DID Number not set and problem on inbound route

› VitalPBX Community Support › General Discussion › DID Number not set and problem on inbound route

  • Post
    • August 9, 2018 at 4:14 pm
    pdifeo
    Participant

    Hi,

    I have a SIP Trunk with address 50 numbers. 

    The problem is that VitalPBX recognize always the principal number and never the others.

    The number are serial starting from xxxx900 to xxxx949. 

    If I call, for example, xxxx949 never reach inbound route on his number. Always routed to xxxx900.

    Debugging SIP you can see (below) that the TO header have xxxxx949 as contatted number.

    Can anyone help me ?

    Regards
    Pasquale

    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 83.211.227.21:5060;branch=z9hG4bK4005.31fd3927.1;received=83.211.227.21
    Via: SIP/2.0/UDP 62.94.8.72:5060;branch=z9hG4bK.iIiIiI.c0a8aa4b.e7aeb8b5.Acc0270.Tra00070
    Record-Route: <sip:83.211.227.21;lr;ftag=fb2e98c2.Acc270.B2b129;did=dcf.6cc83b94>
    Record-Route: <sip:AcCaPl0270.AcCpNo0629.AcCiSt0129.DiAgOuT00.Outgoing.ModB2B.xCSCF-ApLz00284.PnO00602.IsT00263.iIiIiI.c0a8aa4b@62.94.8.72:5060;transport=UDP;lr>
    From: “xxxxxxxxxx” <sip:xxxxxxxxxx@clouditalia769.it;user=phone>;tag=fb2e98c2.Acc270.B2b129
    To: <sip:xxxxxx949@83.211.227.21:5060;user=phone>
    Call-ID: 3bf11822cbb654d0-Acc270-B2b129@10.15.15.8
    CSeq: 1 INVITE
    Server: VitalPBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Contact: <sip:xxxxxx900@x.x.x.x:5060>
    Content-Length: 0

    0
Viewing 15 replies - 1 through 15 (of 22 total)
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  • Replies
    • August 9, 2018 at 4:28 pm
    Jose Miguel Rivera
    Keymaster
    NI
    Up
    0
    Down

    Please post a full call trace of an incoming call to anlayze it

    0
    • August 9, 2018 at 5:11 pm
    pdifeo
    Participant
    Up
    0
    Down

    Thanks for fast reply. Below the trace from full log with sip debugging set to on. I make hidden the ip address of the server, the CID and at end the called number with xxxxx20949 and trunk principal number with xxxxx20900

    Thank you in advance
    Pasquale

    [2018-08-09 18:59:39] VERBOSE[17592] chan_sip.c:
    <— SIP read from UDP:83.211.227.21:5060 —>
    INVITE sip:xxxxx20900@192.168.200.131:5060 SIP/2.0
    Record-Route: <sip:83.211.227.21;lr;ftag=1dff4b84.Acc269.B2b409;did=94e1.d38c3e24>
    Via: SIP/2.0/UDP 83.211.227.21:5060;branch=z9hG4bK08ef.c8e0d7c5.1
    Via: SIP/2.0/UDP 62.94.8.72:5060;branch=z9hG4bK.iIiIiI.c0a8aa4a.5e8ec29a.Acc0269.Tra00635
    Call-ID: 5b6b3c7cd4e27722-Acc269-B2b409@10.15.15.8
    From: “xxxxxxxxxx” <sip:xxxxxxxxxx@clouditalia769.it;user=phone>;tag=1dff4b84.Acc269.B2b409
    To: <sip:xxxxx20949@83.211.227.21:5060;user=phone>
    CSeq: 1 INVITE
    Contact: <sip:xxxxxxxxxx.iIiIiI.0a0f0e09.@62.94.8.72:5060>
    Record-Route: <sip:AcCaPl0269.AcCpNo0629.AcCiSt0409.DiAgOuT00.Outgoing.ModB2B.xCSCF-ApLz00286.PnO00602.IsT00483.iIiIiI.c0a8aa4a@62.94.8.72:5060;transport=UDP;lr>
    Max-Forwards: 16
    Accept: application/sdp, application/isup, application/xml
    Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, MESSAGE, NOTIFY, REGISTER, INFO, REFER, SUBSCRIBE, PUBLISH, UPDATE
    Content-Length: 327
    Content-Type: application/sdp

    v=0
    o=- 0 1 IN IP4 62.94.199.42
    s=-
    c=IN IP4 62.94.199.42
    t=0 0
    m=audio 56624 RTP/AVP 18 0 3 100 8 101
    a=ptime:20
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:100 CLEARMODE/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15,32-35
    a=sendrecv
    a=silenceSupp:off – – – –
    a=sqn:0
    a=cdsc: 1 image udptl t38
    <————->
    [2018-08-09 18:59:39] VERBOSE[17592] chan_sip.c: — (15 headers 16 lines) —
    [2018-08-09 18:59:39] VERBOSE[17592] chan_sip.c: Sending to 83.211.227.21:5060 (no NAT)
    [2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Sending to 83.211.227.21:5060 (no NAT)
    [2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Using INVITE request as basis request – 5b6b3c7cd4e27722-Acc269-B2b409@10.15.15.8
    [2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Found peer ‘xxxxx20900’ for ‘xxxxxxxxxx’ from 83.211.227.21:5060
    [2018-08-09 18:59:39] VERBOSE[17592][C-00000010] netsock2.c: Using SIP RTP TOS bits 184
    [2018-08-09 18:59:39] VERBOSE[17592][C-00000010] netsock2.c: Using SIP RTP CoS mark 5
    [2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Found RTP audio format 18
    [2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Found RTP audio format 0
    [2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Found RTP audio format 3
    [2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Found RTP audio format 100
    [2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Found RTP audio format 8
    [2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Found RTP audio format 101
    [2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Found audio description format G729 for ID 18
    [2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Found unknown media description format CLEARMODE for ID 100
    [2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Found audio description format telephone-event for ID 101
    [2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Capabilities: us – (ulaw|alaw|g729), peer – audio=(ulaw|gsm|alaw|g729)/video=(nothing)/text=(nothing), combined – (ulaw|alaw|g729)
    [2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Non-codec capabilities (dtmf): us – 0x1 (telephone-event|), peer – 0x1 (telephone-event|), combined – 0x1 (telephone-event|)
    [2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Peer audio RTP is at port 62.94.199.42:56624
    [2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Looking for xxxxx20900 in default-trunk (domain 192.168.200.131)
    [2018-08-09 18:59:39] VERBOSE[17592][C-00000010] sip/route.c: sip_route_dump: route/path hop: <sip:83.211.227.21;lr;ftag=1dff4b84.Acc269.B2b409;did=94e1.d38c3e24>
    [2018-08-09 18:59:39] VERBOSE[17592][C-00000010] sip/route.c: sip_route_dump: route/path hop: <sip:AcCaPl0269.AcCpNo0629.AcCiSt0409.DiAgOuT00.Outgoing.ModB2B.xCSCF-ApLz00286.PnO00602.IsT00483.iIiIiI.c0a8aa4a@62.94.8.72:5060;transport=UDP;lr>
    [2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c:
    <— Transmitting (no NAT) to 83.211.227.21:5060 —>
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 83.211.227.21:5060;branch=z9hG4bK08ef.c8e0d7c5.1;received=83.211.227.21
    Via: SIP/2.0/UDP 62.94.8.72:5060;branch=z9hG4bK.iIiIiI.c0a8aa4a.5e8ec29a.Acc0269.Tra00635
    Record-Route: <sip:83.211.227.21;lr;ftag=1dff4b84.Acc269.B2b409;did=94e1.d38c3e24>
    Record-Route: <sip:AcCaPl0269.AcCpNo0629.AcCiSt0409.DiAgOuT00.Outgoing.ModB2B.xCSCF-ApLz00286.PnO00602.IsT00483.iIiIiI.c0a8aa4a@62.94.8.72:5060;transport=UDP;lr>
    From: “xxxxxxxxxx” <sip:xxxxxxxxxx@clouditalia769.it;user=phone>;tag=1dff4b84.Acc269.B2b409
    To: <sip:xxxxx20949@83.211.227.21:5060;user=phone>
    Call-ID: 5b6b3c7cd4e27722-Acc269-B2b409@10.15.15.8
    CSeq: 1 INVITE
    Server: VitalPBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Contact: <sip:xxxxx20900@xx.xx.xx.xx:5060>
    Content-Length: 0

    <————>
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [xxxxx20900@default-trunk:1] Gosub(“SIP/xxxxx20900-0000000e”, “sub-check-blacklist,s,1(ebe61b6b5987db2a,xxxxxxxxxx)”) in new stack
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s@sub-check-blacklist:1] NoOp(“SIP/xxxxx20900-0000000e”, “Testing if xxxxxxxxxx is in Black List”) in new stack
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s@sub-check-blacklist:2] GotoIf(“SIP/xxxxx20900-0000000e”, “0?banned”) in new stack
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s@sub-check-blacklist:3] Return(“SIP/xxxxx20900-0000000e”, “”) in new stack
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [xxxxx20900@default-trunk:2] Gosub(“SIP/xxxxx20900-0000000e”, “sub-setup-call-type,s,1(incoming)”) in new stack
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s@sub-setup-call-type:1] NoOp(“SIP/xxxxx20900-0000000e”, “Determinating Call Type”) in new stack
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s@sub-setup-call-type:2] GotoIf(“SIP/xxxxx20900-0000000e”, “0?return”) in new stack
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s@sub-setup-call-type:3] Gosub(“SIP/xxxxx20900-0000000e”, “s-incoming,1()”) in new stack
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s-incoming@sub-setup-call-type:1] NoOp(“SIP/xxxxx20900-0000000e”, “Incoming Call”) in new stack
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s-incoming@sub-setup-call-type:2] Set(“SIP/xxxxx20900-0000000e”, “__CALL_TYPE=2”) in new stack
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s-incoming@sub-setup-call-type:3] Return(“SIP/xxxxx20900-0000000e”, “”) in new stack
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s@sub-setup-call-type:4] Set(“SIP/xxxxx20900-0000000e”, “__CALL_TYPE_CONFIGURED=yes”) in new stack
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s@sub-setup-call-type:5] Set(“SIP/xxxxx20900-0000000e”, “CDR(calltype)=2”) in new stack
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s@sub-setup-call-type:6] Return(“SIP/xxxxx20900-0000000e”, “”) in new stack
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [xxxxx20900@default-trunk:3] Goto(“SIP/xxxxx20900-0000000e”, “incoming-
    calls,xxxxx20900,1”) in new stack
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx_builtins.c: Goto (incoming-calls,xxxxx20900,1)
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Channel ‘SIP/xxxxx20900-0000000e’ sent to invalid extension: context,exten,priority=incoming-calls,xxxxx20900,1
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [i@incoming-calls:1] NoCDR(“SIP/xxxxx20900-0000000e”, “”) in new stack
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [i@incoming-calls:2] Goto(“SIP/xxxxx20900-0000000e”, “invalid-dest,s,1”) in new stack
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx_builtins.c: Goto (invalid-dest,s,1)
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s@invalid-dest:1] NoOp(“SIP/xxxxx20900-0000000e”, “Invalid Route to Dial”) in new stack
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s@invalid-dest:2] Playback(“SIP/xxxxx20900-0000000e”, “im-sorry&no-route-exists-to-dest&vm-goodbye”) in new stack
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] chan_sip.c: Audio is at 18750
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] chan_sip.c: Adding codec ulaw to SDP
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] chan_sip.c: Adding codec alaw to SDP
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] chan_sip.c: Adding codec g729 to SDP
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] chan_sip.c:
    <— Reliably Transmitting (no NAT) to 83.211.227.21:5060 —>
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 83.211.227.21:5060;branch=z9hG4bK08ef.c8e0d7c5.1;received=83.211.227.21
    Via: SIP/2.0/UDP 62.94.8.72:5060;branch=z9hG4bK.iIiIiI.c0a8aa4a.5e8ec29a.Acc0269.Tra00635
    Record-Route: <sip:83.211.227.21;lr;ftag=1dff4b84.Acc269.B2b409;did=94e1.d38c3e24>
    Record-Route: <sip:AcCaPl0269.AcCpNo0629.AcCiSt0409.DiAgOuT00.Outgoing.ModB2B.xCSCF-ApLz00286.PnO00602.IsT00483.iIiIiI.c0a8aa4a@62.94.8.72:5060;transport=UDP;lr>
    From: “xxxxxxxxxx” <sip:xxxxxxxxxx@clouditalia769.it;user=phone>;tag=1dff4b84.Acc269.B2b409
    To: <sip:xxxxx20949@83.211.227.21:5060;user=phone>;tag=as619b41ad
    Call-ID: 5b6b3c7cd4e27722-Acc269-B2b409@10.15.15.8
    CSeq: 1 INVITE
    Server: VitalPBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Contact: <sip:xxxxx20900@xx.xx.xx.xx:5060>
    Content-Type: application/sdp
    Content-Length: 321

    v=0
    o=root 86332321 86332321 IN IP4 xx.xx.xx.xx
    s=Asterisk PBX 13.21.0
    c=IN IP4 xx.xx.xx.xx
    t=0 0
    m=audio 18750 RTP/AVP 0 8 18 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/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=maxptime:150
    a=sendrecv

    <————>
    [2018-08-09 18:59:39] VERBOSE[17592] chan_sip.c:
    <— SIP read from UDP:83.211.227.21:5060 —>
    ACK sip:xxxxx20900@192.168.200.131:5060 SIP/2.0
    Record-Route: <sip:83.211.227.21;lr;ftag=1dff4b84.Acc269.B2b409>
    Via: SIP/2.0/UDP 83.211.227.21:5060;branch=z9hG4bK08ef.c8e0d7c5.3
    Via: SIP/2.0/UDP 62.94.8.72:5060;branch=z9hG4bK.iIiIiI.c0a8aa4a.5e8ec29c.Acc0269.Tra00189
    Call-ID: 5b6b3c7cd4e27722-Acc269-B2b409@10.15.15.8
    From: “xxxxxxxxxx” <sip:xxxxxxxxxx@clouditalia769.it;user=phone>;tag=1dff4b84.Acc269.B2b409
    To: <sip:xxxxx20949@83.211.227.21:5060;user=phone>;tag=as619b41ad
    CSeq: 1 ACK
    Contact: <sip:xxxxxxxxxx.iIiIiI.0a0f0e09.@62.94.8.72:5060>
    Max-Forwards: 16
    Content-Length: 0
    P-hint: rr-enforced

    <————->
    [2018-08-09 18:59:39] VERBOSE[17592] chan_sip.c: — (12 headers 0 lines) —
    [2018-08-09 18:59:39] VERBOSE[32030][C-00000010] file.c: <SIP/xxxxx20900-0000000e> Playing ‘im-sorry.ulaw’ (language ‘it’)
    [2018-08-09 18:59:41] VERBOSE[32030][C-00000010] file.c: <SIP/xxxxx20900-0000000e> Playing ‘no-route-exists-to-dest.ulaw’ (language ‘it’)
    [2018-08-09 18:59:43] VERBOSE[32030][C-00000010] file.c: <SIP/xxxxx20900-0000000e> Playing ‘vm-goodbye.ulaw’ (language ‘it’)
    [2018-08-09 18:59:44] VERBOSE[32030][C-00000010] pbx.c: Executing [s@invalid-dest:3] Hangup(“SIP/xxxxx20900-0000000e”, “”) in new stack
    [2018-08-09 18:59:44] VERBOSE[32030][C-00000010] pbx.c: Spawn extension (invalid-dest, s, 3) exited non-zero on ‘SIP/xxxxx20900-0000000e’
    [2018-08-09 18:59:44] VERBOSE[32030][C-00000010] chan_sip.c: Scheduling destruction of SIP dialog ‘5b6b3c7cd4e27722-Acc269-B2b409@10.15.15.8’ in 6400 ms (Method: ACK)
    [2018-08-09 18:59:44] VERBOSE[32030][C-00000010] chan_sip.c: set_destination: Parsing <sip:83.211.227.21;lr;ftag=1dff4b84.Acc269.B2b409;did=94e1.d38c3e24> for address/port to send to
    [2018-08-09 18:59:44] VERBOSE[32030][C-00000010] chan_sip.c: set_destination: set destination to 83.211.227.21:5060
    [2018-08-09 18:59:44] VERBOSE[32030][C-00000010] chan_sip.c: Reliably Transmitting (no NAT) to 83.211.227.21:5060:
    BYE sip:xxxxxxxxxx.iIiIiI.0a0f0e09.@62.94.8.72:5060 SIP/2.0
    Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK590d774f
    Route: <sip:83.211.227.21;lr;ftag=1dff4b84.Acc269.B2b409;did=94e1.d38c3e24>,<sip:AcCaPl0269.AcCpNo0629.AcCiSt0409.DiAgOuT00.Outgoing.ModB2B.xCSCF-ApLz00286.PnO00602.IsT00483.iIiIiI.c0a8aa4a@62.94.8.72:5060;transport=UDP;lr>
    Max-Forwards: 70
    From: <sip:xxxxx20949@83.211.227.21:5060;user=phone>;tag=as619b41ad
    To: “xxxxxxxxxx” <sip:xxxxxxxxxx@clouditalia769.it;user=phone>;tag=1dff4b84.Acc269.B2b409
    Call-ID: 5b6b3c7cd4e27722-Acc269-B2b409@10.15.15.8
    CSeq: 102 BYE
    User-Agent: VitalPBX
    X-Asterisk-HangupCause: Normal Clearing
    X-Asterisk-HangupCauseCode: 16
    Content-Length: 0

    —
    [2018-08-09 18:59:45] VERBOSE[17592] chan_sip.c:
    <— SIP read from UDP:83.211.227.21:5060 —>
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.200.131:5060;branch=z9hG4bK590d774f
    Call-ID: 5b6b3c7cd4e27722-Acc269-B2b409@10.15.15.8
    From: <sip:xxxxx20949@83.211.227.21:5060;user=phone>;tag=as619b41ad
    To: “xxxxxxxxxx” <sip:xxxxxxxxxx@clouditalia769.it;user=phone>;tag=1dff4b84.Acc269.B2b409
    CSeq: 102 BYE
    Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, MESSAGE, NOTIFY, REGISTER, INFO, REFER, SUBSCRIBE, PUBLISH, UPDATE
    P-Charging-Vector: icid-value=”000dda9f7191c54-002b-0286-0000-0000@10.15.14.9″;icid-generated-at=10.15.11.5;orig-ioi=clouditalia769.it
    Content-Length: 0

    <————->
    [2018-08-09 18:59:45] VERBOSE[17592] chan_sip.c: — (9 headers 0 lines) —
    [2018-08-09 18:59:45] VERBOSE[17592][C-00000010] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived
    [2018-08-09 18:59:45] VERBOSE[17592] chan_sip.c: Really destroying SIP dialog ‘5b6b3c7cd4e27722-Acc269-B2b409@10.15.15.8’ Method: ACK
    [2018-08-09 18:59:45] DEBUG[17592] chan_sip.c:
    ———- SIP HISTORY for ‘5b6b3c7cd4e27722-Acc269-B2b409@10.15.15.8’
    [2018-08-09 18:59:45] DEBUG[17592] chan_sip.c: * SIP Call
    [2018-08-09 18:59:45] DEBUG[17592] chan_sip.c: 001. Rx INVITE / 1 INVITE / sip:xxxxx20900@192.168.200.131:5060
    [2018-08-09 18:59:45] DEBUG[17592] chan_sip.c: 002. NewChan Channel SIP/xxxxx20900-0000000e – from 5b6b3c7cd4e27722-Acc269-
    [2018-08-09 18:59:45] DEBUG[17592] chan_sip.c: 003. TxResp SIP/2.0 / 1 INVITE – 100 Trying
    [2018-08-09 18:59:45] DEBUG[17592] chan_sip.c: 004. TxRespRel SIP/2.0 / 1 INVITE – 200 OK
    [2018-08-09 18:59:45] DEBUG[17592] chan_sip.c: 005. Rx ACK / 1 ACK / sip:xxxxx20900@192.168.200.131:5060
    [2018-08-09 18:59:45] DEBUG[17592] chan_sip.c: 006. Hangup Cause Normal Clearing
    [2018-08-09 18:59:45] DEBUG[17592] chan_sip.c: 007. RTCPaudio Quality:ssrc=642619996;themssrc=130626313;lp=0;rxjitter=0.00000
    [2018-08-09 18:59:45] DEBUG[17592] chan_sip.c: 008. TxRe
    qRel BYE / 102 BYE – BYE
    [2018-08-09 18:59:45] DEBUG[17592] chan_sip.c: 009. SchedDestroy 6400 ms
    [2018-08-09 18:59:45] DEBUG[17592] chan_sip.c: 010. Rx SIP/2.0 / 102 BYE / 200 OK
    [2018-08-09 18:59:45] DEBUG[17592] chan_sip.c: 011. NeedDestroy Setting needdestroy because transaction completed
    [2018-08-09 18:59:45] DEBUG[17592] chan_sip.c:
    ———- END SIP HISTORY for ‘5b6b3c7cd4e27722-Acc269-B2b409@10.15.15.8’

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    • August 9, 2018 at 5:15 pm
    Jose Miguel Rivera
    Keymaster
    NI
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    I was talking about a call log, like in the attachment

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    • August 9, 2018 at 5:23 pm
    pdifeo
    Participant
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    Ok, I disabled SIP debugging and now there are only the information requested. The important think is that I called xxxxx20949 not xxxxx20900

    pbx.c: Executing [xxxxx20900@default-trunk:1] Gosub("SIP/xxxxx20900-0000000f", "sub-check-blacklist,s,1(ebe61b6b5987db2a,xxxxxxxxxx)") in new stack
    pbx.c: Executing [s@sub-check-blacklist:1] NoOp("SIP/xxxxx20900-0000000f", "Testing if xxxxxxxxxx is in Black List") in new stack
    pbx.c: Executing [s@sub-check-blacklist:2] GotoIf("SIP/xxxxx20900-0000000f", "0?banned") in new stack
    pbx.c: Executing [s@sub-check-blacklist:3] Return("SIP/xxxxx20900-0000000f", "") in new stack
    pbx.c: Executing [xxxxx20900@default-trunk:2] Gosub("SIP/xxxxx20900-0000000f", "sub-setup-call-type,s,1(incoming)") in new stack
    pbx.c: Executing [s@sub-setup-call-type:1] NoOp("SIP/xxxxx20900-0000000f", "Determinating Call Type") in new stack
    pbx.c: Executing [s@sub-setup-call-type:2] GotoIf("SIP/xxxxx20900-0000000f", "0?return") in new stack
    pbx.c: Executing [s@sub-setup-call-type:3] Gosub("SIP/xxxxx20900-0000000f", "s-incoming,1()") in new stack
    pbx.c: Executing [s-incoming@sub-setup-call-type:1] NoOp("SIP/xxxxx20900-0000000f", "Incoming Call") in new stack
    pbx.c: Executing [s-incoming@sub-setup-call-type:2] Set("SIP/xxxxx20900-0000000f", "__CALL_TYPE=2") in new stack
    pbx.c: Executing [s-incoming@sub-setup-call-type:3] Return("SIP/xxxxx20900-0000000f", "") in new stack
    pbx.c: Executing [s@sub-setup-call-type:4] Set("SIP/xxxxx20900-0000000f", "__CALL_TYPE_CONFIGURED=yes") in new stack
    pbx.c: Executing [s@sub-setup-call-type:5] Set("SIP/xxxxx20900-0000000f", "CDR(calltype)=2") in new stack
    pbx.c: Executing [s@sub-setup-call-type:6] Return("SIP/xxxxx20900-0000000f", "") in new stack
    pbx.c: Executing [xxxxx20900@default-trunk:3] Goto("SIP/xxxxx20900-0000000f", "incoming-calls,xxxxx20900,1") in new stack
    pbx_builtins.c: Goto (incoming-calls,xxxxx20900,1)
    pbx.c: Channel 'SIP/xxxxx20900-0000000f' sent to invalid extension: context,exten,priority=incoming-calls,xxxxx20900,1
    pbx.c: Executing [i@incoming-calls:1] NoCDR("SIP/xxxxx20900-0000000f", "") in new stack
    pbx.c: Executing [i@incoming-calls:2] Goto("SIP/xxxxx20900-0000000f", "invalid-dest,s,1") in new stack
    pbx_builtins.c: Goto (invalid-dest,s,1)
    pbx.c: Executing [s@invalid-dest:1] NoOp("SIP/xxxxx20900-0000000f", "Invalid Route to Dial") in new stack
    pbx.c: Executing [s@invalid-dest:2] Playback("SIP/xxxxx20900-0000000f", "im-sorry&no-route-exists-to-dest&vm-goodbye") in new stack
    file.c: <SIP/xxxxx20900-0000000f> Playing 'im-sorry.ulaw' (language 'it')
    file.c: <SIP/xxxxx20900-0000000f> Playing 'no-route-exists-to-dest.ulaw' (language 'it')
    file.c: <SIP/xxxxx20900-0000000f> Playing 'vm-goodbye.ulaw' (language 'it')
    pbx.c: Executing [s@invalid-dest:3] Hangup("SIP/xxxxx20900-0000000f", "") in new stack
    pbx.c: Spawn extension (invalid-dest, s, 3) exited non-zero on 'SIP/xxxxx20900-0000000f'
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    • August 9, 2018 at 5:27 pm
    pdifeo
    Participant
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    And here the inbound route

     

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    • August 9, 2018 at 5:36 pm
    Jose Miguel Rivera
    Keymaster
    NI
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    Check your VOIP provider configurations, that’s what your provider sent

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    • August 9, 2018 at 5:59 pm
    pdifeo
    Participant
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    Ok,

    but for make better answer to voip provider can you tell me what he must send ? So, what is missing in SIP conversation ?

    Thanks in advance.

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    • August 9, 2018 at 6:22 pm
    Jose Miguel Rivera
    Keymaster
    NI
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    They are sending the DID number in the asterisk header, specifically in the “To” parameter. You can send them the first call debug log to clarify

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    • August 9, 2018 at 6:28 pm
    pdifeo
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    Sorry, I’m confused. If you read again in the SIP parameters 

    To: <sip:xxxxx20949@83.211.227.21:5060;user=phone>

    there is the correct number that I want that go in right inbound route.

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    • August 9, 2018 at 6:30 pm
    Jose Miguel Rivera
    Keymaster
    NI
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    Yes, is the right number, but is not being sending as DID Number

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    • August 9, 2018 at 6:36 pm
    pdifeo
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    I don’t want annoyng you, maybe is my poor English and I don’t ask in correct way and don’t understand better what you told me.

    You first told me “They are sending the DID number in the asterisk header, specifically in the “To” parameter.” And in the To parameter there is the correct number .

    After told me “but is not being sending as DID Number”.

    My question is: where must exist the DID number ?

    Please can you give me a little example.

    Regards
    Pasquale

     

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    • August 9, 2018 at 6:47 pm
    Jose Miguel Rivera
    Keymaster
    NI
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    Simple, they are sending the Always the same DID Number (Main Number – xxxx900).

    The DID that you want to receive is being sent as a Header Parameter.

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    • August 9, 2018 at 7:13 pm
    pdifeo
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    Also my question is simple: where ? What is the header parameter ? INVITE ? 

    I found a similar problem with a FreePBX based installation at link

    https://sites.google.com/site/samsig/techie/How-to-get-the-DID-of-a-SIP-trunk-when-the-provider-doesnt-send-it-and-why-some-incoming-SIP-calls-fail

    But I’m new to VitalPBX and I don’t know where put the modification.

    I’m surrending. I’d like VitalPBX and I also wanted to buy some addons, but if I not solve this problem not is the right product for me.

    Thanks in any case.

    Regards

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    • August 9, 2018 at 7:28 pm
    Jose Miguel Rivera
    Keymaster
    NI
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    I told you, the DID number is not send as usally, is send as Header Parameters:

    From: “xxxxxxxxxx” <sip:xxxxxxxxxx@clouditalia769.it;user=phone>;tag=1dff4b84.Acc269.B2b409
    To: <sip:xxxxx20949@83.211.227.21:5060;user=phone>

    The kind of behavior that is described in the link that you sent, can be implemented in future versions of VitalPBX

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    • August 10, 2018 at 11:50 am
    pdifeo
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    Hi,

    I found that in FreePBX the workaround is very simple. Put context=from-pstn-toheader into trunk definition and all work fine. This call 

    [from-pstn-toheader]
    exten => _.,1,Goto(from-pstn,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1)

    I hope that this simple solution will be adopted as soon as possible in your product. In the while I must use other PBX solutions.

    Regards
    Pasquale

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