ASTERISK@HOME TRUNKS AND OUTGOING RULES PROBLEM !! (((VOLUME 2)))

Discussion in 'UK VOIP' started by phpguy, Nov 14, 2007.

  1. phpguy

    phpguy Guest

    Hello all i have set up my astyerisk@home system using also a zaptel
    telephony card for my landline, when i call in i can hear my music on
    hold fine and extensions ring but i cannot dialout either using ZAP
    (my landline) or SIP

    the settings are

    i created 2 trunks with sip and ZAP and 2 outbound routings with dial
    patterns 99|. for zap (landline) and 88|. to select the sip provider.
    Both return busy when i dial out like 99+a number or 88+a number

    why does this happen ?

    cheers
    phpguy, Nov 14, 2007
    #1
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  2. phpguy

    phpguy Guest

    sorry cancel the below only the SIP TRUNK doesnt work and gives a busy
    tone

    all settigns are correct though such as ip login and paossword

    i dont understand :/

    On 14 Nov, 16:54, phpguy <> wrote:
    > Hello all i have set up my astyerisk@home system using also a zaptel
    > telephony card for my landline, when i call in i can hear my music on
    > hold fine and extensions ring but i cannot dialout either using ZAP
    > (my landline) or SIP
    >
    > the settings are
    >
    > i created 2 trunks with sip and ZAP and 2 outbound routings with dial
    > patterns 99|. for zap (landline) and 88|. to select the sip provider.
    > Both return busy when i dial out like 99+a number or 88+a number
    >
    > why does this happen ?
    >
    > cheers
    phpguy, Nov 14, 2007
    #2
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  3. phpguy

    phpguy Guest

    another add-on, when i try dialing with the same provider (voipbuster)
    from my sip phone directly it works but it wont work via asterisk it
    gives busy signal and logs show channel unavailable for some reason ?

    cheers
    phpguy, Nov 14, 2007
    #3
  4. phpguy

    Desk Rabbit Guest

    Re: ASTERISK@HOME TRUNKS AND OUTGOING RULES PROBLEM !! (((VOLUME2)))

    phpguy wrote:
    > another add-on, when i try dialing with the same provider (voipbuster)
    > from my sip phone directly it works but it wont work via asterisk it
    > gives busy signal and logs show channel unavailable for some reason ?
    >
    > cheers
    >

    Wild guess, is your Asterisk box behind a NAT router?
    Desk Rabbit, Nov 15, 2007
    #4
  5. phpguy

    phpguy Guest

    yes but so are all ip phones that do work when i enter the
    voipbuster.com settings to them manually inside their own menu

    so what do i do now

    cheers

    Desk Rabbit wrote:

    > phpguy wrote:
    > > another add-on, when i try dialing with the same provider (voipbuster)
    > > from my sip phone directly it works but it wont work via asterisk it
    > > gives busy signal and logs show channel unavailable for some reason ?
    > >
    > > cheers
    > >

    > Wild guess, is your Asterisk box behind a NAT router?
    phpguy, Nov 15, 2007
    #5
  6. phpguy

    Jono Guest

    phpguy explained on 15/11/2007 :

    > Desk Rabbit wrote:
    >
    >> phpguy wrote:
    >>> another add-on, when i try dialing with the same provider (voipbuster)
    >>> from my sip phone directly it works but it wont work via asterisk it
    >>> gives busy signal and logs show channel unavailable for some reason ?
    >>>
    >>> cheers
    >>>

    >> Wild guess, is your Asterisk box behind a NAT router?


    > yes but so are all ip phones that do work when i enter the
    > voipbuster.com settings to them manually inside their own menu
    >
    > so what do i do now
    >
    > cheers


    Post the contents of your sip.conf here.......at the bottom of your
    reply, though!
    Jono, Nov 15, 2007
    #6
  7. phpguy

    phpguy Guest

    ok here it is

    ; Note: If your SIP devices are behind a NAT and your Asterisk
    ; server isn't, try adding "nat=1" to each peer definition to
    ; solve translation problems.

    [general]

    port = 5060 ; Port to bind to (SIP is 5060)
    bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
    disallow=all
    allow=ulaw
    allow=alaw
    context = from-sip-external ; Send unknown SIP callers to this context
    callerid = Unknown

    #include sip_nat.conf
    #include sip_custom.conf
    #include sip_additional.conf
    phpguy, Nov 15, 2007
    #7
  8. phpguy

    Jono Guest

    phpguy formulated on Thursday :
    > ok here it is
    >
    > ; Note: If your SIP devices are behind a NAT and your Asterisk
    > ; server isn't, try adding "nat=1" to each peer definition to
    > ; solve translation problems.
    >
    > [general]
    >
    > port = 5060 ; Port to bind to (SIP is 5060)
    > bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
    > disallow=all
    > allow=ulaw
    > allow=alaw
    > context = from-sip-external ; Send unknown SIP callers to this context
    > callerid = Unknown
    >
    > #include sip_nat.conf
    > #include sip_custom.conf
    > #include sip_additional.conf


    [general]

    port = 5060 ; Port to bind to (SIP is 5060)
    bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
    disallow=all
    allow=ulaw
    allow=alaw
    allow=gsm
    ;context = from-sip-external ; Send unknown SIP callers to this context
    context = from-trunk
    ;defaultexpirey = 600 ; include this only if necessary
    ;maxexpirey = 3600 ; include this only if necessary
    progressinband = yes
    ;dtmfmode=auto

    callerid = Unknown
    externip = myPUBLICipADDRESS/DynDNShostname
    localnet=192.168.1.0/255.255.255.0
    nat=yes

    #include sip_nat.conf
    #include sip_custom.conf
    #include sip_additional.conf
    Jono, Nov 15, 2007
    #8
  9. phpguy

    phpguy Guest

    no luck mate

    a female voice says ALL CIRCUITS ARE BUSY NOW TRY AGAIN LATER

    SO STRANGE

    Jono wrote:

    > phpguy formulated on Thursday :
    > > ok here it is
    > >
    > > ; Note: If your SIP devices are behind a NAT and your Asterisk
    > > ; server isn't, try adding "nat=1" to each peer definition to
    > > ; solve translation problems.
    > >
    > > [general]
    > >
    > > port = 5060 ; Port to bind to (SIP is 5060)
    > > bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
    > > disallow=all
    > > allow=ulaw
    > > allow=alaw
    > > context = from-sip-external ; Send unknown SIP callers to this context
    > > callerid = Unknown
    > >
    > > #include sip_nat.conf
    > > #include sip_custom.conf
    > > #include sip_additional.conf

    >
    > [general]
    >
    > port = 5060 ; Port to bind to (SIP is 5060)
    > bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
    > disallow=all
    > allow=ulaw
    > allow=alaw
    > allow=gsm
    > ;context = from-sip-external ; Send unknown SIP callers to this context
    > context = from-trunk
    > ;defaultexpirey = 600 ; include this only if necessary
    > ;maxexpirey = 3600 ; include this only if necessary
    > progressinband = yes
    > ;dtmfmode=auto
    >
    > callerid = Unknown
    > externip = myPUBLICipADDRESS/DynDNShostname
    > localnet=192.168.1.0/255.255.255.0
    > nat=yes
    >
    > #include sip_nat.conf
    > #include sip_custom.conf
    > #include sip_additional.conf
    phpguy, Nov 15, 2007
    #9
  10. phpguy

    phpguy Guest

    =========================================================================
    =========================================================================
    asterisk1*CLI> Verbosity is at least 3
    Connected to Asterisk 1.0.9 currently running on asterisk1 (pid =
    1313)
    asterisk1*CLI> -- Executing GotoIf("SIP/200-2e40", "1?3:2)") in
    new stack
    -- Goto (macro-dialout-trunk,s,3)
    Verbosity is at least 3
    asterisk1*CLI> -- Executing GotoIf("SIP/200-2e40", "0 > 0?2:4") in
    new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing Macro("SIP/200-2e40", "dialout-trunk|2|
    0030210XXXXXXXX|") in new stack
    asterisk1*CLI> -- Goto (macro-record-enable,s,5)
    -- Executing GotoIf("SIP/200-2e40", "13:2)") in new stack
    asterisk1*CLI> -- DBget: varname=RecEnable, family=RECORD-OUT,
    key=200
    -- Goto (macro-dialout-trunk,s,3)
    asterisk1*CLI> -- Executing SetVar("SIP/200-2e40",
    "CALLFILENAME=OUT200-19990107-023107-915694267.2") in new stack
    -- Executing Goto("SIP/200-2e40", "s|14") in new stack
    -- Executing Macro("SIP/200-2e40", "record-enable|200|OUT") in new
    stack
    asterisk1*CLI> -- Executing GotoIf("SIP/200-2e40", "0?15:99") in
    new stack
    -- Executing GotoIf("SIP/200-2e40", "0 > 02:4") in new stack
    asterisk1*CLI> -- Executing NoOp("SIP/200-2e40", "NO RECORDING
    NEEDED") in new stack
    -- Executing GotoIf("SIP/200-2e40", "1?7") in new stack
    -- Goto (macro-record-enable,s,4)
    asterisk1*CLI> -- Executing GotoIf("SIP/200-2e40", "0?9") in new
    stack
    -- Executing GotoIf("SIP/200-2e40", "15:8") in new stack
    asterisk1*CLI> -- Executing SetGroup("SIP/200-2e40", "OUT_2") in
    new stack
    -- Goto (macro-record-enable,s,5)
    asterisk1*CLI> -- Executing SetVar("SIP/200-2e40",
    "DIAL_NUMBER=XXXXXXXX") in new stack
    -- Executing DBget("SIP/200-2e40", "RecEnable=RECORD-OUT/200") in
    new stack
    asterisk1*CLI> -- DBget: varname=RecEnable, family=RECORD-OUT,
    key=200
    asterisk1*CLI> -- Launched AGI Script /var/lib/asterisk/agi-bin/
    fixlocalprefix
    fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf
    -- DBget: Value not found in database.
    asterisk1*CLI> -- Executing SetVar("SIP/200-2e40",
    "OUTNUM=XXXXXXXXXXX") in new stack
    -- Executing Cut("SIP/200-2e40", "custom=OUT_2|:|1") in new stack
    -- Executing SetVar("SIP/200-2e40",
    "CALLFILENAME=OUT200-19990107-023107-915694267.2") in new stack
    asterisk1*CLI> -- Executing Goto("SIP/200-2e40", "s|14") in new
    stack
    asterisk1*CLI> -- Goto (macro-record-enable,s,14)
    asterisk1*CLI> -- Executing GotoIf("SIP/200-2e40", "015:99") in
    new stack
    asterisk1*CLI> -- Goto (macro-record-enable,s,99)
    asterisk1*CLI> -- Executing NoOp("SIP/200-2e40", "NO RECORDING
    NEEDED") in new stack
    asterisk1*CLI> -- Executing GotoIf("SIP/200-2e40", "17") in new
    stack
    asterisk1*CLI> -- Goto (macro-dialout-trunk,s,7)
    asterisk1*CLI> -- Executing GotoIf("SIP/200-2e40", "09") in new
    stack
    asterisk1*CLI> -- Executing SetCallerID("SIP/200-2e40",
    "asiawatcher") in new stack
    asterisk1*CLI> -- Executing SetGroup("SIP/200-2e40", "OUT_2") in
    new stack
    asterisk1*CLI> -- Executing CheckGroup("SIP/200-2e40", "") in new
    stack
    asterisk1*CLI> -- Executing SetVar("SIP/200-2e40",
    "DIAL_NUMBER=XXXXXXXXXXXXX") in new stack
    asterisk1*CLI> -- Executing SetVar("SIP/200-2e40", "DIAL_TRUNK=2")
    in new stack
    asterisk1*CLI> -- Executing AGI("SIP/200-2e40", "fixlocalprefix")
    in new stack
    asterisk1*CLI> -- Launched AGI Script /var/lib/asterisk/agi-bin/
    fixlocalprefix
    asterisk1*CLI> fixlocalprefix: Could not parse /etc/asterisk/
    localprefixes.conf
    asterisk1*CLI> -- AGI Script fixlocalprefix completed, returning 0
    asterisk1*CLI> -- Executing SetVar("SIP/200-2e40",
    "OUTNUM=XXXXXXXXXXX") in new stack
    asterisk1*CLI> -- Executing Cut("SIP/200-2e40", "custom=OUT_2|:|
    1") in new stack
    asterisk1*CLI> -- Executing GotoIf("SIP/200-2e40", "019") in new
    stack
    asterisk1*CLI> -- Executing Dial("SIP/200-2e40", "SIP/
    voipbusterfinal/XXXXXXXXXXXXXX") in new stack
    asterisk1*CLI> == Everyone is busy/congested at this time
    asterisk1*CLI> -- Executing Goto("SIP/200-2e40", "s-CHANUNAVAIL|
    1") in new stack
    asterisk1*CLI> -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    asterisk1*CLI> -- Executing NoOp("SIP/200-2e40", "Dial failed due
    to CHANUNAVAIL") in new stack
    asterisk1*CLI> -- Executing Macro("SIP/200-2e40", "outisbusy") in
    new stack
    asterisk1*CLI> -- Executing Playback("SIP/200-2e40", "allison7/all-
    circuits-busy-now") in new stack
    asterisk1*CLI> -- Playing 'allison7/all-circuits-busy-
    now' (language 'en')
    asterisk1*CLI> -- Executing Playback("SIP/200-2e40", "allison7/pls-
    try-call-later") in new stack
    asterisk1*CLI> -- Playing 'allison7/pls-try-call-later' (language
    'en')
    asterisk1*CLI> -- Executing Macro("SIP/200-2e40", "hangupcall") in
    new stack
    asterisk1*CLI> -- Executing ResetCDR("SIP/200-2e40", "w") in new
    stack
    asterisk1*CLI> -- Executing NoCDR("SIP/200-2e40", "") in new stack
    asterisk1*CLI> -- Executing Wait("SIP/200-2e40", "5") in new stack
    asterisk1*CLI> -- Executing Hangup("SIP/200-2e40", "") in new
    stack
    asterisk1*CLI> == Spawn extension (macro-hangupcall, s, 4) exited
    non-zero on 'SIP/200-2e40' in macro 'hangupcall'
    asterisk1*CLI> == Spawn extension (macro-outisbusy, s, 3) exited non-
    zero on 'SIP/200-2e40' in macro 'outisbusy'
    asterisk1*CLI> == Spawn extension (from-internal, 8800302107292105,
    2) exited non-zero on 'SIP/200-2e40'
    asterisk1*CLI> -- Executing Macro("SIP/200-2e40", "hangupcall") in
    new stack
    asterisk1*CLI> -- Executing ResetCDR("SIP/200-2e40", "w") in new
    stack
    asterisk1*CLI> -- Executing NoCDR("SIP/200-2e40", "") in new stack
    asterisk1*CLI> -- Executing Wait("SIP/200-2e40", "5") in new stack
    asterisk1*CLI> == Spawn extension (macro-hangupcall, s, 3) exited
    non-zero on 'SIP/200-2e40' in macro 'hangupcall'
    asterisk1*CLI> == Spawn extension (from-internal, h, 1) exited non-
    zero on 'SIP/200-2e40'
    phpguy, Nov 15, 2007
    #10
  11. phpguy

    Jono Guest

    phpguy expressed precisely :
    > no luck mate
    >
    > a female voice says ALL CIRCUITS ARE BUSY NOW TRY AGAIN LATER
    >
    > SO STRANGE


    Post your trunk settings
    Jono, Nov 15, 2007
    #11
  12. phpguy

    Jono Guest

    phpguy pretended :
    > no luck mate


    Did you restart asterisk?
    Jono, Nov 15, 2007
    #12
  13. phpguy

    phpguy Guest

    yes i restarted asterisk nothing

    trunk details are the default SIP ones only i replace my username/
    password and sip domain

    cheers
    phpguy, Nov 15, 2007
    #13
  14. phpguy

    Jono Guest

    phpguy presented the following explanation :
    > yes i restarted asterisk nothing
    >
    > trunk details are the default SIP ones only i replace my username/
    > password and sip domain
    >
    > cheers


    Therein will lie your problem.

    Under trunk settings:-

    Dial rules:
    00441234+XXXXXX; substitute 1274 with a 4 digit std (add an X for 7
    digit local numbers)
    0044+XXXXXXXXXX


    Outgoing:

    allow=ulaw&alaw
    authuser=username
    disallow=all
    dtmfmode=rfc2833
    fromdomain=voipbuster.com
    fromuser=00447xxxxxxxxx; my authenticated mobile number
    host=sip.voipbuster.com
    insecure=very
    nat=yes
    qualify=yes
    secret=password
    type=friend
    username=username

    Registration:

    register string: username:p
    Jono, Nov 15, 2007
    #14
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