Inbound routes don't work. Need help.

Discussion in 'General' started by Jayka, Dec 23, 2010.

  1. Jayka

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    For some reason, the DIDs ported to my system yesterday do not get redirected to extensions, although I can see the calls coming in from the trunk. CLI revealed the following:



    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    -- Executing [+16463504484@from-pstn:1] Set("SIP/BW-SIP-A-00003f2b", "__FROM_DID=+16463504484") in new stack
    -- Executing [+16463504484@from-pstn:2] NoOp("SIP/BW-SIP-A-00003f2b", "Received an unknown call with DID set to +16463504484") in new stack
    -- Executing [+16463504484@from-pstn:3] Goto("SIP/BW-SIP-A-00003f2b", "s,a2") in new stack
    -- Goto (from-pstn,s,2)
    -- Executing [s@from-pstn:2] Answer("SIP/BW-SIP-A-00003f2b", "") in new stack
    -- Executing [s@from-pstn:3] Wait("SIP/BW-SIP-A-00003f2b", "2") in new stack
    -- Executing [s@from-pstn:4] Playback("SIP/BW-SIP-A-00003f2b", "ss-noservice") in new stack
    -- <SIP/BW-SIP-A-00003f2b> Playing 'ss-noservice.gsm' (language 'en')


    Need urgent help, my call center is not operational! :(

    Thank you.
     
  2. Jayka

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    Tried adding this to extensions_custom.conf, no luck:

    [ext-did-catchall]
    include => ext-did-catchall-custom
    exten => s,1,Noop(No DID or CID Match)
    exten => s,n(a2),Answer
    exten => s,n,Wait(2)
    exten => s,n,Playback(ss-noservice)
    exten => s,n,SayAlpha(${FROM_DID})
    exten => s,n,Hangup
    exten => _.,1,Set(__FROM_DID=${EXTEN})
    exten => _.,n,Noop(Received an unknown call with DID set to ${EXTEN})
    exten => _.,n,Goto(s,a2)
    exten => h,1,Hangup
     
  3. Lee Sharp

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    And what does your inbound rout say? And do you have a catch-all inbound route above it?
     
  4. Jayka

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    User Context: from-bandwidth-A


    type=peer
    reinvite=yes
    port=5060
    insecure=invite,port
    host=216.82.224.202
    fromdomain=216.82.224.202
    dtmfmode=rfc2833
    disallow=all
    canreinvite=no
    allow=ulaw&alaw&g729
    qualify=300
     
  5. Lee Sharp

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    That is a trunk. I need the inbound route. For example, I have one labeled TelCentris that sends DID 1213npanxxx /Any CID to extension 3300.

    You need one that sends 16463504484 / Any CID to wherever... Note the 1 in front.
     
  6. Jayka

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    I have +1 in the front. It works with all older numbers, but for some reason not with the ones imported yesterday.
     
  7. Jayka

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    I'm really desperate here...
     
  8. dicko

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    Don't be desperate, RTFM instead

    on the inbound route page read the pop-up help for DID and use pattern matching, for example

    _.16463504484

    instead, or add a custom-context that strips the leading + and ujse that, there are a few examples at FreePBX.org
     
  9. Jayka

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    Apparently the problem is that the server is not able to save any changes made to asterisk settings.

    For example, when I change the IVR action from dialing into a queue to dialing an extension (when 1 is pressed, for example) it still dials into the queue (although the settings show the updated configuration).

    What may be the reason for that?

    P.S. Does anybody know what happened to palosanto support?
     
  10. dicko

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    .
    .
    P.S. Does anybody know what happened to palosanto support?
    .
    .

    They just plain refuse to come here, go figure, you could try the paid support link below, or perhaps wait in line in the bug-tracker link below.

    FreePBX has been truly forked-up by PaloSanto, you need to be explicit as to whether you use the IMHO broken Elastix FreePBX embedded interface provided by the aged and dysfunctional Elastix FreePBX rpm, or the unembedded FreePBX interface , if the later be aware that by making it mainline functional again by applying the currently available FreePBX updates (that never needed rpm's they just do it inline, slick as shit off a chrome shovel) will make you forever an outcast in Elastix and "lot's of things will break" in the embedded Elastix Interface. again "go figure", if you do that "progression" (sic) you will probably want to remove the Elastix inserted contexts in

    /etc/asterisk/extensions_override_freepbx.conf

    they will break ARI, if in fact they haven't already removed ARI, AGAIN (please stop doing that PaloSanto, it's way better than your offering ;) ) , and any legitimate use of the so called userfield in the asteriskcdrdb database, here they have suborned it into a distrospecificfield, again IMHO a transgression that will come back to haunt them.


    happy holidays all . . . .


    dicko
     

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