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    Debug 702

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    ion marian
    The document describes SIP signaling messages exchanged between an IP PBX and a SIP phone during a call setup attempt from one phone to another. Key details: 1. The SIP phone sends an INVITE message to initiate a call to another phone number. 2. The IP PBX responds with a 416 Unsupported URI scheme error, indicating an issue with the destination phone number. 3. The SIP phone acknowledges the error with an ACK message. 4. Debugging shows the call attempt and error response.

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    © All Rights Reserved
    Download as DOCX, PDF, TXT or read online from Scribd

    Debug 702

    Uploaded by

    ion marian
    16 views9 pages
    The document describes SIP signaling messages exchanged between an IP PBX and a SIP phone during a call setup attempt from one phone to another. Key details: 1. The SIP phone sends an INVITE message to initiate a call to another phone number. 2. The IP PBX responds with a 416 Unsupported URI scheme error, indicating an issue with the destination phone number. 3. The SIP phone acknowledges the error with an ACK message. 4. Debugging shows the call attempt and error response.

    Original Description:

    pbx

    Original Title

    debug_702

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    © © All Rights Reserved

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    The document describes SIP signaling messages exchanged between an IP PBX and a SIP phone during a call setup attempt from one phone to another. Key details: 1. The SIP phone sends an INVITE message to initiate a call to another phone number. 2. The IP PBX responds with a 416 Unsupported URI scheme error, indicating an issue with the destination phone number. 3. The SIP phone acknowledges the error with an ACK message. 4. Debugging shows the call attempt and error response.

    Copyright:

    © All Rights Reserved
    Download as DOCX, PDF, TXT or read online from Scribd
    Download as docx, pdf, or txt
    16 views9 pages

    Debug 702

    Uploaded by

    ion marian
    The document describes SIP signaling messages exchanged between an IP PBX and a SIP phone during a call setup attempt from one phone to another. Key details: 1. The SIP phone sends an INVITE message to initiate a call to another phone number. 2. The IP PBX responds with a 416 Unsupported URI scheme error, indicating an issue with the destination phone number. 3. The SIP phone acknowledges the error with an ACK message. 4. Debugging shows the call attempt and error response.

    Copyright:

    © All Rights Reserved
    Download as DOCX, PDF, TXT or read online from Scribd
    Download as docx, pdf, or txt
    of 9

    elastix-phone*CLI> sip set debug peer 700701

    SIP Debugging Enabled for IP: 92.87.198.105

    <--- SIP read from UDP:92.87.198.105:5060 --->

    INVITE sip:[email protected] SIP/2.0

    Via: SIP/2.0/UDP 92.87.198.105:5060;branch=z9hG4bKlxgwoxgzoxtdytdgokd0rv3ykT17341

    Call-ID: asbcvjxv1ihxrz0zzvpr32il12jlxl30a88j@ATS.bucats01.as1.romtelecom.net.144

    From: <tel:0243204135;phone-context=+40>;tag=sbc0403113pigl8-CC-144

    To: <sip:[email protected];user=phone>

    CSeq: 1 INVITE

    Alert-Info: <http://www.huawei.com/ring/>;info=pattern1

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

    Contact: <sip:[email protected]:5060>

    Max-Forwards: 65

    Supported: 100rel,timer

    Session-Expires: 1800

    Min-SE: 600

    P-Asserted-Identity: <tel:0243204135;phone-context=+40>

    P-Called-Party-ID: <tel:+40244700702>

    Content-Length: 197

    Content-Type: application/sdp

    v=0

    o=- 68839818 68839818 IN IP4 92.87.198.106

    s=SBC call
    c=IN IP4 92.87.198.106

    t=0 0

    m=audio 14006 RTP/AVP 8 101

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

    a=ptime:20

    a=sendrecv

    <------------->

    --- (17 headers 10 lines) ---

    Sending to 92.87.198.105:5060 (NAT)

    Sending to 92.87.198.105:5060 (NAT)

    Using INVITE request as basis request -


    asbcvjxv1ihxrz0zzvpr32il12jlxl30a88j@ATS.bucats01.as1.romtelecom.net.144

    [2016-08-22 17:10:55] NOTICE[25952][C-0000001d]: chan_sip.c:18577 check_user_full: From address


    missing 'sip:', using it anyway

    Found peer '700701' for 'tel' from 92.87.198.105:5060

    Found RTP audio format 8

    Found RTP audio format 101

    Found audio description format telephone-event for ID 101

    Capabilities: us - (alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)

    Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined -


    0x1 (telephone-event|)

    Peer audio RTP is at port 92.87.198.106:14006

    [2016-08-22 17:10:55] WARNING[25952][C-0000001d]: chan_sip.c:17696 get_destination: Not a SIP


    header ()?

    <--- Reliably Transmitting (NAT) to 92.87.198.105:5060 --->


    Via: SIP/2.0/UDP
    92.87.198.105:5060;branch=z9hG4bKlxgwoxgzoxtdytdgokd0rv3ykT17341;received=92.87.198.105;rport
    =5060

    From: <tel:0243204135;phone-context=+40>;tag=sbc0403113pigl8-CC-144

    To: <sip:[email protected];user=phone>;tag=as1fefbfd7

    Call-ID: asbcvjxv1ihxrz0zzvpr32il12jlxl30a88j@ATS.bucats01.as1.romtelecom.net.144

    CSeq: 1 INVITE

    Server: FPBX-2.11.0(11.20.0)

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

    Supported: replaces, timer

    Content-Length: 0

    <------------>

    Scheduling destruction of SIP dialog


    'asbcvjxv1ihxrz0zzvpr32il12jlxl30a88j@ATS.bucats01.as1.romtelecom.net.144' in 6400 ms (Method:
    INVITE)

    <--- SIP read from UDP:92.87.198.105:5060 --->

    ACK sip:[email protected] SIP/2.0

    Via: SIP/2.0/UDP
    92.87.198.105:5060;branch=z9hG4bKlxgwoxgzoxtdytdgokd0rv3ykT17341;received=92.87.198.105;rport
    =5060

    Call-ID: asbcvjxv1ihxrz0zzvpr32il12jlxl30a88j@ATS.bucats01.as1.romtelecom.net.144

    From: <tel:0243204135;phone-context=+40>;tag=sbc0403113pigl8-CC-144

    To: <sip:[email protected];user=phone>;tag=as1fefbfd7

    CSeq: 1 ACK
    Max-Forwards: 70

    Content-Length: 0

    <------------->

    --- (8 headers 0 lines) ---

    Really destroying SIP dialog


    'asbcvjxv1ihxrz0zzvpr32il12jlxl30a88j@ATS.bucats01.as1.romtelecom.net.144' Method: ACK

    <--- SIP read from UDP:92.87.198.105:5060 --->

    INVITE sip:[email protected] SIP/2.0

    Via: SIP/2.0/UDP 92.87.198.105:5060;branch=z9hG4bKxworgwrwox1vxedtz0ewegy0dT38934

    Call-ID: asbcha1laphlj0lp0j02hga2p221rxiv2hgr@ATS.bucats01.as1.romtelecom.net.144

    From: <tel:0243204135;phone-context=+40>;tag=sbc04042l8l8zr3-CC-144

    To: <sip:[email protected];user=phone>

    CSeq: 1 INVITE

    Alert-Info: <http://www.huawei.com/ring/>;info=pattern1

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

    Contact: <sip:[email protected]:5060>

    Max-Forwards: 65

    Supported: 100rel,timer

    Session-Expires: 1800

    Min-SE: 600

    P-Asserted-Identity: <tel:0243204135;phone-context=+40>

    P-Called-Party-ID: <tel:+40244700702>

    Content-Length: 197

    Content-Type: application/sdp
    v=0

    o=- 68839820 68839820 IN IP4 92.87.198.106

    s=SBC call

    c=IN IP4 92.87.198.106

    t=0 0

    m=audio 14020 RTP/AVP 8 101

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

    a=ptime:20

    a=sendrecv

    <------------->

    --- (17 headers 10 lines) ---

    Sending to 92.87.198.105:5060 (NAT)

    Sending to 92.87.198.105:5060 (NAT)

    Using INVITE request as basis request -


    asbcha1laphlj0lp0j02hga2p221rxiv2hgr@ATS.bucats01.as1.romtelecom.net.144

    [2016-08-22 17:10:55] NOTICE[25952][C-0000001e]: chan_sip.c:18577 check_user_full: From address


    missing 'sip:', using it anyway

    Found peer '700701' for 'tel' from 92.87.198.105:5060

    Found RTP audio format 8

    Found RTP audio format 101

    Found audio description format telephone-event for ID 101

    Capabilities: us - (alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)

    Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined -


    0x1 (telephone-event|)

    Peer audio RTP is at port 92.87.198.106:14020


    [2016-08-22 17:10:55] WARNING[25952][C-0000001e]: chan_sip.c:17696 get_destination: Not a SIP
    header ()?

    <--- Reliably Transmitting (NAT) to 92.87.198.105:5060 --->

    SIP/2.0 416 Unsupported URI scheme

    Via: SIP/2.0/UDP
    92.87.198.105:5060;branch=z9hG4bKxworgwrwox1vxedtz0ewegy0dT38934;received=92.87.198.105;rp
    ort=5060

    From: <tel:0243204135;phone-context=+40>;tag=sbc04042l8l8zr3-CC-144

    To: <sip:[email protected];user=phone>;tag=as38a96b8c

    Call-ID: asbcha1laphlj0lp0j02hga2p221rxiv2hgr@ATS.bucats01.as1.romtelecom.net.144

    CSeq: 1 INVITE

    Server: FPBX-2.11.0(11.20.0)

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

    Supported: replaces, timer

    Content-Length: 0

    <------------>

    Scheduling destruction of SIP dialog


    'asbcha1laphlj0lp0j02hga2p221rxiv2hgr@ATS.bucats01.as1.romtelecom.net.144' in 6400 ms (Method:
    INVITE)

    <--- SIP read from UDP:92.87.198.105:5060 --->

    ACK sip:[email protected] SIP/2.0

    Via: SIP/2.0/UDP
    92.87.198.105:5060;branch=z9hG4bKxworgwrwox1vxedtz0ewegy0dT38934;received=92.87.198.105;rp
    ort=5060
    Call-ID: asbcha1laphlj0lp0j02hga2p221rxiv2hgr@ATS.bucats01.as1.romtelecom.net.144

    From: <tel:0243204135;phone-context=+40>;tag=sbc04042l8l8zr3-CC-144

    To: <sip:[email protected];user=phone>;tag=as38a96b8c

    CSeq: 1 ACK

    Max-Forwards: 70

    Content-Length: 0

    <------------->

    --- (8 headers 0 lines) ---

    Really destroying SIP dialog


    'asbcha1laphlj0lp0j02hga2p221rxiv2hgr@ATS.bucats01.as1.romtelecom.net.144' Method: ACK

    <--- SIP read from UDP:92.87.198.105:5060 --->

    hello

    <------------->

    Reliably Transmitting (NAT) to 92.87.198.105:5060:

    OPTIONS sip:as1.romtelecom.net SIP/2.0

    Via: SIP/2.0/UDP 2.0.240.82:5060;branch=z9hG4bK493f2afb;rport

    Max-Forwards: 70

    From: "Unknown" <sip:[email protected]>;tag=as244c1c9d

    To: <sip:as1.romtelecom.net>

    Contact: <sip:[email protected]:5060>

    Call-ID: [email protected]:5060

    CSeq: 102 OPTIONS

    User-Agent: FPBX-2.11.0(11.20.0)

    Date: Mon, 22 Aug 2016 14:11:28 GMT


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

    Supported: replaces, timer

    Content-Length: 0

    ---

    <--- SIP read from UDP:92.87.198.105:5060 --->

    SIP/2.0 200 OK

    Via: SIP/2.0/UDP 2.0.240.82:5060;branch=z9hG4bK493f2afb;rport

    Call-ID: [email protected]:5060

    From: "Unknown"<sip:[email protected]>;tag=as244c1c9d

    To: <sip:as1.romtelecom.net>;tag=sbc040603906k99

    CSeq: 102 OPTIONS

    Content-Length: 0

    <------------->

    --- (7 headers 0 lines) ---

    Really destroying SIP dialog '[email protected]:5060' Method:


    OPTIONS

    elastix-phone*CLI> sip show registry

    Host dnsmgr Username Refresh State Reg.Time

    as1.romtelecom.net:5060 N +40244700701 585 Registered Mon, 22 Aug 2016


    17:07:58

    1 SIP registrations.

    elastix-phone*CLI> sip set debug off


    SIP Debugging Disabled

    elastix-phone*CLI>

    Disconnected from Asterisk server

    Asterisk cleanly ending (0).

    Executing last minute cleanups

    [root@elastix-phone asterisk]#

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