Failed incoming calls from asterisk bridge

Discussion in '3CX Phone System - General' started by Soporte Sinet Colombia, Oct 16, 2017.

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  1. Soporte Sinet Colombia

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    Greetings,

    I have a Linux 3CX 15.5 connected with an Asterisk 1.8 through bridge. Registration and calls from 3CX to Asterisk are OK, but calls from Asterisk to 3CX stay Trying. Servers are connected throuhg VPN and located in different cities. Some details:
    • Asterisk box -> 192.168.21.200/24
    • 3CX box -> 147.122.1.200/16
    Calling to 3CX 607 extension:
    192.168.21.200:5060 147.122.1.200:5060
    =================== ==================
    INVITE (SDP)
    ──────────────>
    INVITE (SDP)
    ──────────────>>>
    407 Proxy Authentication R
    <────────────────
    ACK
    ────────────────>
    INVITE (SDP)
    ────────────────>
    407 Proxy Authentication R
    <<<──────────────
    ACK
    ────────────────>
    INVITE (SDP)
    ────────────────>
    100 Trying
    <───────────────
    100 Trying
    <<<─────────────
     
  2. leejor

    leejor Well-Known Member

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    What does the 3CX Activity Log show for the incoming call/ Does it even show up?
     
  3. Soporte Sinet Colombia

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    Hi leejor, in 3CX log we only see this:

    10/17/2017 8:22:10 AM - Source is identified as trunk Lc:10002(@BRIDGE_DIALVOX_CALI[<sip:10002@192.168.21.200:5060>])
    10/17/2017 8:22:10 AM - Source is identified as trunk Lc:10002(@BRIDGE_DIALVOX_CALI[<sip:10002@192.168.21.200:5060>])
    10/17/2017 8:22:10 AM - IPs do not match!
    10/17/2017 8:22:10 AM - Compare IPs: incoming=192.168.21.200; external=0.0.0.0
    10/17/2017 8:22:10 AM - IPs do not match!
    10/17/2017 8:22:10 AM - Compare IPs: incoming=192.168.21.200; external=200.xxx.xxx.xxx

    Using sngrep to capture traffic, we saw call stuck in Trying.
     
  4. StefanW

    StefanW Head of Customer Support and Training
    Staff Member 3CX Support

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    inside of the 3CX logs you will not be able to determine why the Asterisk remains in Trying.
    You need to debug the Asterisk logs to get an idea where the call would be routed to and why there is no response...
     
    Stop hovering to collapse... Click to collapse... Hover to expand... Click to expand...
  5. Soporte Sinet Colombia

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    Hi StefanW, this is what I see from asterisk side:

    2017/10/17 10:12:13.744656 192.168.21.200:5060 -> 147.122.1.200:5060
    INVITE sip:000@147.122.1.200:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK59e507ac;rport
    Max-Forwards: 70
    From: "Adminfo" <sip:10002@192.168.21.200>;tag=as78f7823c
    To: <sip:000@147.122.1.200:5060>
    Contact: <sip:10002@192.168.21.200:5060>
    Call-ID: 0f84ed7803da72a24903e64658bd0f93@192.168.21.200:5060
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX 1.8.17.0
    Date: Tue, 17 Oct 2017 15:12:13 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Remote-Party-ID: "Adminfo" <sip:900@192.168.21.200>;party=calling;privacy=off;screen=no
    Content-Type: application/sdp
    Content-Length: 286

    v=0
    o=root 999649886 999649886 IN IP4 192.168.21.200
    s=Asterisk PBX 1.8.17.0
    c=IN IP4 192.168.21.200
    t=0 0
    m=audio 18896 RTP/AVP 0 18 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv

    2017/10/17 10:12:13.855536 192.168.21.200:5060 -> 147.122.1.200:5060
    INVITE sip:000@147.122.1.200:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK59e507ac;rport
    Max-Forwards: 70
    From: "Adminfo" <sip:10002@192.168.21.200>;tag=as78f7823c
    To: <sip:000@147.122.1.200:5060>
    Contact: <sip:10002@192.168.21.200:5060>
    Call-ID: 0f84ed7803da72a24903e64658bd0f93@192.168.21.200:5060
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX 1.8.17.0
    Date: Tue, 17 Oct 2017 15:12:13 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Remote-Party-ID: "Adminfo" <sip:900@192.168.21.200>;party=calling;privacy=off;screen=no
    Content-Type: application/sdp
    Content-Length: 286

    v=0
    o=root 999649886 999649886 IN IP4 192.168.21.200
    s=Asterisk PBX 1.8.17.0
    c=IN IP4 192.168.21.200
    t=0 0
    m=audio 18896 RTP/AVP 0 18 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv

    2017/10/17 10:12:13.870059 147.122.1.200:5060 -> 192.168.21.200:5060
    SIP/2.0 407 Proxy Authentication Required
    Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK59e507ac;rport=5060
    Proxy-Authenticate: Digest nonce="414d535959e61dcf31:224747789b410d7e54d28f0856656764",algorithm=MD5,
    alm="3CXPhoneSystem"
    To: <sip:000@147.122.1.200:5060>;tag=59c6677f
    From: "Adminfo" <sip:10002@192.168.21.200>;tag=as78f7823c
    Call-ID: 0f84ed7803da72a24903e64658bd0f93@192.168.21.200:5060
    CSeq: 102 INVITE
    User-Agent: 3CXPhoneSystem 15.5.3849.1 (3849)
    Content-Length: 0

    2017/10/17 10:12:13.870213 192.168.21.200:5060 -> 147.122.1.200:5060
    ACK sip:000@147.122.1.200:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK59e507ac;rport
    Max-Forwards: 70
    From: "Adminfo" <sip:10002@192.168.21.200>;tag=as78f7823c
    To: <sip:000@147.122.1.200:5060>;tag=59c6677f
    Contact: <sip:10002@192.168.21.200:5060>
    Call-ID: 0f84ed7803da72a24903e64658bd0f93@192.168.21.200:5060
    CSeq: 102 ACK
    User-Agent: Asterisk PBX 1.8.17.0
    Content-Length: 0

    2017/10/17 10:12:13.870400 192.168.21.200:5060 -> 147.122.1.200:5060
    INVITE sip:000@147.122.1.200:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK5f81d547;rport
    Max-Forwards: 70
    From: "Adminfo" <sip:10002@192.168.21.200>;tag=as78f7823c
    To: <sip:000@147.122.1.200:5060>
    Contact: <sip:10002@192.168.21.200:5060>
    Call-ID: 0f84ed7803da72a24903e64658bd0f93@192.168.21.200:5060
    CSeq: 103 INVITE
    User-Agent: Asterisk PBX 1.8.17.0
    Proxy-Authorization: Digest username="10002", realm="3CXPhoneSystem", algorithm=MD5, uri="sip:000@147
    22.1.200:5060", nonce="414d535959e61dcf31:224747789b410d7e54d28f0856656764", response="c2593e4e47f3a0
    bc8669f33e198e8a"
    Date: Tue, 17 Oct 2017 15:12:13 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Remote-Party-ID: "Adminfo" <sip:900@192.168.21.200>;party=calling;privacy=off;screen=no
    Content-Type: application/sdp
    Content-Length: 286

    v=0
    o=root 999649886 999649887 IN IP4 192.168.21.200
    s=Asterisk PBX 1.8.17.0
    c=IN IP4 192.168.21.200
    t=0 0
    m=audio 18896 RTP/AVP 0 18 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv

    2017/10/17 10:12:13.881841 147.122.1.200:5060 -> 192.168.21.200:5060
    SIP/2.0 407 Proxy Authentication Required
    Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK59e507ac;rport=5060
    Proxy-Authenticate: Digest nonce="414d535959e61dcf31:224747789b410d7e54d28f0856656764",algorithm=MD5,
    alm="3CXPhoneSystem"
    To: <sip:000@147.122.1.200:5060>;tag=59c6677f
    From: "Adminfo" <sip:10002@192.168.21.200>;tag=as78f7823c
    Call-ID: 0f84ed7803da72a24903e64658bd0f93@192.168.21.200:5060
    CSeq: 102 INVITE
    User-Agent: 3CXPhoneSystem 15.5.3849.1 (3849)
    Content-Length: 0

    2017/10/17 10:12:13.881985 192.168.21.200:5060 -> 147.122.1.200:5060
    ACK sip:000@147.122.1.200:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK5f81d547;rport
    Max-Forwards: 70
    From: "Adminfo" <sip:10002@192.168.21.200>;tag=as78f7823c
    To: <sip:000@147.122.1.200:5060>
    Contact: <sip:10002@192.168.21.200:5060>
    Call-ID: 0f84ed7803da72a24903e64658bd0f93@192.168.21.200:5060
    CSeq: 102 ACK
    User-Agent: Asterisk PBX 1.8.17.0
    Content-Length: 0

    2017/10/17 10:12:13.980507 192.168.21.200:5060 -> 147.122.1.200:5060
    INVITE sip:000@147.122.1.200:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK5f81d547;rport
    Max-Forwards: 70
    From: "Adminfo" <sip:10002@192.168.21.200>;tag=as78f7823c
    To: <sip:000@147.122.1.200:5060>
    Contact: <sip:10002@192.168.21.200:5060>
    Call-ID: 0f84ed7803da72a24903e64658bd0f93@192.168.21.200:5060
    CSeq: 103 INVITE
    User-Agent: Asterisk PBX 1.8.17.0
    Proxy-Authorization: Digest username="10002", realm="3CXPhoneSystem", algorithm=MD5, uri="sip:000@147
    22.1.200:5060", nonce="414d535959e61dcf31:224747789b410d7e54d28f0856656764", response="c2593e4e47f3a0
    bc8669f33e198e8a"
    Date: Tue, 17 Oct 2017 15:12:13 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Remote-Party-ID: "Adminfo" <sip:900@192.168.21.200>;party=calling;privacy=off;screen=no
    Content-Type: application/sdp
    Content-Length: 286

    v=0
    o=root 999649886 999649887 IN IP4 192.168.21.200
    s=Asterisk PBX 1.8.17.0
    c=IN IP4 192.168.21.200
    t=0 0
    m=audio 18896 RTP/AVP 0 18 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv

    2017/10/17 10:12:14.020568 147.122.1.200:5060 -> 192.168.21.200:5060
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK5f81d547;rport=5060
    To: <sip:000@147.122.1.200:5060>
    From: "Adminfo" <sip:10002@192.168.21.200>;tag=as78f7823c
    Call-ID: 0f84ed7803da72a24903e64658bd0f93@192.168.21.200:5060
    CSeq: 103 INVITE
    Content-Length: 0

    2017/10/17 10:12:14.045784 147.122.1.200:5060 -> 192.168.21.200:5060
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK5f81d547;rport=5060
    To: <sip:000@147.122.1.200:5060>
    From: "Adminfo" <sip:10002@192.168.21.200>;tag=as78f7823c
    Call-ID: 0f84ed7803da72a24903e64658bd0f93@192.168.21.200:5060
    CSeq: 103 INVITE
    Content-Length: 0
     
  6. leejor

    leejor Well-Known Member

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    286
    Is extension 000, a valid extension in the 3CX PBX?
     
  7. Soporte Sinet Colombia

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    Hi leejor,

    Yes, it's a valid extension. I'm attaching a log for extension 604 anyway.

    2017/10/17 11:13:47.885228 192.168.21.200:5060 -> 147.122.1.200:5060
    INVITE sip:604@147.122.1.200:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK0af942b7;rport
    Max-Forwards: 70
    From: "Adminfo" <sip:10002@192.168.21.200>;tag=as1e851238
    To: <sip:604@147.122.1.200:5060>
    Contact: <sip:10002@192.168.21.200:5060>

    Call-ID: 354e34275733dfc17cfa853a0e1dfc49@192.168.21.200:5060
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX 1.8.17.0
    Date: Tue, 17 Oct 2017 16:13:47 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Remote-Party-ID: "Adminfo" <sip:900@192.168.21.200>;party=calling;privacy=off;screen=no
    Content-Type: application/sdp
    Content-Length: 286

    v=0
    o=root 139235737 139235737 IN IP4 192.168.21.200
    s=Asterisk PBX 1.8.17.0
    c=IN IP4 192.168.21.200
    t=0 0
    m=audio 17500 RTP/AVP 0 18 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv

    2017/10/17 11:13:48.028263 192.168.21.200:5060 -> 147.122.1.200:5060
    INVITE sip:604@147.122.1.200:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK0af942b7;rport
    Max-Forwards: 70
    From: "Adminfo" <sip:10002@192.168.21.200>;tag=as1e851238
    To: <sip:604@147.122.1.200:5060>
    Contact: <sip:10002@192.168.21.200:5060>
    Call-ID: 354e34275733dfc17cfa853a0e1dfc49@192.168.21.200:5060
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX 1.8.17.0
    Date: Tue, 17 Oct 2017 16:13:47 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Remote-Party-ID: "Adminfo" <sip:900@192.168.21.200>;party=calling;privacy=off;screen=no
    Content-Type: application/sdp
    Content-Length: 286

    v=0
    o=root 139235737 139235737 IN IP4 192.168.21.200
    s=Asterisk PBX 1.8.17.0
    c=IN IP4 192.168.21.200
    t=0 0
    m=audio 17500 RTP/AVP 0 18 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv

    2017/10/17 11:13:48.028761 147.122.1.200:5060 -> 192.168.21.200:5060
    SIP/2.0 407 Proxy Authentication Required
    Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK0af942b7;rport=5060
    Proxy-Authenticate: Digest nonce="414d535959e62c3e38:7092c032f85c525ca9f0a5d56147aedb",algorithm=MD5,
    alm="3CXPhoneSystem"
    To: <sip:604@147.122.1.200:5060>;tag=7c726410
    From: "Adminfo" <sip:10002@192.168.21.200>;tag=as1e851238
    Call-ID: 354e34275733dfc17cfa853a0e1dfc49@192.168.21.200:5060
    CSeq: 102 INVITE
    User-Agent: 3CXPhoneSystem 15.5.3849.1 (3849)
    Content-Length: 0

    2017/10/17 11:13:48.028895 192.168.21.200:5060 -> 147.122.1.200:5060
    ACK sip:604@147.122.1.200:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK0af942b7;rport
    Max-Forwards: 70
    From: "Adminfo" <sip:10002@192.168.21.200>;tag=as1e851238
    To: <sip:604@147.122.1.200:5060>;tag=7c726410
    Contact: <sip:10002@192.168.21.200:5060>
    Call-ID: 354e34275733dfc17cfa853a0e1dfc49@192.168.21.200:5060
    CSeq: 102 ACK
    User-Agent: Asterisk PBX 1.8.17.0
    Content-Length: 0

    2017/10/17 11:13:48.029069 192.168.21.200:5060 -> 147.122.1.200:5060
    INVITE sip:604@147.122.1.200:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK78f4c9de;rport
    Max-Forwards: 70
    From: "Adminfo" <sip:10002@192.168.21.200>;tag=as1e851238
    To: <sip:604@147.122.1.200:5060>
    Contact: <sip:10002@192.168.21.200:5060>
    Call-ID: 354e34275733dfc17cfa853a0e1dfc49@192.168.21.200:5060
    CSeq: 103 INVITE
    User-Agent: Asterisk PBX 1.8.17.0
    Proxy-Authorization: Digest username="10002", realm="3CXPhoneSystem", algorithm=MD5, uri="sip:604@147
    22.1.200:5060", nonce="414d535959e62c3e38:7092c032f85c525ca9f0a5d56147aedb", response="793ff613c03452
    682c776c03942ab8"
    Date: Tue, 17 Oct 2017 16:13:48 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Remote-Party-ID: "Adminfo" <sip:900@192.168.21.200>;party=calling;privacy=off;screen=no
    Content-Type: application/sdp
    Content-Length: 286

    v=0
    o=root 139235737 139235738 IN IP4 192.168.21.200
    s=Asterisk PBX 1.8.17.0
    c=IN IP4 192.168.21.200
    t=0 0
    m=audio 17500 RTP/AVP 0 18 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv

    2017/10/17 11:13:48.046709 147.122.1.200:5060 -> 192.168.21.200:5060
    SIP/2.0 407 Proxy Authentication Required
    Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK0af942b7;rport=5060
    Proxy-Authenticate: Digest nonce="414d535959e62c3e38:7092c032f85c525ca9f0a5d56147aedb",algorithm=MD5,
    alm="3CXPhoneSystem"
    To: <sip:604@147.122.1.200:5060>;tag=7c726410
    From: "Adminfo" <sip:10002@192.168.21.200>;tag=as1e851238
    Call-ID: 354e34275733dfc17cfa853a0e1dfc49@192.168.21.200:5060
    CSeq: 102 INVITE
    User-Agent: 3CXPhoneSystem 15.5.3849.1 (3849)
    Content-Length: 0

    2017/10/17 11:13:48.046830 192.168.21.200:5060 -> 147.122.1.200:5060
    ACK sip:604@147.122.1.200:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK78f4c9de;rport
    Max-Forwards: 70
    From: "Adminfo" <sip:10002@192.168.21.200>;tag=as1e851238
    To: <sip:604@147.122.1.200:5060>
    Contact: <sip:10002@192.168.21.200:5060>
    Call-ID: 354e34275733dfc17cfa853a0e1dfc49@192.168.21.200:5060
    CSeq: 102 ACK
    User-Agent: Asterisk PBX 1.8.17.0
    Content-Length: 0

    2017/10/17 11:13:48.172066 192.168.21.200:5060 -> 147.122.1.200:5060
    INVITE sip:604@147.122.1.200:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK78f4c9de;rport
    Max-Forwards: 70
    From: "Adminfo" <sip:10002@192.168.21.200>;tag=as1e851238
    To: <sip:604@147.122.1.200:5060>
    Contact: <sip:10002@192.168.21.200:5060>
    Call-ID: 354e34275733dfc17cfa853a0e1dfc49@192.168.21.200:5060
    CSeq: 103 INVITE
    User-Agent: Asterisk PBX 1.8.17.0
    Proxy-Authorization: Digest username="10002", realm="3CXPhoneSystem", algorithm=MD5, uri="sip:604@147
    22.1.200:5060", nonce="414d535959e62c3e38:7092c032f85c525ca9f0a5d56147aedb", response="793ff613c03452
    682c776c03942ab8"
    Date: Tue, 17 Oct 2017 16:13:48 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Remote-Party-ID: "Adminfo" <sip:900@192.168.21.200>;party=calling;privacy=off;screen=no
    Content-Type: application/sdp
    Content-Length: 286

    v=0
    o=root 139235737 139235738 IN IP4 192.168.21.200
    s=Asterisk PBX 1.8.17.0
    c=IN IP4 192.168.21.200
    t=0 0
    m=audio 17500 RTP/AVP 0 18 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv

    2017/10/17 11:13:48.184206 147.122.1.200:5060 -> 192.168.21.200:5060
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK78f4c9de;rport=5060
    To: <sip:604@147.122.1.200:5060>
    From: "Adminfo" <sip:10002@192.168.21.200>;tag=as1e851238
    Call-ID: 354e34275733dfc17cfa853a0e1dfc49@192.168.21.200:5060
    CSeq: 103 INVITE
    Content-Length: 0


    2017/10/17 11:13:48.234734 147.122.1.200:5060 -> 192.168.21.200:5060
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK78f4c9de;rport=5060
    To: <sip:604@147.122.1.200:5060>
    From: "Adminfo" <sip:10002@192.168.21.200>;tag=as1e851238
    Call-ID: 354e34275733dfc17cfa853a0e1dfc49@192.168.21.200:5060
    CSeq: 103 INVITE
    Content-Length: 0
     
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