#54625: Clear last request

    [5] 2008/09/18 15:12:02: BYE Response: Terminate 26ed22a4@pbx

    [6] 2008/09/18 15:12:11: SIP TCP/TLS timeout on 192.168.1.2:42354, closing connection

    [5] 2008/09/18 15:13:12: SIP port accept from 192.168.1.2:30667

    [6] 2008/09/18 15:14:11: SIP TCP/TLS timeout on 192.168.1.2:42365, closing connection

    [5] 2008/09/18 15:15:12: SIP port accept from 192.168.1.2:30674

  3. Mitel SIP Phone

    in VoIP Phones

    Posted

    I remember testing Mitel phones some time ago (well, must be more than a year...). But anyway, it worked and I don't remember any significant problem. The only disadvantage I can think of is the PnP will not work out of the box.

    is there a link or instructions of some sort on how to set one up non pnp.

  4. Mitel SIP Phone

    in VoIP Phones

    Posted

    Hi there,

     

    i have been testing out pbxnsip with Exchange UM and OCS, and things are going pretty good. I want to be able to test out dual forking with the use of office communicator and a desk phone. I currently only have a mitel 5220 SIP Phone. Can this phone work with pbxnsip? How do i go about this.

     

    Thanks

     

    Joso

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