VitalPBX 2.3.1-3 Inbound problem

VitalPBX Community Support General Discussion VitalPBX 2.3.1-3 Inbound problem

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    voiprehberi
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    Setup of the trunk:

     

    I enabled the allow guest disabled strict rtp.

     

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    voiprehberi
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    Hi,

    When i look at the asterisk -rvv logs it says “”im-sorry&no-route-exists-to-dest&vm-goodbye”

    On the ITSP side everything is ok. Please help !

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    May you share a detailed call log (From Asterisk CLI) of an incoming call to your DID?

    Important Note: Hide any sensitive data.

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    voiprehberi
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    Executing [s@sub-setup-call-type:1] NoOp(“SIP/-0000a805”, “Determinating Call Type”) in new stack
    — Executing [s@sub-setup-call-type:2] GotoIf(“SIP/-0000a805”, “0?return”) in new stack
    — Executing [s@sub-setup-call-type:3] Gosub(“SIP/-0000a805”, “s-incoming,1()”) in new stack
    — Executing [s-incoming@sub-setup-call-type:1] NoOp(“SIP/-0000a805”, “Incoming Call”) in new stack
    — Executing [s-incoming@sub-setup-call-type:2] Set(“SIP/-0000a805”, “__CALL_TYPE=2”) in new stack
    — Executing [s-incoming@sub-setup-call-type:3] Return(“SIP/-0000a805”, “”) in new stack
    — Executing [s@sub-setup-call-type:4] Set(“SIP/-0000a805”, “__CALL_TYPE_CONFIGURED=yes”) in new stack
    — Executing [s@sub-setup-call-type:5] Set(“SIP/-0000a805”, “CDR(calltype)=2”) in new stack
    — Executing [s@sub-setup-call-type:6] Return(“SIP/-0000a805”, “”) in new stack
    — Executing [00441244739005@default-trunk:4] Gosub(“SIP/-0000a805”, “dynamic-routing-in,s,1(cvcsexapp)”) in new stack
    — Executing [s@dynamic-routing-in:1] NoOp(“SIP/-0000a805”, “Test if must to apply dynamic routing”) in new stack
    — Executing [s@dynamic-routing-in:2] Set(“SIP/-0000a805”, “EXTERNAL_CALLER=cvcsexapp”) in new stack
    — Executing [s@dynamic-routing-in:3] Set(“SIP/-0000a805”, “DYNROUTING_DM=0”) in new stack
    — Executing [s@dynamic-routing-in:4] GotoIf(“SIP/-0000a805”, “1?gd”) in new stack
    — Goto (dynamic-routing-in,s,6)
    — Executing [s@dynamic-routing-in:6] GotoIf(“SIP/-0000a805”, “0?:rb”) in new stack
    — Goto (dynamic-routing-in,s,11)
    — Executing [s@dynamic-routing-in:11] Return(“SIP/-0000a805”, “”) in new stack
    — Executing [00441244739005@default-trunk:5] Goto(“SIP/-0000a805”, “incoming-calls,00441244739005,1”) in new stack
    — Goto (incoming-calls,00441244739005,1)
    — Channel ‘SIP/-0000a805’ sent to invalid extension: context,exten,priority=incoming-calls,00441244739005,1
    — Executing [i@incoming-calls:1] NoCDR(“SIP/-0000a805”, “”) in new stack
    — Executing [i@incoming-calls:2] Goto(“SIP/-0000a805”, “invalid-dest,s,1”) in new stack
    — Goto (invalid-dest,s,1)
    — Executing [s@invalid-dest:1] NoOp(“SIP/-0000a805”, “Invalid Route to Dial”) in new stack
    — Executing [s@invalid-dest:2] Playback(“SIP/-0000a805”, “im-sorry&no-route-exists-to-dest&vm-goodbye”) in new stack
    — <SIP/-0000a804> Playing ‘no-route-exists-to-dest.ulaw’ (language ‘en’)
    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    — Executing [011441294507632@sip-default:1] NoOp(“SIP/-0000a806”, “Guest call to DID 011441294507632”) in new stack
    — Executing [011441294507632@sip-default:2] Goto(“SIP/-0000a806”, “default-trunk,011441294507632,1”) in new stack
    — Goto (default-trunk,011441294507632,1)
    — Executing [011441294507632@default-trunk:1] Gosub(“SIP/-0000a806”, “set-global-tenant-vars,s,1”) in new stack
    — Executing [s@set-global-tenant-vars:1] NoOp(“SIP/-0000a806”, “Setting Global Vars for vitalpbx Tenant”) in new stack
    — Executing [s@set-global-tenant-vars:2] Set(“SIP/-0000a806”, “__TENANT_PATH=1cea30e5c2dd6ca7”) in new stack
    — Executing [s@set-global-tenant-vars:3] Set(“SIP/-0000a806”, “__TENANT_PREFIX=”) in new stack
    — Executing [s@set-global-tenant-vars:4] Set(“SIP/-0000a806”, “__QUEUE_AGENTS_CONTEXT=queue-call-to-agents”) in new stack
    — Executing [s@set-global-tenant-vars:5] Set(“SIP/-0000a806”, “__FOLLOWME_CONTEXT=ext-followme”) in new stack
    — Executing [s@set-global-tenant-vars:6] Set(“SIP/-0000a806”, “__HINTS_CONTEXT=extension-hints”) in new stack
    — Executing [s@set-global-tenant-vars:7] Return(“SIP/-0000a806”, “”) in new stack
    — Executing [011441294507632@default-trunk:2] Gosub(“SIP/-0000a806”, “sub-check-blacklist,s,1(1cea30e5c2dd6ca7,Tropez)”) in new stack
    — Executing [s@sub-check-blacklist:1] NoOp(“SIP/-0000a806”, “Testing if Tropez is in Black List”) in new stack
    — Executing [s@sub-check-blacklist:2] GotoIf(“SIP/-0000a806”, “0?banned”) in new stack
    — Executing [s@sub-check-blacklist:3] Return(“SIP/-0000a806”, “”) in new stack
    — Executing [011441294507632@default-trunk:3] Gosub(“SIP/-0000a806”, “sub-setup-call-type,s,1(incoming)”) in new stack
    — Executing [s@sub-setup-call-type:1] NoOp(“SIP/-0000a806”, “Determinating Call Type”) in new stack
    — Executing [s@sub-setup-call-type:2] GotoIf(“SIP/-0000a806”, “0?return”) in new stack
    — Executing [s@sub-setup-call-type:3] Gosub(“SIP/-0000a806”, “s-incoming,1()”) in new stack
    — Executing [s-incoming@sub-setup-call-type:1] NoOp(“SIP/-0000a806”, “Incoming Call”) in new stack
    — Executing [s-incoming@sub-setup-call-type:2] Set(“SIP/-0000a806”, “__CALL_TYPE=2”) in new stack
    — Executing [s-incoming@sub-setup-call-type:3] Return(“SIP/-0000a806”, “”) in new stack
    — Executing [s@sub-setup-call-type:4] Set(“SIP/-0000a806”, “__CALL_TYPE_CONFIGURED=yes”) in new stack
    — Executing [s@sub-setup-call-type:5] Set(“SIP/-0000a806”, “CDR(calltype)=2”) in new stack
    — Executing [s@sub-setup-call-type:6] Return(“SIP/-0000a806”, “”) in new stack
    — Executing [011441294507632@default-trunk:4] Gosub(“SIP/-0000a806”, “dynamic-routing-in,s,1(Tropez)”) in new stack
    — Executing [s@dynamic-routing-in:1] NoOp(“SIP/-0000a806”, “Test if must to apply dynamic routing”) in new stack
    — Executing [s@dynamic-routing-in:2] Set(“SIP/-0000a806”, “EXTERNAL_CALLER=Tropez”) in new stack
    — Executing [s@dynamic-routing-in:3] Set(“SIP/-0000a806”, “DYNROUTING_DM=0”) in new stack
    — Executing [s@dynamic-routing-in:4] GotoIf(“SIP/-0000a806”, “1?gd”) in new stack
    — Goto (dynamic-routing-in,s,6)
    — Executing [s@dynamic-routing-in:6] GotoIf(“SIP/-0000a806”, “0?:rb”) in new stack
    — Goto (dynamic-routing-in,s,11)
    — Executing [s@dynamic-routing-in:11] Return(“SIP/-0000a806”, “”) in new stack
    — Executing [011441294507632@default-trunk:5] Goto(“SIP/-0000a806”, “incoming-calls,011441294507632,1”) in new stack
    — Goto (incoming-calls,011441294507632,1)
    — Channel ‘SIP/-0000a806’ sent to invalid extension: context,exten,priority=incoming-calls,011441294507632,1
    — Executing [i@incoming-calls:1] NoCDR(“SIP/-0000a806”, “”) in new stack
    — Executing [i@incoming-calls:2] Goto(“SIP/-0000a806”, “invalid-dest,s,1”) in new stack
    — Goto (invalid-dest,s,1)
    — Executing [s@invalid-dest:1] NoOp(“SIP/-0000a806”, “Invalid Route to Dial”) in new stack
    — Executing [s@invalid-dest:2] Playback(“SIP/-0000a806”, “im-sorry&no-route-exists-to-dest&vm-goodbye”) in new stack

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    VitalPBX is receiving unknown DIDs: 00441244739005, 011441294507632

    You have two options:

    1. Create an Any / Any item on the Inbound Routes to handle unknown DIDs (Check the example attached)
    2. Enable SIP debug and check if your provider is sending the DID on SIP headers (You may share the sip debug here if you want)

    To enable sip debug, execute the following command in the asterisk CLI: sip set debug on, after enable the SIP debug, you must to make a call to your DID and check what is sending.

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    voiprehberi
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    Nothing is changed when i add the any inbound route.

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    May you share the call log again, just to analyze why is not going to any / any?

    Do you have enabled the option “allow sip guest”?

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    voiprehberi
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    Yes i did. Enable the allowsip guest. 

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    Did you already apply changes after create the Inbound Route?

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    voiprehberi
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    Yes I did.

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    May you share a call log from Asterisk CLI of an Incoming call to your DID?

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    voiprehberi
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    — 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

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    According to the log that you have posted, the call is forward to an IVR, is that right?

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    voiprehberi
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    Yes, exactly

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    voiprehberi
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    There is a nat problem here i think. Can you help from remote please ?

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