Reply To: RE: VitalPBX 2.3.1-3 Inbound problem

VitalPBX Community Support General Discussion VitalPBX 2.3.1-3 Inbound problem Reply To: RE: VitalPBX 2.3.1-3 Inbound problem

    voiprehberi
    Participant

    — Executing [t@IVR-1:4] Goto(“SIP/-0000bfb7”, “s,retry”) in new stack
    — Goto (IVR-1,s,9)
    — Executing [s@IVR-1:9] NoOp(“SIP/-0000bfb7”, “IVR Retry Section”) in new stack
    — Executing [s@IVR-1:10] WaitExten(“SIP/-0000bfb7”, “10”) in new stack
    [2019-04-23 21:54:37] WARNING[4642]: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission 142793644-1083765191-1346616516 for seqno 1 (Critical Response) — See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 32000ms with no response
    Really destroying SIP dialog ‘142793644-1083765191-1346616516’ Method: INVITE
    — Executing [s@IVR-1:9] NoOp(“SIP/-0000bfbe”, “IVR Retry Section”) in new stack
    — Executing [s@IVR-1:10] WaitExten(“SIP/-0000bfbe”, “10”) in new stack
    Retransmitting #2 (no NAT) to 46.166.151.160:59781:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 46.166.151.160:59781;branch=z9hG4bK15212067;received=46.166.151.160
    From: <sip:voip53@>;tag=121300696
    To: <sip:00441294507632@>;tag=as4b4e395c
    Call-ID: 382367288-1200743055-782559463
    CSeq: 1 INVITE
    Server: VitalPBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Contact: <sip:00441294507632@:5060>
    Content-Type: application/sdp
    Content-Length: 275

    v=0
    o=root 890010195 890010195 IN IP4
    s=Asterisk PBX 16.2.1
    c=IN IP4
    t=0 0
    m=audio 17046 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=maxptime:150
    a=sendrecv


    Retransmitting #3 (no NAT) to 46.166.151.160:56540:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 46.166.151.160:56540;branch=z9hG4bK652342916;received=46.166.151.160
    From: <sip:Pamtel@>;tag=748218597
    To: <sip:00441254929805@>;tag=as1d9a9cf2
    Call-ID: 1408826927-1415972875-793910904
    CSeq: 1 INVITE
    Server: VitalPBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Contact: <sip:00441254929805@:5060>
    Content-Type: application/sdp
    Content-Length: 275

    v=0
    o=root 688498921 688498921 IN IP4
    s=Asterisk PBX 16.2.1
    c=IN IP4
    t=0 0
    m=audio 19312 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=maxptime:150
    a=sendrecv


    Retransmitting #7 (no NAT) to 46.166.151.160:58244:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 46.166.151.160:58244;branch=z9hG4bK366944290;received=46.166.151.160
    From: <sip:RPT8@>;tag=1455925790
    To: <sip:00441244739005@>;tag=as6b873e37
    Call-ID: 1373145822-521514607-2106744764
    CSeq: 1 INVITE
    Server: VitalPBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Contact: <sip:00441244739005@:5060>
    Content-Type: application/sdp
    Content-Length: 275

    v=0
    o=root 595295727 595295727 IN IP4
    s=Asterisk PBX 16.2.1
    c=IN IP4
    t=0 0
    m=audio 13972 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=maxptime:150
    a=sendrecv


    Retransmitting #4 (no NAT) to 46.166.151.160:52046:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 46.166.151.160:52046;branch=z9hG4bK1229374470;received=46.166.151.160
    From: <sip:RPT8@>;tag=544882253
    To: <sip:+441244739005@>;tag=as7e37825c
    Call-ID: 961702511-518768254-1699657564
    CSeq: 1 INVITE
    Server: VitalPBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Contact: <sip:+441244739005@:5060>
    Content-Type: application/sdp
    Content-Length: 273

    v=0
    o=root 36437634 36437634 IN IP4
    s=Asterisk PBX 16.2.1
    c=IN IP4
    t=0 0
    m=audio 19904 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=maxptime:150
    a=sendrecv


    Retransmitting #8 (no NAT) to 46.166.151.160:54153:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 46.166.151.160:54153;branch=z9hG4bK1883151552;received=46.166.151.160
    From: <sip:FkfFlw@>;tag=983250134
    To: <sip:9011441254929805@>;tag=as1af0b9e5
    Call-ID: 782015512-878982276-2075579793
    CSeq: 1 INVITE
    Server: VitalPBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Contact: <sip:9011441254929805@:5060>
    Content-Type: application/sdp
    Content-Length: 277

    v=0
    o=root 1720900573 1720900573 IN IP4
    s=Asterisk PBX 16.2.1
    c=IN IP4
    t=0 0
    m=audio 14636 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=maxptime:150
    a=sendrecv


    — Executing [t@IVR-1:4] Goto(“SIP/-0000bfba”, “s,retry”) in new stack
    — Goto (IVR-1,s,9)
    — Executing [s@IVR-1:9] NoOp(“SIP/-0000bfba”, “IVR Retry Section”) in new stack
    — Executing [s@IVR-1:10] WaitExten(“SIP/-0000bfba”, “10”) in new stack
    Retransmitting #3 (no NAT) to 46.166.151.160:59781:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 46.166.151.160:59781;branch=z9hG4bK15212067;received=46.166.151.160
    From: <sip:voip53@>;tag=121300696
    To: <sip:00441294507632@>;tag=as4b4e395c
    Call-ID: 382367288-1200743055-782559463
    CSeq: 1 INVITE
    Server: VitalPBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Contact: <sip:00441294507632@:5060>
    Content-Type: application/sdp
    Content-Length: 275

    v=0
    o=root 890010195 890010195 IN IP4
    s=Asterisk PBX 16.2.1
    c=IN IP4
    t=0 0
    m=audio 17046 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=maxptime:150
    a=sendrecv


    Retransmitting #6 (no NAT) to 46.166.151.160:54887:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 46.166.151.160:54887;branch=z9hG4bK622044709;received=46.166.151.160
    From: <sip:voip53@>;tag=1242532760
    To: <sip:011441294507632@>;tag=as36b618fb
    Call-ID: 1611942025-2014181121-2131988889
    CSeq: 1 INVITE
    Server: VitalPBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Contact: <sip:011441294507632@:5060>
    Content-Type: application/sdp
    Content-Length: 277

    v=0
    o=root 1791862187 1791862187 IN IP4
    s=Asterisk PBX 16.2.1
    c=IN IP4
    t=0 0
    m=audio 11996 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=maxptime:150
    a=sendrecv


    [2019-04-23 21:54:40] NOTICE[4642]: chan_sip.c:29854 check_rtp_timeout: Disconnecting call ‘SIP/-0000bfb7’ for lack of RTP activity in 31 seconds
    == Spawn extension (IVR-1, s, 10) exited non-zero on ‘SIP/-0000bfb7’
    Scheduling destruction of SIP dialog ‘1475841371-440628607-713708505’ in 32000 ms (Method: INVITE)
    — Executing [t@IVR-1:4] Goto(“SIP/-0000bfbb”, “s,retry”) in new stack
    — Goto (IVR-1,s,9)
    — Executing [s@IVR-1:9] NoOp(“SIP/-0000bfbb”, “IVR Retry Section”) in new stack
    — Executing [s@IVR-1:10] WaitExten(“SIP/-0000bfbb”, “10”) in new stack
    Retransmitting #10 (no NAT) to 46.166.151.160:61042:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 46.166.151.160:61042;branch=z9hG4bK1432211764;received=46.166.151.160
    From: <sip:DI
    RECPAT@>;tag=1753243728
    To: <sip:9011441294507632@>;tag=as770e5708
    Call-ID: 1475841371-440628607-713708505
    CSeq: 1 INVITE
    Server: VitalPBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Contact: <sip:9011441294507632@:5060>
    Content-Type: application/sdp
    Content-Length: 275

    v=0
    o=root 952701884 952701884 IN IP4
    s=Asterisk PBX 16.2.1
    c=IN IP4
    t=0 0
    m=audio 12456 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=maxptime:150
    a=sendrecv


    — Timeout on SIP/-0000bfb8, going to ‘t’
    — Executing [t@IVR-1:1] Set(“SIP/-0000bfb8”, “TIMEOUTATTEMPTS=2”) in new stack
    — Executing [t@IVR-1:2] GotoIf(“SIP/-0000bfb8”, “0?timeout”) in new stack
    — Executing [t@IVR-1:3] BackGround(“SIP/-0000bfb8”, “option-is-invalid”) in new stack
    — <SIP/-0000bfb8> Playing ‘option-is-invalid.ulaw’ (language ‘en’)
    Retransmitting #6 (no NAT) to 46.166.151.160:56690:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 46.166.151.160:56690;branch=z9hG4bK582921204;received=46.166.151.160
    From: <sip:Pamtel@>;tag=459103237
    To: <sip:011441254929805@>;tag=as731147c4
    Call-ID: 1776749608-1698815950-2043857812
    CSeq: 1 INVITE
    Server: VitalPBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Contact: <sip:011441254929805@:5060>
    Content-Type: application/sdp
    Content-Length: 277

    v=0
    o=root 1462762709 1462762709 IN IP4
    s=Asterisk PBX 16.2.1
    c=IN IP4
    t=0 0
    m=audio 18450 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=maxptime:150
    a=sendrecv

    0