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TomH

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  3. RTP Port Issue

    in Fax Setup

    Posted

    When it works you go to media aware passthrough and get the rtp/t38 ports once

     

    also wanted to add I believe but not 100% that it may be pass through calls coming from extensions work, coming from trunks do not.

     

    [7] 2008/04/30 08:39:44: ac205a6825c71d48@192.168.6.102#74ebbabf04: Media-aware pass-through mode

     

    The issue is the RTP ports are incorrect and an RTP /T38 session is never established on the one that does not work.

     

    You get the ports twice

     

    My question is why do you get the ports twice sometime and once othertimes see below

     

    Passthrough log records when it does not work

     

    [7] 2008/04/30 08:30:00: Determine pass-through mode after receiving response

     

    [7] 2008/04/30 08:30:00: 62be9fb0@pbx#34346: RTP pass-through mode

     

    [7] 2008/04/30 08:30:00: 8609-3418547751-802139@NXT01.broadvox.net#945f1258df: RTP pass-through mode

     

    [7] 2008/04/30 08:30:02: Determine pass-through mode after receiving response

     

    [8] 2008/04/30 08:30:02: Passthrough: Changing destination to 172.26.1.81:8545

     

     

    Passthrough log records when it Works

     

    [7] 2008/04/30 08:39:19: Determine pass-through mode after receiving response

     

    [7] 2008/04/30 08:39:19: 3297304d@pbx#64201: RTP pass-through mode

     

    [7] 2008/04/30 08:39:19: ac205a6825c71d48@192.168.6.102#74ebbabf04: RTP pass-through mode

     

    [7] 2008/04/30 08:39:21: Determine pass-through mode after receiving response

     

    [7] 2008/04/30 08:39:44: ac205a6825c71d48@192.168.6.102#74ebbabf04: Media-aware pass-through mode

     

     

    **************When it does not work you retrieve the RPT ports twice************

     

    [7] 2008/04/30 08:30:02: SIP Rx tcp:172.26.1.81:5067:

    INVITE sip:103@172.26.1.75:2594;transport=tcp SIP/2.0

    FROM: <sip:103@be7.ezoutlook.com;user=phone>;epid=BE122FA941;tag=887c2779c4

    TO: <sip:103@ezoutlook.com>;tag=34346

    CSEQ: 1 INVITE

    CALL-ID: 62be9fb0@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 172.26.1.81:5067;branch=z9hG4bKb7c3d736

    CONTACT: <sip:be7.ezoutlook.com:5067;transport=Tcp;maddr=172.26.1.81;ms-opaque=0d02c8b9260b7733>;automata

    CONTENT-LENGTH: 283

    USER-AGENT: RTCC/3.0.0.0

    CONTENT-TYPE: application/sdp

     

    v=0

    o=- 0 1 IN IP4 172.26.1.81

    s=session

    c=IN IP4 172.26.1.81

    t=0 0

    m=audio 0 RTP/AVP 0 8 101 13

    a=rtpmap:0 PCMU/8000/1

    a=rtpmap:8 PCMA/8000/1

    a=rtpmap:101 telephone-event/8000

    m=image 8545 udptl t38

    a=T38FaxRateManagement:transferredTCF

    a=T38FaxUdpEC:t38UDPRedundancy

     

    ****************first time*************************

    [7] 2008/04/30 08:30:02: UDP: Opening socket on port 9098

    [7] 2008/04/30 08:30:02: UDP: Opening socket on port 9050

     

    .

    .

    .

     

    [7] 2008/04/30 08:30:02: SIP Rx udp:209.249.3.59:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/UDP 172.26.1.75:5060;branch=z9hG4bK-605578d4313fe553d739d96e6861ef95;rport

    To: "EZ OUTLOOK WEB " <sip:5024255328@209.249.3.59>;tag=3418547751-802148

    From: <sip:10000555024102925@209.249.3.56:5060>;tag=945f1258df

    Call-ID: 8609-3418547751-802139@NXT01.broadvox.net

    CSeq: 23081 INVITE

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

    Contact: <sip:5024255328@209.249.3.59:5060>

    Call-Info: <sip:209.249.3.59>;method="NOTIFY;Event=telephone-event;Duration=1000"

    Content-Type: application/sdp

    Content-Length: 315

     

    v=0

    o=NXT01 0 1 IN IP4 209.249.3.59

    s=sip call

    c=IN IP4 209.249.3.60

    t=0 0

    m=audio 44164 RTP/AVP 0 101

    a=rtpmap:0 PCMU/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

    a=sendrecv

    a=ptime:20

    m=image 44170 udptl t38

    a=T38FaxRateManagement:transferredTCF

    a=T38FaxUdpEC:t38UDPRedundancy

    a=sendrecv

     

    [7] 2008/04/30 08:30:02: Call 8609-3418547751-802139@NXT01.broadvox.net#945f1258df: Clear last INVITE

    ****************second time*************************

    [7] 2008/04/30 08:30:02: UDP: Opening socket on port 9006

    [7] 2008/04/30 08:30:02: UDP: Opening socket on port 9064

     

     

    When it does work you get them once

     

    [7] 2008/04/30 08:39:19: SIP Rx tcp:172.26.1.81:5067:

    INVITE sip:103@172.26.1.75:2594;transport=tcp SIP/2.0

    FROM: <sip:103@be7.ezoutlook.com;user=phone>;epid=BE122FA941;tag=7f988c487

    TO: <sip:103@ezoutlook.com>;tag=64201

    CSEQ: 1 INVITE

    CALL-ID: 3297304d@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 172.26.1.81:5067;branch=z9hG4bK44a2a66f

    CONTACT: <sip:be7.ezoutlook.com:5067;transport=Tcp;maddr=172.26.1.81;ms-opaque=0d02c8b9260b7733>;automata

    CONTENT-LENGTH: 283

    USER-AGENT: RTCC/3.0.0.0

    CONTENT-TYPE: application/sdp

     

    v=0

    o=- 0 1 IN IP4 172.26.1.81

    s=session

    c=IN IP4 172.26.1.81

    t=0 0

    m=audio 0 RTP/AVP 0 8 101 13

    a=rtpmap:0 PCMU/8000/1

    a=rtpmap:8 PCMA/8000/1

    a=rtpmap:101 telephone-event/8000

    m=image 8630 udptl t38

    a=T38FaxRateManagement:transferredTCF

    a=T38FaxUdpEC:t38UDPRedundancy

     

    ****************first time and only time*************************

    [7] 2008/04/30 08:39:19: UDP: Opening socket on port 9080

    [7] 2008/04/30 08:39:19: UDP: Opening socket on port 9012

     

    .

    .

    .

     

    [7] 2008/04/30 08:39:21: SIP Rx udp:74.143.31.154:14062:

    SIP/2.0 200 OK

    Via: SIP/2.0/UDP 172.26.1.75:5060;branch=z9hG4bK-3b7e804129e942888eb09361ddde3b82;rport

    From: <sip:103@pbx.ezoutlook.com>;tag=74ebbabf04

    To: "Fax Machine" <sip:105@pbx.ezoutlook.com>;tag=a68eb379542a755d

    Call-ID: ac205a6825c71d48@192.168.6.102

    CSeq: 28130 INVITE

    User-Agent: Grandstream HT287 1.1.0.3

    Warning: 399 74.143.31.154 "detected NAT type is full cone"

    Contact: <sip:105@74.143.31.154:14062>

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

    Content-Type: application/sdp

    Supported: replaces, timer

    Content-Length: 249

     

    v=0

    o=105 8000 8004 IN IP4 74.143.31.154

    s=SIP Call

    c=IN IP4 74.143.31.154

    t=0 0

    m=audio 0 RTP/AVP 0

    a=sendrecv

    a=rtpmap:0 PCMU/8000

    m=image 50168 udptl t38

    a=T38FaxRateManagement:transferredTCF

    a=T38FaxUdpEC:t38UDPRedundancy

    a=ptime:20

     

    [7] 2008/04/30 08:39:21: Call ac205a6825c71d48@192.168.6.102#74ebbabf04: Clear last INVITE

     

    *************************you use the original RTP port pair********************

    [9] 2008/04/30 08:39:21: Resolve 7467: tcp 172.26.1.81 5067

    [7] 2008/04/30 08:39:21: SIP Tx tcp:172.26.1.81:5067:

  4. RTP Port Issue

    in Fax Setup

    Posted

    The bind() message should not be the problem (we already took it out in a later build). The PBX just tries to get a port, and if that port is not available, it tries another one.

     

    Of course, make sure that you have a large RTP port range, so that you are not running out of RTP ports.

     

    I still don't clearly understand what makes the difference between the failed call and the successful call. Do you say that is depends on what port it chooses? If that is the case, can you double-check the firewall port range?

     

    Other potential reasons for such behavior are usually race conditions, e.g. the answer comes earlier or later. Is there anything in this direction?

     

    I agree the Bind is fine. In a test situation with plenty of ports. The issue is on the faxes that do not work PBXnSIP retrieves a pair of open ports for RTP/T38 twice. Once correctly and once after after the recieved OK from BroadVox which invalidates the port number originally given to BroadVox.

     

    In the example I gave you:

     

    [7] 2008/04/23 13:42:28: UDP: Opening socket on port 9048

    [7] 2008/04/23 13:42:28: UDP: Opening socket on port 9008

     

    is the original pair

     

    and 9008 was given to BroadVox

     

    Then you retrieved a second pair this time

     

    [7] 2008/04/23 13:42:29: UDP: Opening socket on port 9064

    [7] 2008/04/23 13:42:29: UDP: Opening socket on port 9030

     

    and used 9064 to go to broadvox. i.e. all goes in the bit bucket.

     

     

    When you dont retrieve the secnd pair it always works. The question is why do you retrieve a second pair sometimes and not others. You are consistantly doing the same thing between devices. If its going for device A to device B it always has the problem or always works depending if you retrieve the second pair.

     

    I am guessing that the issue may be PBXnSIP:

     

    IF IT WORKS - returns from end to end understanding it is a return and remembers and uses the original pair.

    understands FAX > PBXnSIP > BroadVox > PBXnSIP

     

     

    IF IT FAILS - returns from end to end does not understanding it is a return and does not remember and get a new pair pair.

    thinks BroadVox > PBXnSIP

     

    Totally a guess though.

     

     

     

    Thanks

     

    Tom

  5. RTP Port Issue

    in Fax Setup

    Posted

    Configuration

     

    Exchange inbound fax <> PBXnSIP <> BroadVox

    HT286 Fax

    Kapaga fax phone

     

    Issue

     

    After review we find that we consistantly work or don't work. The difference being that the inbound fax device changes to T38 with a reinvite using PORT 9001, PBXnSIP gets two ports for RTP (10001, 10002) tells Broadvox he will use port 10001, BroadVox says he will use port 20002 in an OK,

     

    If it works PBXnSIP OK's the client invite and uses 10002 so that

     

    fax to pbxnsip RTP is 9001 <> 10002

    PBXnSIP to broadvox RTP is 10001 <> 20002

     

     

     

    If it does not work PBXnSIP gets two new ports (10003, 10004) and OK's the client invite with 10003 so that:

     

    fax to pbxnsip RTP is 9001 <> 10003

    PBXnSIP to broadvox RTP is 10004 <> 20002

     

    The issue is I believe that a state/route has been setup on Broadvox linking 10001 <> 20002 when messages come in 10004 <> 20002 they go in the bit bucket.

     

    My questions is why is there an inconsistancy? Some times you use the original ports (this works), and sometimes you get new ports (this fails).

     

     

    Thanks for any guidance. I have included and commented the relevant part of the log.

     

    Tom

     

     

    Invite from fax client requesting T38

     

    [7] 2008/04/23 13:42:28: SIP Rx tcp:172.1.1.81:5065:

    INVITE sip:103@172.1.1.75:4957;transport=tcp SIP/2.0

    FROM: <sip:103@be7.mydomain.com;user=phone>;epid=34AEC95009;tag=54acd35ba1

    TO: <sip:103@ezoutlook.com>;tag=5392

    CSEQ: 1 INVITE

    CALL-ID: 46269039@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 172.1.1.81:5065;branch=z9hG4bK12926ec

    CONTACT: <sip:be7.mydomain.com:5065;transport=Tcp;maddr=172.1.1.81;ms-opaque=2eb6a8402dbb9419>;automata

    CONTENT-LENGTH: 284

    USER-AGENT: RTCC/3.0.0.0

    CONTENT-TYPE: application/sdp

     

    v=0

    o=- 0 1 IN IP4 172.1.1.81

    s=session

    c=IN IP4 172.1.1.81

    t=0 0

    m=audio 0 RTP/AVP 0 8 101 13

    a=rtpmap:0 PCMU/8000/1

    a=rtpmap:8 PCMA/8000/1

    a=rtpmap:101 telephone-event/8000

    m=image 16861 udptl t38

    a=T38FaxRateManagement:transferredTCF

    a=T38FaxUdpEC:t38UDPRedundancy

     

     

    [0] 2008/04/23 13:42:28: UDP: bind() to port 9066 failed

     

    Gets two ports, above bind failure does not always happen

     

    [7] 2008/04/23 13:42:28: UDP: Opening socket on port 9048

    [7] 2008/04/23 13:42:28: UDP: Opening socket on port 9008

    [9] 2008/04/23 13:42:28: Resolve 72699: url sip:5025555555@200.200.3.59:5060

    [9] 2008/04/23 13:42:28: Resolve 72699: udp 200.200.3.59 5060

     

    Invites broadvox

     

    [7] 2008/04/23 13:42:28: SIP Tx udp:200.200.3.59:5060:

    INVITE sip:5025555555@200.200.3.59:5060 SIP/2.0

    Via: SIP/2.0/UDP 172.1.1.75:5060;branch=z9hG4bK-48ad4751e9b09902de83fe8ca58bdf61;rport

    From: <sip:10000555024444444@200.200.3.56:5060>;tag=bfe14f6334

    To: "EZ OUTLOOK WEB " <sip:5025555555@200.200.3.59>;tag=3417961711-711590

    Call-ID: 1560-3417961711-711581@NXT01.broadvox.net

    CSeq: 13831 INVITE

    Max-Forwards: 70

    Contact: <sip:Anonymous@172.1.1.75:5060;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/2.1.8.2463

    Content-Type: application/sdp

    Content-Length: 283

     

    v=0

    o=- 17827 17828 IN IP4 172.1.1.75

    s=-

    c=IN IP4 172.1.1.75

    t=0 0

    m=audio 0 RTP/AVP 0 8 101 13

    a=rtpmap:0 PCMU/8000/1

    a=rtpmap:8 PCMA/8000/1

    a=rtpmap:101 telephone-event/8000

    m=image 9008 udptl t38

    a=T38FaxRateManagement:transferredTCF

    a=T38FaxUdpEC:t38UDPRedundancy

     

    [9] 2008/04/23 13:42:28: Resolve 72700: tcp 172.1.1.81 5065

    [7] 2008/04/23 13:42:28: SIP Tx tcp:172.1.1.81:5065:

    SIP/2.0 100 Trying

    Via: SIP/2.0/TCP 172.1.1.81:5065;branch=z9hG4bK12926ec

    From: <sip:103@be7.mydomain.com;user=phone>;epid=34AEC95009;tag=54acd35ba1

    To: <sip:103@ezoutlook.com>;tag=5392

    Call-ID: 46269039@pbx

    CSeq: 1 INVITE

    Content-Length: 0

     

     

    [7] 2008/04/23 13:42:29: SIP Rx udp:200.200.3.59:5060:

    SIP/2.0 100 Trying

    Via: SIP/2.0/UDP 172.1.1.75:5060;branch=z9hG4bK-48ad4751e9b09902de83fe8ca58bdf61;rport

    From: <sip:10000555024444444@200.200.3.56:5060>;tag=bfe14f6334

    To: "EZ OUTLOOK WEB " <sip:5025555555@200.200.3.59>;tag=3417961711-711590

    Call-ID: 1560-3417961711-711581@NXT01.broadvox.net

    CSeq: 13831 INVITE

    Content-Length: 0

     

     

    Broadvox OK

     

    [7] 2008/04/23 13:42:29: SIP Rx udp:200.200.3.59:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/UDP 172.1.1.75:5060;branch=z9hG4bK-48ad4751e9b09902de83fe8ca58bdf61;rport

    To: "EZ OUTLOOK WEB " <sip:5025555555@200.200.3.59>;tag=3417961711-711590

    From: <sip:10000555024444444@200.200.3.56:5060>;tag=bfe14f6334

    Call-ID: 1560-3417961711-711581@NXT01.broadvox.net

    CSeq: 13831 INVITE

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

    Contact: <sip:5025555555@200.200.3.59:5060>

    Call-Info: <sip:200.200.3.59>;method="NOTIFY;Event=telephone-event;Duration=1000"

    Content-Type: application/sdp

    Content-Length: 315

     

    v=0

    o=NXT01 0 1 IN IP4 200.200.3.59

    s=sip call

    c=IN IP4 200.200.3.60

    t=0 0

    m=audio 16222 RTP/AVP 0 101

    a=rtpmap:0 PCMU/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

    a=sendrecv

    a=ptime:20

    m=image 16228 udptl t38

    a=T38FaxRateManagement:transferredTCF

    a=T38FaxUdpEC:t38UDPRedundancy

    a=sendrecv

     

    [7] 2008/04/23 13:42:29: Call 1560-3417961711-711581@NXT01.broadvox.net#bfe14f6334: Clear last INVITE

     

     

    HERE IS THE ISSUE IF YOU GET TWO NEW PORTS IT FAILS IF YOU KEEP THE ORIGINALS IT WORKS

     

    [7] 2008/04/23 13:42:29: UDP: Opening socket on port 9064

    [7] 2008/04/23 13:42:29: UDP: Opening socket on port 9030

     

    Now you are using a port 9064 for RTP to Broadvox instead of the stated port 9008

     

    [9] 2008/04/23 13:42:29: Resolve 72701: tcp 172.1.1.81 5065

    [7] 2008/04/23 13:42:29: SIP Tx tcp:172.1.1.81:5065:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 172.1.1.81:5065;branch=z9hG4bK12926ec

    From: <sip:103@be7.mydomain.com;user=phone>;epid=34AEC95009;tag=54acd35ba1

    To: <sip:103@ezoutlook.com>;tag=5392

    Call-ID: 46269039@pbx

    CSeq: 1 INVITE

    Contact: <sip:103@172.1.1.75:4957;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/2.1.8.2463

    Content-Type: application/sdp

    Content-Length: 282

     

    v=0

    o=- 1981 1982 IN IP4 172.1.1.75

    s=-

    c=IN IP4 172.1.1.75

    t=0 0

    m=audio 9048 RTP/AVP 0 101

    a=rtpmap:0 PCMU/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

    a=ptime:20

    m=image 9030 udptl t38

    a=T38FaxRateManagement:transferredTCF

    a=T38FaxUdpEC:t38UDPRedundancy

     

    [9] 2008/04/23 13:42:29: Resolve 72702: url sip:5025555555@200.200.3.59:5060

    [9] 2008/04/23 13:42:29: Resolve 72702: udp 200.200.3.59 5060

    [7] 2008/04/23 13:42:29: SIP Tx udp:200.200.3.59:5060:

    ACK sip:5025555555@200.200.3.59:5060 SIP/2.0

    Via: SIP/2.0/UDP 172.1.1.75:5060;branch=z9hG4bK-41742d16b815e8fef8a65e845f1a72cc;rport

    From: <sip:10000555024444444@200.200.3.56:5060>;tag=bfe14f6334

    To: "EZ OUTLOOK WEB " <sip:5025555555@200.200.3.59>;tag=3417961711-711590

    Call-ID: 1560-3417961711-711581@NXT01.broadvox.net

    CSeq: 13831 ACK

    Max-Forwards: 70

    Contact: <sip:Anonymous@172.1.1.75:5060;transport=udp>

    Content-Length: 0

     

     

    [7] 2008/04/23 13:42:29: Determine pass-through mode after receiving response

    [7] 2008/04/23 13:42:29: SIP Rx tcp:172.1.1.81:5065:

    ACK sip:103@172.1.1.75:4957;transport=tcp SIP/2.0

    FROM: <sip:103@be7.mydomain.com;user=phone>;epid=34AEC95009;tag=54acd35ba1

    TO: <sip:103@ezoutlook.com>;tag=5392

    CSEQ: 1 ACK

    CALL-ID: 46269039@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 172.1.1.81:5065;branch=z9hG4bK28915536

    CONTENT-LENGTH: 0

    USER-AGENT: RTCC/3.0.0.0

     

     

    [8] 2008/04/23 13:42:29: Passthrough: Changing destination to 172.1.1.81:16861

    [5] 2008/04/23 13:43:08: SIP port accept from 172.1.1.81:16868

    [7] 2008/04/23 13:43:08: SIP Rx tcp:172.1.1.81:16868:

    OPTIONS sip:172.1.1.75:5060 SIP/2.0

    FROM: <sip:be7.mydomain.com:5060;transport=Tcp;ms-opaque=932002b23e9bfaae>;epid=6618ACCBF4;tag=b56e3cb8e9

    TO: <sip:172.1.1.75:5060>

    CSEQ: 6449 OPTIONS

    CALL-ID: d6996cf8fab742ee91d089bf12794b6b

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 172.1.1.81:16868;branch=z9hG4bK13c7d83c

    ACCEPT: application/sdp

    CONTENT-LENGTH: 0

    USER-AGENT: RTCC/3.0.0.0

     

     

    [9] 2008/04/23 13:43:08: Resolve 72703: tcp 172.1.1.81 16868

    [7] 2008/04/23 13:43:08: SIP Tx tcp:172.1.1.81:16868:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 172.1.1.81:16868;branch=z9hG4bK13c7d83c

    From: <sip:be7.mydomain.com:5060;transport=Tcp;ms-opaque=932002b23e9bfaae>;epid=6618ACCBF4;tag=b56e3cb8e9

    To: <sip:172.1.1.75:5060>;tag=7905a6d1f3

    Call-ID: d6996cf8fab742ee91d089bf12794b6b

    CSeq: 6449 OPTIONS

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

    Content-Length: 0

     

     

    [7] 2008/04/23 13:43:15: SIP Rx tcp:172.1.1.81:5065:

    BYE sip:103@172.1.1.75:4957;transport=tcp SIP/2.0

    FROM: <sip:103@be7.mydomain.com;user=phone>;epid=34AEC95009;tag=54acd35ba1

    TO: <sip:103@ezoutlook.com>;tag=5392

    CSEQ: 2 BYE

    CALL-ID: 46269039@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 172.1.1.81:5065;branch=z9hG4bK1643ff5

    CONTENT-LENGTH: 0

    USER-AGENT: RTCC/3.0.0.0

     

     

    [9] 2008/04/23 13:43:15: Resolve 72704: tcp 172.1.1.81 5065

    [7] 2008/04/23 13:43:15: SIP Tx tcp:172.1.1.81:5065:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 172.1.1.81:5065;branch=z9hG4bK1643ff5

    From: <sip:103@be7.mydomain.com;user=phone>;epid=34AEC95009;tag=54acd35ba1

    To: <sip:103@ezoutlook.com>;tag=5392

    Call-ID: 46269039@pbx

    CSeq: 2 BYE

    Contact: <sip:103@172.1.1.75:4957;transport=tcp>

    User-Agent: pbxnsip-PBX/2.1.8.2463

    RTP-RxStat: Dur=49,Pkt=129,Oct=11466,Underun=0

    RTP-TxStat: Dur=49,Pkt=62,Oct=10664

    Content-Length: 0

     

     

    [7] 2008/04/23 13:43:15: 1560-3417961711-711581@NXT01.broadvox.net#bfe14f6334: Media-aware pass-through mode

    [7] 2008/04/23 13:43:15: Other Ports: 1

    [7] 2008/04/23 13:43:15: Call Port: 1560-3417961711-711581@NXT01.broadvox.net#bfe14f6334

    [9] 2008/04/23 13:43:15: Resolve 72705: url sip:5025555555@200.200.3.59:5060

    [9] 2008/04/23 13:43:15: Resolve 72705: udp 200.200.3.59 5060

    [7] 2008/04/23 13:43:15: SIP Tx udp:200.200.3.59:5060:

    BYE sip:5025555555@200.200.3.59:5060 SIP/2.0

    Via: SIP/2.0/UDP 172.1.1.75:5060;branch=z9hG4bK-9965f8d5a9dd9483b8b0fadae730f1ec;rport

    From: <sip:10000555024444444@200.200.3.56:5060>;tag=bfe14f6334

    To: "EZ OUTLOOK WEB " <sip:5025555555@200.200.3.59>;tag=3417961711-711590

    Call-ID: 1560-3417961711-711581@NXT01.broadvox.net

    CSeq: 13832 BYE

    Max-Forwards: 70

    Contact: <sip:Anonymous@172.1.1.75:5060;transport=udp>

    RTP-RxStat: Dur=49,Pkt=89,Oct=15308,Underun=0

    RTP-TxStat: Dur=47,Pkt=170,Oct=18518

    Content-Length: 0

     

     

    [7] 2008/04/23 13:43:16: SIP Rx udp:200.200.3.59:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/UDP 172.1.1.75:5060;branch=z9hG4bK-9965f8d5a9dd9483b8b0fadae730f1ec;rport

    To: "EZ OUTLOOK WEB " <sip:5025555555@200.200.3.59>;tag=3417961711-711590

    From: <sip:10000555024444444@200.200.3.56:5060>;tag=bfe14f6334

    Call-ID: 1560-3417961711-711581@NXT01.broadvox.net

    CSeq: 13832 BYE

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

    Contact: <sip:5025555555@200.200.3.59:5060>

    Content-Length: 0

  6. T.38 pass through to Exchange 2007

    in Microsoft Exchange

    Posted

    1) There is a global setting called "support_update", if you set it to false the PBX will use INVITE (see http://wiki.pbxnsip.com/index.php/Global_Configuration_File on how to set it).

     

    2) Yea, I agree that the audio with port 0 is really strange (well, that's how we get it!). Maybe Broadvox can try with the INVITE and if that does no work we can make a version that takes the audio part out in this case.

     

     

    Here is where it stands Broadvox is not going to fix the issue with multiple m records in the sdp any time soon. I need to move on. I am going to look for a SIP trunk vendor that does support this. At the same time does PBXnSIP have the abilility to recieve and forward inbound faxes? this is another alternative for us.

  7. T.38 pass through to Exchange 2007

    in Microsoft Exchange

    Posted

    I also found this it appears CISCO call manager is having the same issue can you help?

     

    http://forums.microsoft.com/TechNet/ShowPo...1&SiteID=17

     

    Following is exerpt from microsoft forum.

     

    "We've been down this path, and unfortunately were not able to reach any sort of solution via CUCM 6.01 & Ex07 SP1 (regardless of inband detection enabled). Neither or Cisco or Microsoft are really able to help on this unless they come to some sort of agreement. The details are basically as follows:

     

    Microsoft and Cisco use two different standards for handling t.38 calls over SIP. The SIP signaling used by Microsoft Exchange to switch a call from audio to T.38 fax is not understood by CallManager, causing fax calls to fail.

     

    Specifically, Exchange sends SDP with two "m=" lines, one to terminate audio and one to enable fax session. However, CallManager interprets it as audio requests that terminate the audio channels.

     

    Currently CallManager only supports send/receive SIP INVITE signal to switch an audio call to T.38 call by a single image m=line in SIP SDP portion.

     

    This SDP signals the endpoint to replace the existing channels (audio in our case) and establish a new channel for T.38.

     

    Both implementations by Cisco CallManager and Microsoft Exchange conform to the standards (RFC3264 (MS) and ITU T.38 (Cisco)).

     

    Support for both standards is in development for CallManager, This functionality will be fully supported in CallManager 7.0. It is currently in a 'resolved' state meaning this functionality is tested and confirmed in development builds of 7.0.

     

    This is tracked via BugID CSCsg60357, but was private the last I checked.

     

    Hope this helps. If anyone can use this information to get this functional, I'd love to see it."

  8. T.38 pass through to Exchange 2007

    in Microsoft Exchange

    Posted

    1) There is a global setting called "support_update", if you set it to false the PBX will use INVITE (see http://wiki.pbxnsip.com/index.php/Global_Configuration_File on how to set it).

     

    2) Yea, I agree that the audio with port 0 is really strange (well, that's how we get it!). Maybe Broadvox can try with the INVITE and if that does no work we can make a version that takes the audio part out in this case.

     

    Well, just heard back from BroadVox. They are saying that the issue is that there are two codex in the SDP:

     

    m=audio 0 RTP/AVP 0 8 101 13

    a=rtpmap:0 PCMU/8000/1

    a=rtpmap:8 PCMA/8000/1

    a=rtpmap:101 telephone-event/8000

    m=image 9200 udptl t38

    a=T38FaxRateManagement:transferredTCF

    a=T38FaxUdpEC:t38UDPFEC

     

    I need to get rid of the audio which does have port 0. I know you all are just passing this through from exchange but can you offer any help on correcting it on exchange or correcting it in the pass through. I believe they are correct and if I can remove it the fax will pass through correctly.

     

    Also, are there other PBXNSIP users sucessfully doing inbound faxing. If so maybe I need to reinstall.

     

     

    Tom

  9. T.38 pass through to Exchange 2007

    in Microsoft Exchange

    Posted

    Well, sending to port 0 is not an option in IP, so that must be changed anyway. If that was the last obstable? You never know!

     

    BroadVox has two questions.

     

    1) see below we recieve an "INVITE" from exchange but passs it through to broadvox as a "UPDATE" can it be passed through as an "INVITE"?

     

    2) and more important can we limit the protocols to just T38 so there is no choise?

     

    Thanks

     

    Receives INVITE from 172.26.1.81 (Exchange) to go to T38

    [7] 2008/02/28 16:45:39: SIP Rx tcp:172.26.1.81:5065:

    INVITE sip:103@172.26.1.75:3006;transport=tcp SIP/2.0

    FROM: <sip:103@be7.ezoutlook.com;user=phone>;epid=AA79D0A609;tag=a9fab924e2

    TO: <sip:103@ezoutlook.com>;tag=59403

    CSEQ: 1 INVITE

    CALL-ID: a0b0509d@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 172.26.1.81:5065;branch=z9hG4bK2dfacb75

    CONTACT: <sip:be7.ezoutlook.com:5065;transport=Tcp;maddr=172.26.1.81;ms-opaque=c9e23a1203e9a49b>;automata

    CONTENT-LENGTH: 276

    USER-AGENT: RTCC/3.0.0.0

    CONTENT-TYPE: application/sdp

     

    v=0

    o=- 0 1 IN IP4 172.26.1.81

    s=session

    c=IN IP4 172.26.1.81

    t=0 0

    m=audio 0 RTP/AVP 0 8 101 13

    a=rtpmap:0 PCMU/8000/1

    a=rtpmap:8 PCMA/8000/1

    a=rtpmap:101 telephone-event/8000

    m=image 9200 udptl t38

    a=T38FaxRateManagement:transferredTCF

    a=T38FaxUdpEC:t38UDPFEC

    Sends UPDATE to BroadVox

     

    [7] 2008/02/28 16:45:39: SIP Tx udp:64.152.60.75:5060:

    UPDATE sip:2163736227@64.152.60.75:5060 SIP/2.0

    Via: SIP/2.0/UDP 172.26.1.75:5060;branch=z9hG4bK-7dcdaaf32ca71adac0facc4140d0d153;rport

    From: <sip:5024102925@172.26.1.75>;tag=c98233afea

    To: "JERGENS INC " <sip:2163736227@64.152.60.75>;tag=gK0229cd68

    Call-ID: 1661104583_128358926@64.152.60.75

    CSeq: 2771 UPDATE

    Max-Forwards: 70

    Contact: <sip:Anonymous@172.26.1.75:5060;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/2.1.6.2448

    Content-Type: application/sdp

    Content-Length: 276

     

    v=0

    o=- 43466 43467 IN IP4 172.26.1.75

    s=-

    c=IN IP4 172.26.1.75

    t=0 0

    m=audio 0 RTP/AVP 0 8 101 13

    a=rtpmap:0 PCMU/8000/1

    a=rtpmap:8 PCMA/8000/1

    a=rtpmap:101 telephone-event/8000

    m=image 9098 udptl t38

    a=T38FaxRateManagement:transferredTCF

    a=T38FaxUdpEC:t38UDPFEC

  10. T.38 pass through to Exchange 2007

    in Microsoft Exchange

    Posted

    Yes that is definitevely a problem. Broadvox is usually quite responsive, would be great to have them provide this kind of service! Did you also ask them if they can do anything about it?

     

     

    Yes, I provided them the same info and will let you know what they say. From your standpoint this is something that should work and the issues appears to be on their side?

  11. T.38 pass through to Exchange 2007

    in Microsoft Exchange

    Posted

    My issue is T.38 pass through to Exchange 2007.

     

    Any help on missing pieces, trouble shooting, instructions, or examples would be appreciated.

    Thanks to anyone for any help.

     

     

    The flow is as follows

     

    Fax Machine >> Broadvox >> PBXnSIP >> Exchange 2007

     

    Some facts are:

    • If called on a telephone line pbxnsip rings the phone and then passes the call on to exchange which answers and forward to exchange voice mail if no answer. Everything works great.

    • Exchange is 172.x.x.81

    • PBXnSIP is 172.x.x.75

     

     

    When a fax machine rings:

     

    *PBXnSIP rings phones

    INVITE sip:103@be7.domain.com;user=phone SIP/2.0

    .

    .

    .

    *PBXnSIP Transfers to Exchange

    [7] 2008/02/24 16:57:47: SIP Tx tcp:172.x.x.81:5065:

    INVITE sip:103@be7.domain.com:5065;user=phone;transport=TCP SIP/2.0

    .

    .

    .

    *Exchange pickups

    *Exchange recognizes fax

    *Exchange transitions to T38

    [7] 2008/02/24 16:57:58: SIP Rx tcp:172.x.x.81:5065:

    INVITE sip:103@172.x.x.75:3444;transport=tcp SIP/2.0

    FROM: <sip:103@be7.domain.com;user=phone>;epid=AA79D0A609;tag=f1f4aa832

    TO: <sip:103@domain.com>;tag=20019

    .

    .

    m=image 9200 udptl t38

    a=T38FaxRateManagement:transferredTCF

    a=T38FaxUdpEC:t38UDPFEC

     

     

    *PBXnSIP transition to T38

    [7] 2008/02/24 16:57:58: SIP Tx udp:64.152.60.75:5060:

    UPDATE sip:5552907492@64.152.60.75:5060 SIP/2.0

    Via: SIP/2.0/UDP 172.x.x.75:5060;branch=z9hG4bK-b070d615504a1056cba841938ea3e077;rport

    From: <sip:5554102925@172.x.x.75>;tag=a9a6181f02

    To: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4

    .

    .

    m=image 9086 udptl t38

    a=T38FaxRateManagement:transferredTCF

    a=T38FaxUdpEC:t38UDPFEC

     

     

     

    *BroadVox transitions to T38

    [7] 2008/02/24 16:57:58: SIP Rx udp:64.152.60.75:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/UDP 172.x.x.75:5060;branch=z9hG4bK-b070d615504a1056cba841938ea3e077;rport=5060

    From: <sip:5554102925@172.x.x.75>;tag=a9a6181f02

    To: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4

    Call-ID: 1459988973_58241008@64.152.60.75

    .

    .

    m=image 0 udptl t38

    a=T38FaxRateManagement:transferredTCF

    a= T38FaxUdpEC:t38UDPFEC

    a=sendrecv

     

     

    Issues occurs now, it appear when Broadvox starts pinging port 0 instead of port given. In the m parameter of the SDP protocol I see a zero instead of a valid port# Could this be the issue.

     

    Thanks again.

     

     

    Full PBXnSIP Log follow:

     

    [9] 2008/02/24 16:57:47: Resolve 1975: aaaa udp 64.152.60.75 5060

    [9] 2008/02/24 16:57:47: Resolve 1975: a udp 64.152.60.75 5060

    [9] 2008/02/24 16:57:47: Resolve 1975: udp 64.152.60.75 5060

    [7] 2008/02/24 16:57:47: SIP Tx udp:64.152.60.75:5060:

    SIP/2.0 183 Ringing

    Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK05B04d42fb1611ec6f9

    From: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4

    To: <sip:5554102925@172.x.x.75>;tag=a9a6181f02

    Call-ID: 1459988973_58241008@64.152.60.75

    CSeq: 18297 INVITE

    Contact: <sip:Anonymous@172.x.x.75:5060;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/2.1.6.2448

    Require: 100rel

    RSeq: 1

    Content-Type: application/sdp

    Content-Length: 201

     

    v=0

    o=- 51427 51427 IN IP4 172.x.x.75

    s=-

    c=IN IP4 172.x.x.75

    t=0 0

    m=audio 9036 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=sendrecv

     

     

     

     

     

     

     

     

     

    [8] 2008/02/24 16:57:47: DNS: Add dns_a be7.domain.com 172.x.x.81 (ttl=1200)

    [9] 2008/02/24 16:57:47: Resolve 1974: a tcp be7.domain.com 5060

    [9] 2008/02/24 16:57:47: Resolve 1974: tcp 172.x.x.81 5060

     

     

     

     

     

    [7] 2008/02/24 16:57:47: SIP Tx tcp:172.x.x.81:5060:

    INVITE sip:103@be7.domain.com;user=phone SIP/2.0

    Via: SIP/2.0/TCP 172.x.x.75:3443;branch=z9hG4bK-3a09d7e4ebaaeab8dbc1872cd2c1b08d;rport

    From: "Tom Haselden" <sip:103@domain.com>;tag=20019

    To: <sip:103@be7.domain.com;user=phone>

    Call-ID: 708d4bca@pbx

    CSeq: 8815 INVITE

    Max-Forwards: 70

    Contact: <sip:103@172.x.x.75:3443;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/2.1.6.2448

    Diversion: <tel:103>;reason=no-answer;screen=no;privacy=off

    Content-Type: application/sdp

    Content-Length: 287

     

    v=0

    o=- 62722 62722 IN IP4 172.x.x.75

    s=-

    c=IN IP4 172.x.x.75

    t=0 0

    m=audio 9042 RTP/AVP 0 8 9 2 3 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:9 g722/8000

    a=rtpmap:2 g726-32/8000

    a=rtpmap:3 gsm/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=sendrecv

     

     

     

     

     

     

     

    [7] 2008/02/24 16:57:47: SIP Rx udp:64.152.60.75:5060:

    PRACK sip:Anonymous@216.x.x.75:5060;transport=udp SIP/2.0

    Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK05B05037402bed8a5c3

    From: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4

    To: <sip:5554102925@172.x.x.75>;tag=a9a6181f02

    Call-ID: 1459988973_58241008@64.152.60.75

    CSeq: 18298 PRACK

    Max-Forwards: 70

    RAck: 1 18297 INVITE

    Content-Length: 0

     

     

    [9] 2008/02/24 16:57:47: Resolve 1976: aaaa udp 64.152.60.75 5060

    [9] 2008/02/24 16:57:47: Resolve 1976: a udp 64.152.60.75 5060

    [9] 2008/02/24 16:57:47: Resolve 1976: udp 64.152.60.75 5060

     

     

     

     

     

     

     

     

     

     

     

    [7] 2008/02/24 16:57:47: SIP Tx udp:64.152.60.75:5060:

    SIP/2.0 200 Ok

    Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK05B05037402bed8a5c3

    From: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4

    To: <sip:5554102925@172.x.x.75>;tag=a9a6181f02

    Call-ID: 1459988973_58241008@64.152.60.75

    CSeq: 18298 PRACK

    Contact: <sip:Anonymous@172.x.x.75:5060;transport=udp>

    User-Agent: pbxnsip-PBX/2.1.6.2448

    Content-Length: 0

     

     

     

     

     

     

     

     

    [7] 2008/02/24 16:57:47: SIP Rx tcp:172.x.x.81:5060:

    SIP/2.0 100 Trying

    FROM: "Tom Haselden"<sip:103@domain.com>;tag=20019

    TO: <sip:103@be7.domain.com;user=phone>

    CSEQ: 8815 INVITE

    CALL-ID: 708d4bca@pbx

    VIA: SIP/2.0/TCP 172.x.x.75:3443;branch=z9hG4bK-3a09d7e4ebaaeab8dbc1872cd2c1b08d;rport

    CONTENT-LENGTH: 0

     

     

     

     

     

     

     

     

     

     

    [7] 2008/02/24 16:57:47: SIP Rx tcp:172.x.x.81:5060:

    SIP/2.0 302 Moved Temporarily

    FROM: "Tom Haselden"<sip:103@domain.com>;tag=20019

    TO: <sip:103@be7.domain.com;user=phone>;tag=60116e5511

    CSEQ: 8815 INVITE

    CALL-ID: 708d4bca@pbx

    VIA: SIP/2.0/TCP 172.x.x.75:3443;branch=z9hG4bK-3a09d7e4ebaaeab8dbc1872cd2c1b08d;rport

    CONTACT: <sip:103@be7.domain.com:5065;user=phone;transport=TCP>

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0

    Diversion: <tel:103>;reason=no-answer;screen=no;privacy=off

     

     

    [7] 2008/02/24 16:57:47: Call 708d4bca@pbx#20019: Clear last INVITE

    [9] 2008/02/24 16:57:47: Resolve 1977: url sip:103@be7.domain.com;user=phone

    [9] 2008/02/24 16:57:47: Resolve 1977: naptr be7.domain.com

    [5] 2008/02/24 16:57:47: Redirecting call

    [9] 2008/02/24 16:57:47: Resolve 1978: url sip:103@be7.domain.com:5065;user=phone;transport=TCP

    [9] 2008/02/24 16:57:47: Resolve 1978: a tcp be7.domain.com 5065

    [9] 2008/02/24 16:57:47: Resolve 1978: tcp 172.x.x.81 5065

     

     

     

     

     

     

    [7] 2008/02/24 16:57:47: SIP Tx tcp:172.x.x.81:5065:

    INVITE sip:103@be7.domain.com:5065;user=phone;transport=TCP SIP/2.0

    Via: SIP/2.0/TCP 172.x.x.75:3444;branch=z9hG4bK-beeb0046e039369145a9685ba9818c99;rport

    From: "Tom Haselden" <sip:103@domain.com>;tag=20019

    To: <sip:103@be7.domain.com;user=phone>

    Call-ID: 708d4bca@pbx

    CSeq: 8816 INVITE

    Max-Forwards: 70

    Contact: <sip:103@172.x.x.75:3444;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/2.1.6.2448

    Diversion: <tel:103>;reason=no-answer;screen=no;privacy=off

    Content-Type: application/sdp

    Content-Length: 287

     

    v=0

    o=- 62722 62722 IN IP4 172.x.x.75

    s=-

    c=IN IP4 172.x.x.75

    t=0 0

    m=audio 9042 RTP/AVP 0 8 9 2 3 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:9 g722/8000

    a=rtpmap:2 g726-32/8000

    a=rtpmap:3 gsm/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=sendrecv

     

     

     

     

     

     

    [8] 2008/02/24 16:57:47: DNS: Add dns_naptr be7.domain.com (ttl=3600)

    [9] 2008/02/24 16:57:47: Resolve 1977: naptr be7.domain.com

    [9] 2008/02/24 16:57:47: Resolve 1977: srv tls _sips._tcp.be7.domain.com

    [8] 2008/02/24 16:57:47: DNS: Add dns_srv _sips._tcp.be7.domain.com (ttl=3600)

    [9] 2008/02/24 16:57:47: Resolve 1977: srv tls _sips._tcp.be7.domain.com

    [9] 2008/02/24 16:57:47: Resolve 1977: srv tcp _sip._tcp.be7.domain.com

    [8] 2008/02/24 16:57:47: DNS: Add dns_srv _sip._tcp.be7.domain.com (ttl=3600)

    [9] 2008/02/24 16:57:47: Resolve 1977: srv tcp _sip._tcp.be7.domain.com

    [9] 2008/02/24 16:57:47: Resolve 1977: srv udp _sip._udp.be7.domain.com

    [8] 2008/02/24 16:57:47: DNS: Add dns_srv _sip._udp.be7.domain.com (ttl=3600)

    [9] 2008/02/24 16:57:47: Resolve 1977: srv udp _sip._udp.be7.domain.com

    [9] 2008/02/24 16:57:47: Resolve 1977: a udp be7.domain.com 5060

    [9] 2008/02/24 16:57:47: Resolve 1977: udp 172.x.x.81 5060

     

     

     

     

     

     

     

     

    [7] 2008/02/24 16:57:47: SIP Tx udp:172.x.x.81:5060:

    ACK sip:103@be7.domain.com;user=phone SIP/2.0

    Via: SIP/2.0/UDP 172.x.x.75:5060;branch=z9hG4bK-3a09d7e4ebaaeab8dbc1872cd2c1b08d;rport

    From: "Tom Haselden" <sip:103@domain.com>;tag=20019

    To: <sip:103@be7.domain.com;user=phone>;tag=60116e5511

    Call-ID: 708d4bca@pbx

    CSeq: 8815 ACK

    Max-Forwards: 70

    Contact: <sip:103@172.x.x.75:5060;transport=udp>

    Content-Length: 0

     

     

     

    [8] 2008/02/24 16:57:47: UDP: recvfrom receives ICMP message

    [5] 2008/02/24 16:57:47: Connection refused on udp:172.x.x.81:5060

    [6] 2008/02/24 16:57:47: Could not determine destination address on 1977

     

     

     

    [7] 2008/02/24 16:57:47: SIP Rx tcp:172.x.x.81:5065:

    SIP/2.0 100 Trying

    FROM: "Tom Haselden"<sip:103@domain.com>;tag=20019

    TO: <sip:103@be7.domain.com;user=phone>

    CSEQ: 8816 INVITE

    CALL-ID: 708d4bca@pbx

    VIA: SIP/2.0/TCP 172.x.x.75:3444;branch=z9hG4bK-beeb0046e039369145a9685ba9818c99;rport

    CONTENT-LENGTH: 0

     

     

     

    [7] 2008/02/24 16:57:49: SIP Rx tcp:172.x.x.81:5065:

    SIP/2.0 180 Ringing

    FROM: "Tom Haselden"<sip:103@domain.com>;tag=20019

    TO: <sip:103@be7.domain.com;user=phone>;epid=AA79D0A609;tag=f1f4aa832

    CSEQ: 8816 INVITE

    CALL-ID: 708d4bca@pbx

    VIA: SIP/2.0/TCP 172.x.x.75:3444;branch=z9hG4bK-beeb0046e039369145a9685ba9818c99;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0

     

     

     

    [8] 2008/02/24 16:57:49: Play audio_en/ringback.wav

     

     

     

    [7] 2008/02/24 16:57:52: SIP Rx tcp:172.x.x.81:5065:

    SIP/2.0 200 OK

    FROM: "Tom Haselden"<sip:103@domain.com>;tag=20019

    TO: <sip:103@be7.domain.com;user=phone>;epid=AA79D0A609;tag=f1f4aa832

    CSEQ: 8816 INVITE

    CALL-ID: 708d4bca@pbx

    VIA: SIP/2.0/TCP 172.x.x.75:3444;branch=z9hG4bK-beeb0046e039369145a9685ba9818c99;rport

    CONTACT: <sip:be7.domain.com:5065;transport=Tcp;maddr=172.x.x.81>;automata

    CONTENT-LENGTH: 193

    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 172.x.x.81

    s=Microsoft Exchange Speech Engine

    c=IN IP4 172.x.x.81

    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] 2008/02/24 16:57:52: Call 708d4bca@pbx#20019: Clear last INVITE

    [7] 2008/02/24 16:57:52: Set packet length to 20

    [6] 2008/02/24 16:57:52: Sending RTP for 708d4bca@pbx#20019 to 172.x.x.81:35840

    [9] 2008/02/24 16:57:52: Resolve 1979: aaaa tcp 172.x.x.81 5065

    [9] 2008/02/24 16:57:52: Resolve 1979: a tcp 172.x.x.81 5065

    [9] 2008/02/24 16:57:52: Resolve 1979: tcp 172.x.x.81 5065

     

     

     

     

     

     

     

     

     

    [7] 2008/02/24 16:57:52: SIP Tx tcp:172.x.x.81:5065:

    ACK sip:be7.domain.com:5065;transport=Tcp;maddr=172.x.x.81 SIP/2.0

    Via: SIP/2.0/TCP 172.x.x.75:3444;branch=z9hG4bK-abf7641b93baa986f47ec986f0906a1b;rport

    From: "Tom Haselden" <sip:103@domain.com>;tag=20019

    To: <sip:103@be7.domain.com;user=phone>;tag=f1f4aa832

    Call-ID: 708d4bca@pbx

    CSeq: 8816 ACK

    Max-Forwards: 70

    Contact: <sip:103@172.x.x.75:3444;transport=tcp>

    Content-Length: 0

     

     

     

     

     

     

     

    [7] 2008/02/24 16:57:52: Determine pass-through mode after receiving response

    [9] 2008/02/24 16:57:52: Resolve 1980: aaaa udp 64.152.60.75 5060

    [9] 2008/02/24 16:57:52: Resolve 1980: a udp 64.152.60.75 5060

    [9] 2008/02/24 16:57:52: Resolve 1980: udp 64.152.60.75 5060

     

     

     

     

     

     

     

     

     

    [7] 2008/02/24 16:57:52: SIP Tx udp:64.152.60.75:5060:

    SIP/2.0 200 Ok

    Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK05B04d42fb1611ec6f9

    From: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4

    To: <sip:5554102925@172.x.x.75>;tag=a9a6181f02

    Call-ID: 1459988973_58241008@64.152.60.75

    CSeq: 18297 INVITE

    Contact: <sip:Anonymous@172.x.x.75:5060;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/2.1.6.2448

    Content-Type: application/sdp

    Content-Length: 201

     

    v=0

    o=- 51427 51427 IN IP4 172.x.x.75

    s=-

    c=IN IP4 172.x.x.75

    t=0 0

    m=audio 9036 RTP/AVP 0 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=sendrecv

     

     

     

     

     

     

     

     

    [7] 2008/02/24 16:57:52: 708d4bca@pbx#20019: RTP pass-through mode

    [7] 2008/02/24 16:57:52: 1459988973_58241008@64.152.60.75#a9a6181f02: RTP pass-through mode

    [7] 2008/02/24 16:57:52: SIP Rx udp:64.152.60.75:5060:

    ACK sip:Anonymous@172.x.x.75:5060;transport=udp SIP/2.0

    Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK05B051200e6bed8a5c3

    From: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4

    To: <sip:5554102925@172.x.x.75>;tag=a9a6181f02

    Call-ID: 1459988973_58241008@64.152.60.75

    CSeq: 18297 ACK

    Max-Forwards: 70

    Content-Length: 0

     

     

    [5] 2008/02/24 16:57:55: Call bf515e7b@pbx#19778: Last request not finished

    [9] 2008/02/24 16:57:55: Resolve 1981: tcp 172.x.x.130 5060

     

     

     

     

     

    [7] 2008/02/24 16:57:55: SIP Tx tcp:172.x.x.130:5060:

    CANCEL sip:55552907492@sip:172.x.x.130:5060;transport=tcp;user=phone SIP/2.0

    Via: SIP/2.0/TCP 172.x.x.75:3442;branch=z9hG4bK-3c718048209c673690214a2681b04258;rport

    From: "Tom Haselden" <sip:103@domain.com>;tag=19778

    To: <sip:55552907492@sip:172.x.x.130:5060;transport=tcp;user=phone>

    Call-ID: bf515e7b@pbx

    CSeq: 26279 CANCEL

    Max-Forwards: 70

    Content-Length: 0

     

     

    [8] 2008/02/24 16:57:55: Hangup: Call bf515e7b@pbx#19778 not found

     

     

     

     

     

     

    [7] 2008/02/24 16:57:58: SIP Rx tcp:172.x.x.81:5065:

    INVITE sip:103@172.x.x.75:3444;transport=tcp SIP/2.0

    FROM: <sip:103@be7.domain.com;user=phone>;epid=AA79D0A609;tag=f1f4aa832

    TO: <sip:103@domain.com>;tag=20019

    CSEQ: 1 INVITE

    CALL-ID: 708d4bca@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 172.x.x.81:5065;branch=z9hG4bK2e2c2c66

    CONTACT: <sip:be7.domain.com:5065;transport=Tcp;maddr=172.x.x.81;ms-opaque=c9e23a1203e9a49b>;automata

    CONTENT-LENGTH: 276

    USER-AGENT: RTCC/3.0.0.0

    CONTENT-TYPE: application/sdp

     

    v=0

    o=- 0 1 IN IP4 172.x.x.81

    s=session

    c=IN IP4 172.x.x.81

    t=0 0

    m=audio 0 RTP/AVP 0 8 101 13

    a=rtpmap:0 PCMU/8000/1

    a=rtpmap:8 PCMA/8000/1

    a=rtpmap:101 telephone-event/8000

    m=image 9200 udptl t38

    a=T38FaxRateManagement:transferredTCF

    a=T38FaxUdpEC:t38UDPFEC

     

     

     

     

    [7] 2008/02/24 16:57:58: UDP: Opening socket on port 9060

    [7] 2008/02/24 16:57:58: UDP: Opening socket on port 9086

     

     

     

    [9] 2008/02/24 16:57:58: Resolve 1982: url sip:5552907492@64.152.60.75:5060

    [9] 2008/02/24 16:57:58: Resolve 1982: udp 64.152.60.75 5060

     

     

     

     

     

    [7] 2008/02/24 16:57:58: SIP Tx udp:64.152.60.75:5060:

    UPDATE sip:5552907492@64.152.60.75:5060 SIP/2.0

    Via: SIP/2.0/UDP 172.x.x.75:5060;branch=z9hG4bK-b070d615504a1056cba841938ea3e077;rport

    From: <sip:5554102925@172.x.x.75>;tag=a9a6181f02

    To: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4

    Call-ID: 1459988973_58241008@64.152.60.75

    CSeq: 707 UPDATE

    Max-Forwards: 70

    Contact: <sip:Anonymous@172.x.x.75:5060;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/2.1.6.2448

    Content-Type: application/sdp

    Content-Length: 276

    v=0

    o=- 51427 51428 IN IP4 172.x.x.75

    s=-

    c=IN IP4 172.x.x.75

    t=0 0

    m=audio 0 RTP/AVP 0 8 101 13

    a=rtpmap:0 PCMU/8000/1

    a=rtpmap:8 PCMA/8000/1

    a=rtpmap:101 telephone-event/8000

    m=image 9086 udptl t38

    a=T38FaxRateManagement:transferredTCF

    a=T38FaxUdpEC:t38UDPFEC

     

     

     

     

    [9] 2008/02/24 16:57:58: Resolve 1983: tcp 172.x.x.81 5065

     

     

     

     

     

    [7] 2008/02/24 16:57:58: SIP Tx tcp:172.x.x.81:5065:

    SIP/2.0 100 Trying

    Via: SIP/2.0/TCP 172.x.x.81:5065;branch=z9hG4bK2e2c2c66

    From: <sip:103@be7.domain.com;user=phone>;epid=AA79D0A609;tag=f1f4aa832

    To: <sip:103@domain.com>;tag=20019

    Call-ID: 708d4bca@pbx

    CSeq: 1 INVITE

    Content-Length: 0

     

     

     

     

     

     

     

    [7] 2008/02/24 16:57:58: SIP Rx udp:64.152.60.75:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/UDP 172.x.x.75:5060;branch=z9hG4bK-b070d615504a1056cba841938ea3e077;rport=5060

    From: <sip:5554102925@172.x.x.75>;tag=a9a6181f02

    To: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4

    Call-ID: 1459988973_58241008@64.152.60.75

    CSeq: 707 UPDATE

    Contact: "Haselden Tom " <sip:5552907492@64.152.60.75:5060>

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

    Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed

    Supported: timer

    Session-Expires: 1800;refresher=uas

    Content-Length: 329

    Content-Disposition: session; handling=required

    Content-Type: application/sdp

     

    v=0

    o=Sonus_UAC 825 15704 IN IP4 64.152.60.75

    s=SIP Media Capabilities

    c=IN IP4 64.152.60.71

    t=0 0

    m=audio 32370 RTP/AVP 0 101

    a=rtpmap:0 PCMU/8000

     

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

    a=sendrecv

    a=ptime:20

    m=image 0 udptl t38

    a=T38FaxRateManagement:transferredTCF

    a= T38FaxUdpEC:t38UDPFEC

    a=sendrecv

     

     

     

     

     

    [7] 2008/02/24 16:57:58: Call 1459988973_58241008@64.152.60.75#a9a6181f02: Clear last request

    [9] 2008/02/24 16:57:58: Resolve 1984: tcp 172.x.x.81 5065

     

     

     

     

     

     

     

     

     

    [7] 2008/02/24 16:57:58: SIP Tx tcp:172.x.x.81:5065:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 172.x.x.81:5065;branch=z9hG4bK2e2c2c66

    From: <sip:103@be7.domain.com;user=phone>;epid=AA79D0A609;tag=f1f4aa832

    To: <sip:103@domain.com>;tag=20019

    Call-ID: 708d4bca@pbx

    CSeq: 1 INVITE

    Contact: <sip:103@172.x.x.75:3444;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/2.1.6.2448

    Content-Type: application/sdp

    Content-Length: 275

     

    v=0

    o=- 62722 62723 IN IP4 172.x.x.75

    s=-

    c=IN IP4 172.x.x.75

    t=0 0

    m=audio 9060 RTP/AVP 0 101

    a=rtpmap:0 PCMU/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

    a=ptime:20

    m=image 0 udptl t38

    a=T38FaxRateManagement:transferredTCF

    a= T38FaxUdpEC:t38UDPFEC

     

     

     

     

     

     

     

     

     

     

     

    [7] 2008/02/24 16:57:58: SIP Rx tcp:172.x.x.81:5065:

    ACK sip:103@172.x.x.75:3444;transport=tcp SIP/2.0

    FROM: <sip:103@be7.domain.com;user=phone>;epid=AA79D0A609;tag=f1f4aa832

    TO: <sip:103@domain.com>;tag=20019

    CSEQ: 1 ACK

    CALL-ID: 708d4bca@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 172.x.x.81:5065;branch=z9hG4bK68d5d4f

    CONTENT-LENGTH: 0

    USER-AGENT: RTCC/3.0.0.0

     

     

     

    [5] 2008/02/24 16:58:02: SIP port accept from 172.x.x.81:18364

     

     

     

     

     

     

     

     

    [7] 2008/02/24 16:58:02: SIP Rx tcp:172.x.x.81:18364:

    OPTIONS sip:172.x.x.75:5060 SIP/2.0

    FROM: <sip:be7.domain.com:5060;transport=Tcp;ms-opaque=95944f3af7203520>;epid=7BE54968BB;tag=e59d6c85f0

    TO: <sip:172.x.x.75:5060>

    CSEQ: 6 OPTIONS

    CALL-ID: ff3f5d4ff10b4315a6865919470a673f

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 172.x.x.81:18364;branch=z9hG4bK8948f21

    ACCEPT: application/sdp

    CONTENT-LENGTH: 0

    USER-AGENT: RTCC/3.0.0.0

     

     

    [9] 2008/02/24 16:58:02: Resolve 1985: tcp 172.x.x.81 18364

     

     

     

     

     

     

    [7] 2008/02/24 16:58:02: SIP Tx tcp:172.x.x.81:18364:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 172.x.x.81:18364;branch=z9hG4bK8948f21

    From: <sip:be7.domain.com:5060;transport=Tcp;ms-opaque=95944f3af7203520>;epid=7BE54968BB;tag=e59d6c85f0

    To: <sip:172.x.x.75:5060>;tag=b3edb24a21

    Call-ID: ff3f5d4ff10b4315a6865919470a673f

    CSeq: 6 OPTIONS

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

    Content-Length: 0

     

     

     

     

     

     

     

    [7] 2008/02/24 16:58:31: SIP Rx udp:64.152.60.75:5060:

    BYE sip:Anonymous@172.x.x.75:5060;transport=udp SIP/2.0

    Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK05B05813609bed8a5c3

    From: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4

    To: <sip:5554102925@172.x.x.75>;tag=a9a6181f02

    Call-ID: 1459988973_58241008@64.152.60.75

    CSeq: 18299 BYE

    Max-Forwards: 70

    Supported: 100rel

    Content-Length: 0

     

     

     

     

     

     

    [9] 2008/02/24 16:58:31: Resolve 1986: aaaa udp 64.152.60.75 5060

    [9] 2008/02/24 16:58:31: Resolve 1986: a udp 64.152.60.75 5060

    [9] 2008/02/24 16:58:31: Resolve 1986: udp 64.152.60.75 5060

     

     

     

     

     

     

     

    [7] 2008/02/24 16:58:31: SIP Tx udp:64.152.60.75:5060:

    SIP/2.0 200 Ok

    Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK05B05813609bed8a5c3

    From: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4

    To: <sip:5554102925@172.x.x.75>;tag=a9a6181f02

    Call-ID: 1459988973_58241008@64.152.60.75

    CSeq: 18299 BYE

    Contact: <sip:Anonymous@172.x.x.75:5060;transport=udp>

    User-Agent: pbxnsip-PBX/2.1.6.2448

    RTP-RxStat: Dur=88,Pkt=272,Oct=46784,Underun=0

    RTP-TxStat: Dur=63,Pkt=272,Oct=46784

    Content-Length: 0

     

     

     

     

     

    [7] 2008/02/24 16:58:31: 708d4bca@pbx#20019: Media-aware pass-through mode

    [7] 2008/02/24 16:58:31: Other Ports: 1

    [7] 2008/02/24 16:58:31: Call Port: 708d4bca@pbx#20019

    [8] 2008/02/24 16:58:31: UDP: recvfrom receives ICMP message

    [8] 2008/02/24 16:58:31: Last message repeated 13 times

     

    [9] 2008/02/24 16:58:31: Resolve 1987: aaaa tcp 172.x.x.81 5065

    [9] 2008/02/24 16:58:31: Resolve 1987: a tcp 172.x.x.81 5065

    [9] 2008/02/24 16:58:31: Resolve 1987: tcp 172.x.x.81 5065

     

     

     

     

     

     

     

     

    [7] 2008/02/24 16:58:31: SIP Tx tcp:172.x.x.81:5065:

    BYE sip:be7.domain.com:5065;transport=Tcp;maddr=172.x.x.81 SIP/2.0

    Via: SIP/2.0/TCP 172.x.x.75:3444;branch=z9hG4bK-846f98ba102960c739a4a8f2c078531a;rport

    From: "Tom Haselden" <sip:103@domain.com>;tag=20019

    To: <sip:103@be7.domain.com;user=phone>;tag=f1f4aa832

    Call-ID: 708d4bca@pbx

    CSeq: 8817 BYE

    Max-Forwards: 70

    Contact: <sip:103@172.x.x.75:3444;transport=tcp>

    RTP-RxStat: Dur=68,Pkt=4,Oct=688,Underun=0

    RTP-TxStat: Dur=63,Pkt=285,Oct=49020

    Content-Length: 0

     

     

     

     

     

    [8] 2008/02/24 16:58:31: UDP: recvfrom receives ICMP message

    [8] 2008/02/24 16:58:31: Last message repeated 3 times

     

     

     

     

     

     

     

     

    [7] 2008/02/24 16:58:31: SIP Rx tcp:172.x.x.81:5065:

    SIP/2.0 200 OK

    FROM: "Tom Haselden"<sip:103@domain.com>;tag=20019

    TO: <sip:103@be7.domain.com;user=phone>;tag=f1f4aa832;epid=AA79D0A609

    CSEQ: 8817 BYE

    CALL-ID: 708d4bca@pbx

    VIA: SIP/2.0/TCP 172.x.x.75:3444;branch=z9hG4bK-846f98ba102960c739a4a8f2c078531a;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0

     

     

     

     

     

     

     

     

     

    [7] 2008/02/24 16:58:31: Call 708d4bca@pbx#20019: Clear last request

    [5] 2008/02/24 16:58:31: BYE Response: Terminate 708d4bca@pbx

    [3] 2008/02/24 16:58:32: SMTP: Cannot resolve mail.domain.com

  13. Splitting an inbound trunk across several domains

    in Trunk Setup

    Posted

    Usually those problems can be solved using the tel:-alias feature (if you give an account an alias name starting with "tel:" then is has a system-global scope). The PBX will then send the call into the right domain automatically, no need for an external proxy doing that job. See http://wiki.pbxnsip.com/index.php/Inbound_Calls_on_Trunk.

     

     

    That did the trick.

     

    Thanks so much I am server literate put PBX challenged. Good to have the help.

  14. Statiscally register an external extension

    in Extension Setup

    Posted

    I have a pbx account "1000" that I can sucessfully call, this is in the domain and PBX

     

    I have an external extension "71000" that I can sucessfully call, it goes through the dial plan locates the trunk and forwards the call.

     

    I can add the extension as a static registration (add contact) but the PBX always looks to resolve it as a URI instead of an extension.

     

    Is there a way to make a static registration for this external extension?

     

    Thanks

  15. Splitting an inbound trunk across several domains

    in Trunk Setup

    Posted

    I have external calls for a single sip gateway coming into a single trunk.

     

    Right now I can make the trunk visible to all domains and set up seperate dial plans for each domain. The issue is the calls always go to the domain the trunk is in and the service cannot resolve the extension of the other domain.

     

    So if domain primary.com wants all calls to 1111xxxx, where 1111 is the domain extension prefix, it works if the trunk is in primary.com.

     

    But if a call comes through for 2222xxxx where 2222 is the domain prefix for domain alternate.com the service cannot resolve it. Even though there is a dial plan for 2222 in the alternate.com domain using the primary.com domain's visible trunk.

     

    I am trying to avoid having a SIP gateway for each domain is there a way to route these external calls to the proper domain based stricly on the extension number prefix.

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