New did no route?

› VitalPBX Community Support › General Discussion › New did no route?

  • Post
    • February 20, 2019 at 7:44 am
    Ofloo
    Participant

    I’ve got a new did number added it to inbound calls setup the target extension, however when i call to it it says no route exists?

    Checked dialplan:

    exten => _3214xxxxxx/_1001,1,NoOp(INBOUND_ROUTE: 014xxxxxx)
    same => n,Set(CHANNEL(language)=en)
    same => n,Set(CHANNEL(musicclass)=default)
    same => n,SIPAddHeader(Alert-Info:014xxxxxx)
    same => n,Gosub(sub-set-call-vars,s-incoming,1(${CALLERID(num)},${EXTEN},2a4e48aa120a8d8c))
    same => n,Set(_APP_RECORDING=yes)
    same => n,Gosub(cid-lookup,CIDLOOKUP-1,1)
    same => n,Set(ICALL=yes)
    same => n,Goto(cos-all,1001,1)
    same => n,Hangup()

    Result:

     -- Executing [s@invalid-dest:1] NoOp("SIP/xxxxxx-00000040", "Invalid Route to Dial") in new stack
    -- Executing [s@invalid-dest:2] Playback("SIP/xxxxxx-00000040", "im-sorry&no-route-exists-to-dest&vm-goodbye") in new stack
    -- <SIP/xxxxxx-00000040> Playing 'im-sorry.ulaw' (language 'en')
    -- <SIP/xxxxxx-00000040> Playing 'no-route-exists-to-dest.ulaw' (language 'en')
    -- <SIP/xxxxxx-00000040> Playing 'vm-goodbye.ulaw' (language 'en')
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Viewing 6 replies - 1 through 6 (of 6 total)
  • Replies
    • February 20, 2019 at 2:47 pm
    Jose Miguel Rivera
    Keymaster
    NI
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    May you share the full call log?

    0
    • February 21, 2019 at 5:32 am
    Ofloo
    Participant
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    # asterisk -rvvv
    Asterisk 13.23.1, Copyright (C) 1999 - 2014, Digium, Inc. and others.
    Created by Mark Spencer <markster@digium.com>
    Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
    This is free software, with components licensed under the GNU General Public
    License version 2 and other licenses; you are welcome to redistribute it under
    certain conditions. Type 'core show license' for details.
    =========================================================================
    Connected to Asterisk 13.23.1 currently running on pbx (pid = 2940)
      == Using SIP RTP TOS bits 184
      == Using SIP RTP CoS mark 5
        -- Executing [SIPIDSRV@trk-6-in:1] NoOp("SIP/SIPIDSRV-00000045", "Incoming call through: PROVIDERNAME") in new stack
        -- Executing [SIPIDSRV@trk-6-in:2] Set("SIP/SIPIDSRV-00000045", "DID=SIPIDSRV") in new stack
        -- Executing [SIPIDSRV@trk-6-in:3] Goto("SIP/SIPIDSRV-00000045", "default-trunk,SIPIDSRV,1") in new stack
        -- Goto (default-trunk,SIPIDSRV,1)
        -- Executing [SIPIDSRV@default-trunk:1] Gosub("SIP/SIPIDSRV-00000045", "sub-check-blacklist,s,1(2a4e48aa120a8d8c,SOURCEPHONE)") in new stack
        -- Executing [s@sub-check-blacklist:1] NoOp("SIP/SIPIDSRV-00000045", "Testing if SOURCEPHONE is in Black List") in new stack
        -- Executing [s@sub-check-blacklist:2] GotoIf("SIP/SIPIDSRV-00000045", "0?banned") in new stack
        -- Executing [s@sub-check-blacklist:3] Return("SIP/SIPIDSRV-00000045", "") in new stack
        -- Executing [SIPIDSRV@default-trunk:2] Gosub("SIP/SIPIDSRV-00000045", "sub-setup-call-type,s,1(incoming)") in new stack
        -- Executing [s@sub-setup-call-type:1] NoOp("SIP/SIPIDSRV-00000045", "Determinating Call Type") in new stack
        -- Executing [s@sub-setup-call-type:2] GotoIf("SIP/SIPIDSRV-00000045", "0?return") in new stack
        -- Executing [s@sub-setup-call-type:3] Gosub("SIP/SIPIDSRV-00000045", "s-incoming,1()") in new stack
        -- Executing [s-incoming@sub-setup-call-type:1] NoOp("SIP/SIPIDSRV-00000045", "Incoming Call") in new stack
        -- Executing [s-incoming@sub-setup-call-type:2] Set("SIP/SIPIDSRV-00000045", "__CALL_TYPE=2") in new stack
        -- Executing [s-incoming@sub-setup-call-type:3] Return("SIP/SIPIDSRV-00000045", "") in new stack
        -- Executing [s@sub-setup-call-type:4] Set("SIP/SIPIDSRV-00000045", "__CALL_TYPE_CONFIGURED=yes") in new stack
        -- Executing [s@sub-setup-call-type:5] Set("SIP/SIPIDSRV-00000045", "CDR(calltype)=2") in new stack
        -- Executing [s@sub-setup-call-type:6] Return("SIP/SIPIDSRV-00000045", "") in new stack
        -- Executing [SIPIDSRV@default-trunk:3] Gosub("SIP/SIPIDSRV-00000045", "dynamic-routing-in,s,1(SOURCEPHONE)") in new stack
        -- Executing [s@dynamic-routing-in:1] NoOp("SIP/SIPIDSRV-00000045", "Test if must to apply dynamic routing") in new stack
        -- Executing [s@dynamic-routing-in:2] Set("SIP/SIPIDSRV-00000045", "EXTERNAL_CALLER=SOURCEPHONE") in new stack
        -- Executing [s@dynamic-routing-in:3] GotoIf("SIP/SIPIDSRV-00000045", "1?gd") in new stack
        -- Goto (dynamic-routing-in,s,5)
        -- Executing [s@dynamic-routing-in:5] GotoIf("SIP/SIPIDSRV-00000045", "0?:rb") in new stack
        -- Goto (dynamic-routing-in,s,10)
        -- Executing [s@dynamic-routing-in:10] Return("SIP/SIPIDSRV-00000045", "") in new stack
        -- Executing [SIPIDSRV@default-trunk:4] Goto("SIP/SIPIDSRV-00000045", "incoming-calls,SIPIDSRV,1") in new stack
        -- Goto (incoming-calls,SIPIDSRV,1)
        -- Channel 'SIP/SIPIDSRV-00000045' sent to invalid extension: context,exten,priority=incoming-calls,SIPIDSRV,1
        -- Executing [i@incoming-calls:1] NoCDR("SIP/SIPIDSRV-00000045", "") in new stack
        -- Executing [i@incoming-calls:2] Goto("SIP/SIPIDSRV-00000045", "invalid-dest,s,1") in new stack
        -- Goto (invalid-dest,s,1)
        -- Executing [s@invalid-dest:1] NoOp("SIP/SIPIDSRV-00000045", "Invalid Route to Dial") in new stack
        -- Executing [s@invalid-dest:2] Playback("SIP/SIPIDSRV-00000045", "im-sorry&no-route-exists-to-dest&vm-goodbye") in new stack
        -- <SIP/SIPIDSRV-00000045> Playing 'im-sorry.ulaw' (language 'en')
        -- <SIP/SIPIDSRV-00000045> Playing 'no-route-exists-to-dest.ulaw' (language 'en')
        -- <SIP/SIPIDSRV-00000045> Playing 'vm-goodbye.ulaw' (language 'en')
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    • February 21, 2019 at 3:31 pm
    Jose Miguel Rivera
    Keymaster
    NI
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    Is it defined the “SIPIDSRV” DID on your Inbound Routes?

    Did you already apply changes after DID creation?

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    • February 21, 2019 at 4:21 pm
    Ofloo
    Participant
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    Yes

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    • February 21, 2019 at 4:26 pm
    Jose Miguel Rivera
    Keymaster
    NI
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    Well, maybe you have something misconfigured.

    Check the DID received and what you have configured on the Inbound Routes

    Channel 'SIP/SIPIDSRV-00000045' sent to invalid extension: context,exten,priority=incoming-calls,SIPIDSRV,1
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    • February 21, 2019 at 8:08 pm
    Ofloo
    Participant
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    I think I know what was wrong, the did was replaced by the trunk I’d replacing it with the did with to, fixed it I think at least getting ring tone.

     

    Still need to test further.

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