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Thread: Trixbox + Pennytel How To Get Trunk Working (Correct Settings Inside Thread)

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    Default Trixbox + Pennytel How To Get Trunk Working (Correct Settings Inside Thread)



    Couple of weeks ago i setup a new trixbox system running the latest open source release. I had a fair bit of trouble getting the incomming pennytel trunk to work. It would show up as connected(on trixbox and through pennytel support) but go to a pennytel message bank when i tried to ring the number.

    The issue was in the configuration file, i followed instructions from trixbox without tears 1/2 but theres a small mistake. I thought i would post up the correct details for those looking at getting into trixbox.

    *********************Settings********************* **********


    Peer Details
    allow=alaw&ulaw&gsm
    canredirect=no
    canreinvite=no
    disallow=all
    host=dynamic
    insecure=very
    secret=(Password)
    type=peer
    username=(Username)


    User Details
    canreinvite=no
    context=from-sip
    fromuser=(Username)
    qualify=no
    secret=(Password)
    type=user
    username=(Username)

    Register String
    (Username)_:_(Password)@sip.pennytel.com/(Username) (DELETE UNDERSCORES)
    *********************************************
    Replace(Username)/(Password) with relevent details and you'll be off and running



    Hopefully that'll help you guys out, it took alot of messing around for me to sus it out.



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    Quote Originally Posted by warez_kid View Post
    Couple of weeks ago i setup a new trixbox system running the latest open source release. I had a fair bit of trouble getting the incomming pennytel trunk to work. It would show up as connected(on trixbox and through pennytel support) but go to a pennytel message bank when i tried to ring the number.
    Thanks for posting that.

    If I use exactly the same settings (trixbox 2.6.2.1) I can't make an outgoing call via PennyTel ("all circuits are busy now"). trixbox is registered with PT (PT|diagnostics) and in PBX|PBX Status. If I change the PBX|Settings|Trunks|SIP/PennyTel PEER Details section so that the line:
    host=dynamic
    changes to:
    host=sip.pennytel.com
    then I can make outgoing calls.

    Register string is:
    6173121XXXX:ZZZZZ@sip.pennytel.com/6173121XXXX

    PT diagnostics say:
    Status : Online
    Agent : Asterisk PBX
    Contact : sip:6173121XXXX@220.233.XXX.YY:5076

    However, if I dial my PT number from a PSTN service, I get "the number you have dialled is not in service".

    The asterisk CLI console says:
    Code:
    -- Executing [6173121XXXX@from-sip-external:1] NoOp("SIP/6173121XXXX-08e540d0", "Received incoming SIP connection from unknown peer to 6173121XXXX") in new stack
        -- Executing [6173121XXXX@from-sip-external:2] Set("SIP/6173121XXXX-08e540d0", "DID=6173121XXXX") in new stack
        -- Executing [6173121XXXX@from-sip-external:3] Goto("SIP/6173121XXXX-08e540d0", "s|1") in new stack
        -- Goto (from-sip-external,s,1)
        -- Executing [s@from-sip-external:1] GotoIf("SIP/6173121XXXX-08e540d0", "0?from-trunk|6173121XXXX|1") in new stack
        -- Executing [s@from-sip-external:2] Set("SIP/6173121XXXX-08e540d0", "TIMEOUT(absolute)=15") in new stack
        -- Channel will hangup at 2009-01-12 07:28:32 UTC.
        -- Executing [s@from-sip-external:3] Answer("SIP/6173121XXXX-08e540d0", "") in new stack
        -- Executing [s@from-sip-external:4] Wait("SIP/6173121XXXX-08e540d0", "2") in new stack
        -- Executing [s@from-sip-external:5] Playback("SIP/6173121XXXX-08e540d0", "ss-noservice") in new stack
        -- <SIP/6173121XXXX-08e540d0> Playing 'ss-noservice' (language 'en')
      == Spawn extension (from-sip-external, s, 5) exited non-zero on 'SIP/6173121XXXX-08e540d0'
        -- Executing [h@from-sip-external:1] NoOp("SIP/6173121XXXX-08e540d0", "Hangup") in new stack
        -- Executing [h@from-sip-external:2] Set("SIP/6173121XXXX-08e540d0", "DID=s") in new stack
        -- Executing [h@from-sip-external:3] Goto("SIP/6173121XXXX-08e540d0", "s|1") in new stack
        -- Goto (from-sip-external,s,1)
        -- Executing [s@from-sip-external:1] GotoIf("SIP/6173121XXXX-08e540d0", "0?from-trunk|s|1") in new stack
        -- Executing [s@from-sip-external:2] Set("SIP/6173121XXXX-08e540d0", "TIMEOUT(absolute)=15") in new stack
        -- Channel will hangup at 2009-01-12 07:28:38 UTC.
        -- Executing [s@from-sip-external:3] Answer("SIP/6173121XXXX-08e540d0", "") in new stack
      == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/6173121XXXX-08e540d0'
    So we know the call is reaching the trixbox.

    For Trunk Incoming USER Context I have 6173121XXXX.

    If I change PBX|Settings|Trunks|SIP/PennyTel USER Details section from
    context=from-sip
    to:
    context=from-trunk

    then it doesn't seem to make any difference. Sometimes I get busy tone and nothing appears in the CLI.

    I have one inbound route I've called "From PT" with the destingation set to extension 200 (the registered softphone). The extension details say it is "Used as Destination by 1 Object:" and when hovered over it says "Inbound Route: From PT (/)".

    The extension device options are:

    Code:
    This device uses sip technology.
    secret xxxx
    dtmfmode rfc2833
    canreinvite no
    context from-internal
    host dynamic
    type friend
    nat yes
    port 5060
    qualify yes
    callgroup 	
    pickupgroup 	
    disallow 	
    allow 	
    dial SIP/200
    accountcode 
    mailbox 200@default
    Language
    I'm using a Billion 7404VGP router, with its VoIP ports registered to VoXaLot (my current method of using VoIP). VoXaLot is not registered to PT.

    For my testing of trixbox, I have SJPhone. It's registered to the trixbox, can make external calls, receive calls from another extension, dial the echo test and speaking clock, etc.
    No ports in the Billion have been forwarded for VoIP purposes.

    If I open DMZ to the trixbox, incoming still does not work.

    What have I missed?

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    Ok - answering myself here... the trick is, in PBX Settings|General Settings|
    Allow Anonymous Inbound SIP Calls?|yes. There's probably a way to make things more secure, but that gets me going.

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    Ok - answering myself here... the trick is, in PBX Settings|General Settings|
    Allow Anonymous Inbound SIP Calls?|yes. There's probably a way to make things more secure, but that gets me going.
    I was already know that if we take a mini tour on any new software setting section than we don't need to question to any other ?

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    Yes if we take a mini tour on any new software setting section than we don't need to question to any other.




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    I have a trixbox digium card up for grabs at the moment in BS&T

    Reality is an invention of my imagination.
    ಠ_ಠ

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    fernbay that card is ROCK SOLID + well supported in trixbox shouldnt have any problems selling it
    Last edited by urban_s0ulja; 31-07-09 at 04:17 PM.

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