ASTERISK@HOME TRUNKS AND OUTGOING RULES PROBLEM !!

Discussion in 'UK VOIP' started by phpguy, Nov 10, 2007.

  1. phpguy

    phpguy Guest

    hello all i got a simple question im using asterisk@home and i got x2
    trunks one zap trunk that is using my normal landline for inbound and
    outgoing calls and i also got a sip trunk with voipbuster

    im trying to set when i dial 99+a number to use the zap trunk and when
    i dial 88+a number to dial the specified number over the SIP trunk

    in few words

    99 077123456 should dial a uk mobile over the zap trunk connected to
    my landline using a fxo card and when i dial 88 003031072XXXX should
    dial greece over the SIP trunk

    im trying to do that with the rules both in trunk and outgoing routing
    of:

    99|. for zap and 88|. for sip

    problem is it works for zap but for sip it wont work

    what am i doing wrong please ?

    cheers
     
    phpguy, Nov 10, 2007
    #1
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  2. phpguy

    alexd Guest

    phpguy wrote:

    > im trying to do that with the rules both in trunk and outgoing routing
    > of:
    >
    > 99|. for zap and 88|. for sip
    >
    > problem is it works for zap but for sip it wont work
    >
    > what am i doing wrong please ?


    What happens when it doesn't work? My first suggestion would be to look at
    the Asterisk console ['asterisk -rc'] and see what it says. If it doesn't
    say much, then run 'set verbose 5', and try again. Also, does your SIP
    trunk work at all, ie without fancy routing? Can you receive calls on it
    [if applicable]?

    --
    <http://ale.cx/> (AIM:troffasky) ()
    18:34:38 up 8 days, 11:19, 3 users, load average: 0.16, 0.17, 0.17
    50,000 watts of funking power
     
    alexd, Nov 10, 2007
    #2
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  3. phpguy

    phpguy Guest

    =========================================================================
    Connected to Asterisk 1.0.9 currently running on asterisk1 (pid =
    1313)
    Verbosity is at least 3
    asterisk1*CLI> set verbose 5
    Verbosity was 3 and is now 5
    -- Executing Macro("SIP/201-f9d3", "dialout-trunk|3|
    00302107292105|") in new stack
    -- Executing GotoIf("SIP/201-f9d3", "1?3:2)") in new stack
    -- Goto (macro-dialout-trunk,s,3)
    -- Executing Macro("SIP/201-f9d3", "record-enable|201|OUT") in new
    stack
    -- Executing GotoIf("SIP/201-f9d3", "0 > 0?2:4") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing GotoIf("SIP/201-f9d3", "1?5:8") in new stack
    -- Goto (macro-record-enable,s,5)
    -- Executing DBget("SIP/201-f9d3", "RecEnable=RECORD-OUT/201") in
    new stack
    -- DBget: varname=RecEnable, family=RECORD-OUT, key=201
    -- DBget: set variable RecEnable to DISABLED
    -- Executing SetVar("SIP/201-f9d3",
    "CALLFILENAME=OUT201-19990101-232229-915
    250949.2") in new stack
    -- Executing Goto("SIP/201-f9d3", "s|14") in new stack
    -- Goto (macro-record-enable,s,14)
    -- Executing GotoIf("SIP/201-f9d3", "0?15:99") in new stack
    -- Goto (macro-record-enable,s,99)
    -- Executing NoOp("SIP/201-f9d3", "NO RECORDING NEEDED") in new
    stack
    -- Executing GotoIf("SIP/201-f9d3", "1?7") in new stack
    -- Goto (macro-dialout-trunk,s,7)
    -- Executing GotoIf("SIP/201-f9d3", "1?9") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing SetGroup("SIP/201-f9d3", "OUT_3") in new stack
    -- Executing CheckGroup("SIP/201-f9d3", "") in new stack
    -- Executing SetVar("SIP/201-f9d3", "DIAL_NUMBER=00302107292105")
    in new sta ck
    -- Executing SetVar("SIP/201-f9d3", "DIAL_TRUNK=3") in new stack
    -- Executing AGI("SIP/201-f9d3", "fixlocalprefix") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
    asterisk1*CLI>
    ! abort add agi cdr database
    debug dont dump exit extensions help
    iax2 include init load local logger
    meetme mgcp no pri quit reload
    remove restart set show sip skinny
    soft stop unload zap
    -- AGI Script fixlocalprefix completed, returning 0
    -- Executing SetVar("SIP/201-f9d3", "OUTNUM=00302107292105") in
    new stack
    -- Executing Cut("SIP/201-f9d3", "custom=OUT_3|:|1") in new stack
    -- Executing GotoIf("SIP/201-f9d3", "0?19") in new stack
    -- Executing Dial("SIP/201-f9d3", "SIP/
    voipbuster2/00302107292105") in new s tack
    == Everyone is busy/congested at this time
    -- Executing Goto("SIP/201-f9d3", "s-CHANUNAVAIL|1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing NoOp("SIP/201-f9d3", "Dial failed due to
    CHANUNAVAIL") in new s tack
    -- Executing Macro("SIP/201-f9d3", "outisbusy") in new stack
    -- Executing Playback("SIP/201-f9d3", "allison7/all-circuits-busy-
    now") in n ew stack
    -- Playing 'allison7/all-circuits-busy-now' (language 'en')
    -- Executing Playback("SIP/201-f9d3", "allison7/pls-try-call-
    later") in new stack
    -- Playing 'allison7/pls-try-call-later' (language 'en')
    -- Executing Macro("SIP/201-f9d3", "hangupcall") in new stack
    -- Executing ResetCDR("SIP/201-f9d3", "w") in new stack
    -- Executing NoCDR("SIP/201-f9d3", "") in new stack
    -- Executing Wait("SIP/201-f9d3", "5") in new stack
    == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/
    201-f9d3' in macro 'hangupcall'
    == Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/
    201-f9d3' i n macro 'outisbusy'
    == Spawn extension (from-internal, 800302107292105, 2) exited non-
    zero on 'SIP /201-f9d3'
    -- Executing Macro("SIP/201-f9d3", "hangupcall") in new stack
    -- Executing ResetCDR("SIP/201-f9d3", "w") in new stack
    -- Executing NoCDR("SIP/201-f9d3", "") in new stack
    -- Executing Wait("SIP/201-f9d3", "5") in new stack
    == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/
    201-f9d3' in macro 'hangupcall'
    == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/201-
    f9d3'
    asterisk1*CLI>
    asterisk1*CLI> Connected to Asterisk 1.0.9 currently running on
    asterisk1 (pid = 1313)
    Verbosity is at least 3
    asterisk1*CLI> set verbose 5
    Verbosity was 3 and is now 5
    Connected to Asterisk 1.0.9 currently running on asterisk1 (pid =
    1313)
    asterisk1*CLI> -- Executing GotoIf("SIP/201-f9d3", "1?3:2)") in
    new stack
    Verbosity is at least 3
    asterisk1*CLI> -- Executing Macro("SIP/201-f9d3", "record-enable|
    201|OUT") in new stack
    asterisk1*CLI> set verbose 5
    asterisk1*CLI> -- Goto (macro-record-enable,s,4)
    -- Executing GotoIf("SIP/201-f9d3", "1?5:8") in new stack
    -- Goto (macro-record-enable,s,5)
    -- Executing DBget("SIP/201-f9d3", "RecEnable=RECORD-OUT/201") in
    new stack
    -- DBget: varname=RecEnable, family=RECORD-OUT, key=201
    -- DBget: set variable RecEnable to DISABLED
    -- Executing SetVar("SIP/201-f9d3",
    "CALLFILENAME=OUT201-19990101-232229-915
    250949.2") in new stack
    -- Executing Goto("SIP/201-f9d3", "s|14") in new stack
    Verbosity was 3 and is now 5
    asterisk1*CLI> -- Executing Macro("SIP/201-f9d3", "dialout-trunk|3|
    00302107292105|") in new stack
    asterisk1*CLI> -- Goto (macro-record-enable,s,99)
    -- Executing NoOp("SIP/201-f9d3", "NO RECORDING NEEDED") in new
    stack
    -- Executing GotoIf("SIP/201-f9d3", "1?7") in new stack
    -- Executing GotoIf("SIP/201-f9d3", "13:2)") in new stack
    asterisk1*CLI> -- Executing GotoIf("SIP/201-f9d3", "1?9") in new
    stack
    -- Goto (macro-dialout-trunk,s,3)
    asterisk1*CLI> -- Executing SetGroup("SIP/201-f9d3", "OUT_3") in
    new stack
    -- Executing CheckGroup("SIP/201-f9d3", "") in new stack
    -- Executing Macro("SIP/201-f9d3", "record-enable|201|OUT") in new
    stack
    asterisk1*CLI> -- Executing GotoIf("SIP/201-f9d3", "0 > 02:4") in
    new stack
    asterisk1*CLI> -- Executing SetVar("SIP/201-f9d3", "DIAL_TRUNK=3")
    in new stack
    -- Goto (macro-record-enable,s,4)
    asterisk1*CLI> -- Launched AGI Script /var/lib/asterisk/agi-bin/
    fixlocalprefix
    asterisk1*CLI>
    ! abort add agi cdr database
    debug dont dump exit extensions help
    -- Executing GotoIf("SIP/201-f9d3", "15:8") in new stack
    asterisk1*CLI> meetme mgcp no pri
    quit reload
    remove restart set show sip skinny
    -- Goto (macro-record-enable,s,5)
    asterisk1*CLI> -- Executing DBget("SIP/201-f9d3",
    "RecEnable=RECORD-OUT/201") in new stack
    asterisk1*CLI> -- DBget: varname=RecEnable, family=RECORD-OUT,
    key=201
    asterisk1*CLI> -- DBget: set variable RecEnable to DISABLED
    asterisk1*CLI> -- Executing GotoIf("SIP/201-f9d3", "0?19") in new
    stack
    -- Executing SetVar("SIP/201-f9d3",
    "CALLFILENAME=OUT201-19990101-232229-915
    250949.2") in new stack
    asterisk1*CLI> -- Executing Dial("SIP/201-f9d3", "SIP/
    voipbuster2/00302107292105") in new s tack
    == Everyone is busy/congested at this time
    -- Executing Goto("SIP/201-f9d3", "s|14") in new stack
    asterisk1*CLI> -- Executing Goto("SIP/201-f9d3", "s-CHANUNAVAIL|
    1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Goto (macro-record-enable,s,14)
    asterisk1*CLI> -- Executing GotoIf("SIP/201-f9d3", "015:99") in
    new stack
    asterisk1*CLI> -- Playing 'allison7/all-circuits-busy-
    now' (language 'en')
    -- Goto (macro-record-enable,s,99)
    asterisk1*CLI> -- Executing Playback("SIP/201-f9d3", "allison7/pls-
    try-call-later") in new stack
    -- Executing NoOp("SIP/201-f9d3", "NO RECORDING NEEDED") in new
    stack
    asterisk1*CLI> -- Executing GotoIf("SIP/201-f9d3", "17") in new
    stack
    asterisk1*CLI> -- Goto (macro-dialout-trunk,s,7)
    asterisk1*CLI> -- Executing GotoIf("SIP/201-f9d3", "19") in new
    stack
    asterisk1*CLI> -- Goto (macro-dialout-trunk,s,9)
    asterisk1*CLI> -- Executing SetGroup("SIP/201-f9d3", "OUT_3") in
    new stack
    asterisk1*CLI> -- Executing CheckGroup("SIP/201-f9d3", "") in new
    stack
    asterisk1*CLI> -- Executing SetVar("SIP/201-f9d3",
    "DIAL_NUMBER=00302107292105") in new sta ck
    asterisk1*CLI> -- Executing SetVar("SIP/201-f9d3", "DIAL_TRUNK=3")
    in new stack
    asterisk1*CLI> -- Executing AGI("SIP/201-f9d3", "fixlocalprefix")
    in new stack
    asterisk1*CLI> -- Launched AGI Script /var/lib/asterisk/agi-bin/
    fixlocalprefix
    asterisk1*CLI> asterisk1*CLI>
    asterisk1*CLI> ! abort add agi
    cdr database
    /bin/sh: line 1: abort: command not found
    asterisk1*CLI> debug dont dump exit
    extensions help
    asterisk1*CLI> iax2 include init load
    local logger
    asterisk1*CLI> meetme mgcp no pri
    quit reload
    asterisk1*CLI> remove restart set show
    sip skinny
    asterisk1*CLI> soft stop unload zap
    asterisk1*CLI> -- AGI Script fixlocalprefix completed, returning 0
    asterisk1*CLI> -- Executing SetVar("SIP/201-f9d3",
    "OUTNUM=00302107292105") in new stack
    asterisk1*CLI> -- Executing Cut("SIP/201-f9d3", "custom=OUT_3|:|
    1") in new stack
    asterisk1*CLI> -- Executing GotoIf("SIP/201-f9d3", "019") in new
    stack
    asterisk1*CLI> -- Executing Dial("SIP/201-f9d3", "SIP/
    voipbuster2/00302107292105") in new s tack
    asterisk1*CLI> == Everyone is busy/congested at this time
    asterisk1*CLI> -- Executing Goto("SIP/201-f9d3", "s-CHANUNAVAIL|
    1") in new stack
    asterisk1*CLI> -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    asterisk1*CLI> -- Executing NoOp("SIP/201-f9d3", "Dial failed due
    to CHANUNAVAIL") in new s tack
    asterisk1*CLI> -- Executing Macro("SIP/201-f9d3", "outisbusy") in
    new stack
    asterisk1*CLI> -- Executing Playback("SIP/201-f9d3", "allison7/all-
    circuits-busy-now") in n ew stack
    asterisk1*CLI> -- Playing 'allison7/all-circuits-busy-
    now' (language 'en')
    asterisk1*CLI> -- Executing Playback("SIP/201-f9d3", "allison7/pls-
    try-call-later") in new stack
    asterisk1*CLI> -- Playing 'allison7/pls-try-call-later' (language
    'en')
    asterisk1*CLI> -- Executing Macro("SIP/201-f9d3", "hangupcall") in
    new stack
    asterisk1*CLI> -- Executing ResetCDR("SIP/201-f9d3", "w") in new
    stack
    asterisk1*CLI> -- Executing NoCDR("SIP/201-f9d3", "") in new stack
    asterisk1*CLI> -- Executing Wait("SIP/201-f9d3", "5") in new stack
    asterisk1*CLI> == Spawn extension (macro-hangupcall, s, 3) exited
    non-zero on 'SIP/201-f9d3' in macro
    'hangupcall'
    asterisk1*CLI> == Spawn extension (macro-outisbusy, s, 3) exited non-
    zero on 'SIP/201-f9d3' i n macro 'outisbusy'
    asterisk1*CLI> == Spawn extension (from-internal, 800302107292105,
    2) exited non-zero on 'SIP /201-f9d3'
    asterisk1*CLI> -- Executing Macro("SIP/201-f9d3", "hangupcall") in
    new stack
    asterisk1*CLI> -- Executing ResetCDR("SIP/201-f9d3", "w") in new
    stack
    asterisk1*CLI> -- Executing NoCDR("SIP/201-f9d3", "") in new stack
    asterisk1*CLI> -- Executing Wait("SIP/201-f9d3", "5") in new stack
    asterisk1*CLI> == Spawn extension (macro-hangupcall, s, 3) exited
    non-zero on 'SIP/201-f9d3' in macro
    'hangupcall'
    asterisk1*CLI> == Spawn extension (from-internal, h, 1) exited non-
    zero on 'SIP/201-f9d3'
     
    phpguy, Nov 10, 2007
    #3
  4. phpguy

    phpguy Guest

    thanks for your repl;y i pasted what you requested

    i get a busy tone or sometimes a timeout
     
    phpguy, Nov 10, 2007
    #4
  5. phpguy

    Paul Hayes Guest

    phpguy wrote:
    [snip]

    voipbuster2/00302107292105") in new s tack
    == Everyone is busy/congested at this time

    Your service provider is returning busy when you send that number to
    them. Nothing wrong with your dialplan by the looks of it but you
    should check your sip.conf settings for your Voipbuster account and that
    your account actually works if you just use the same settings directly
    on an IP phone.

    cheers,
    Paul.

    --
    Working Email:

    paul-at-polog40-dot-co-dot-uk
     
    Paul Hayes, Nov 12, 2007
    #5
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