help with dialplan

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Asterisk Users 22 Comments

Hi all,

I’ve been have problems with getting this system on line and would like to acquire some help with the extensions.conf.

My current problem is that the phones won’t dialout.on the VOIP lines listed as dialout1, dialout2, dialout3. This version of asterisk is 1.6.2.11. Below is the extensions.conf file.

[globals]

QPHONE0=SIP/10
QPHONE1=SIP/11
QPHONE2=SIP/12
QPHONE3=SIP/13
QPHONE4=SIP/14
QPHONE5=SIP/15
QPHONE6=SIP/16
QPHONE7=SIP/17

ACAPHONE0=SIP/20
ACAPHONE1=SIP/21
ACAPHONE2=SIP/22
ACAPHONE3=SIP/23
ACAPHONE4=SIP/24
ACAPHONE5=SIP/25
ACAPHONE6=SIP/26
ACAPHONE7=SIP/27

GMNETPHONE0=SIP/30
GMNETPHONE1=SIP/31
GMNETPHONE2=SIP/32
GMNETPHONE3=SIP/33
GMNETPHONE4=SIP/34
GMNETPHONE5=SIP/35
GMNETPHONE6=SIP/36
GMNETPHONE7=SIP/37

EXTERNPHONE0=SIP/150

CPHONE1=SIP/16780000000
CPHONE2=SIP/17700000000

EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/G1
; Change this for production use:
EMERGENCY_NUM=6789542133

[from-pstn]
exten => s,1,Set(FROM_DID=”6780000000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn1]
exten => s,1,Set(FROM_DID=”6780000000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn2]
exten => s,1,Set(FROM_DID=”6780000000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn3]
exten => s,1,Set(FROM_DID=”6780000000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn4]
exten => s,1,Set(FROM_DID=”6780000000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming3,s,1)

[from-pstn5]
exten => s,1,Set(FROM_DID=”6780000000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming2,s,1)

[from-pstn6]
exten => s,1,Set(FROM_DID=”6780000000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn7]
exten => s,1,Set(FROM_DID=”6780000000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn8]
exten => s,1,Set(FROM_DID=”6780000000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[incoming1]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten =>
s,n,Dial(${QPHONE0}&${QPHONE1}&${QPHONE2}&${QPHONE3}&${QPHONE4}&${QPHONE5}&${QPHONE6}&${QPHONE7},40,Ttr)
exten => s,n,Hangup

[incoming2]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten =>
s,n,Dial(${ACAPHONE0}&${ACAPHONE1}&${ACAPHONE2}&${ACAPHONE3}&${ACAPHONE4}&${ACAPHONE5}&${ACAPHONE6}&${ACAPHONE7},40,TTr)
exten => s,n,Hangup

[incoming3]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten => s,n,GotoIfTime(17:01-09:00|*|*|*?closed,s,1)
exten =>
s,n,Dial(${GMNETPHONE0}&${GMNETPHONE1}&${GMNETPHONE2}&${GMNETPHONE3}&${GMNETPHONE4}&${GMNETPHONE5}&${GMNETPHONE6}&${GMNETPHONE7},40,Ttr)
exten => s,n,Hangup

[from-interal]
include => dialout1
include => dialout2
include => dialout3
include => parkedcalls
include => intercom

exten => 10,1,Macro(oneline,${QPHONE0})
exten => 11,1,Macro(oneline,${QPHONE1})
exten => 12,1,Macro(oneline,${QPHONE2})
exten => 13,1,Macro(oneline,${QPHONE3})
exten => 14,1,Macro(oneline,${QPHONE4})
exten => 15,1,Macro(oneline,${QPHONE5})
exten => 16,1,Macro(oneline,${QPHONE6})
exten => 17,1,Macro(oneline,${QPHONE7})

exten => 20,1,Macro(oneline,${ACAPHONE0})
exten => 21,1,Macro(oneline,${ACAPHONE1})
exten => 22,1,Macro(oneline,${ACAPHONE2})
exten => 23,1,Macro(oneline,${ACAPHONE3})
exten => 24,1,Macro(oneline,${ACAPHONE4})
exten => 25,1,Macro(oneline,${ACAPHONE5})
exten => 26,1,Macro(oneline,${ACAPHONE6})
exten => 27,1,Macro(oneline,${ACAPHONE7})

exten => 30,1,Macro(oneline,${GMNETPHONE0})
exten => 31,1,Macro(oneline,${GMNETPHONE1})
exten => 32,1,Macro(oneline,${GMNETPHONE2})
exten => 33,1,Macro(oneline,${GMNETPHONE3})
exten => 34,1,Macro(oneline,${GMNETPHONE4})
exten => 35,1,Macro(oneline,${GMNETPHONE5})
exten => 36,1,Macro(oneline,${GMNETPHONE6})
exten => 37,1,Macro(oneline,${GMNETPHONE7})

exten => 40,1,Macro(oneline,${QPHONE0})
exten => 41,1,Macro(oneline,${QPHONE1})
exten => 42,1,Macro(oneline,${QPHONE2})
exten => 43,1,Macro(oneline,${QPHONE3})
exten => 44,1,Macro(oneline,${QPHONE4})
exten => 45,1,Macro(oneline,${QPHONE5})
exten => 46,1,Macro(oneline,${QPHONE6})
exten => 47,1,Macro(oneline,${QPHONE7})

exten => 150,1,Macro(oneline,${EXTERNPHONE0})

[macro-oneline]
exten => s,1,Set(CHANNEL(musicclass)=default)
exten => s,n,Dial(${ARG1},20,Ttr)
exten => s,n,Voicemail(${MACRO_EXTEN})
exten => s,n,Hangup
exten => s,102,Voicemail(${MACRO_EXTEN})
exten => s,103,Hangup

[dialout1]
include => from-internal
include => 411
exten => 911,1,Goto(nineoneone,s,1)
exten => _1NXXNXXXXXX,n,Dial(SIP/v6781234567/${EXTEN},40,Ttr)
exten => _NXXNXXXXXX,n,Dial(SIP/v6781234567/${EXTEN},40,Ttr)

[dialout2]
include => from-internal
include => 411
exten => 911,1,Goto(nineoneone,s,1)
exten => _1NXXNXXXXXX,n,Dial(SIP/voipdialACA/${EXTEN},40,Ttr)
exten => _NXXNXXXXXX,n,Dial(SIP/voipdialACA/${EXTEN},40,Ttr)

[dialout3]
include => from-internal
include => 411
exten => 911,1,Goto(nineoneone,s,1)
exten => _1NXXNXXXXXX,n,Dial(SIP/v6781234567/${EXTEN},40,Ttr)
exten => _1NXXNXXXXXX,n,Dial(SIP/v6781234567/${EXTEN},40,Ttr)

[voicemail]
exten => 8500,1,VoicemailMain
exten => 8500,2,Hangup

[411]
exten => 411,1,Dial(SIP/v6781234567/18004664411,,Ttr)

[nineoneone]
exten => s,1,Set(SET_EMERG_FLAG=0)
exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten => s,n,Set(EMERGENCY=1,g)
exten => s,n,Set(SET_EMERG_FLAG=1)
exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})
exten => s,s+2(trunkbusy),GotoIf($[${EMERGENCY} = 1]?inprogress)
exten => s,n,SoftHangup(${EMERGENCY_TRUNK}-1)
exten => s,n,Wait(12)
exten => s,n,Goto(checkavail)
exten => s,s+2(inprogress),Congestion
exten => s,checkavail+101(notavail),Goto(trunkbusy)
exten => h,1,GotoIf($[${SET_EMERG_FLAG} = 1]?3)
exten => h,3,Set(EMERGENCY=0,g)

[closed]
exten => s,n,Dial(Dial(SIP/v6781234567/${CPHONE1},40,Ttr)
exten => s,n,Hangup

[intercom]
exten => 59,1,SIPAddHeader(Call-Info: answer-after=0)
exten =>
59,2,Page(${GMNETPHONE0}&${GMNETPHONE1}&${GMNETPHONE2}&${GMNETPHONE3}&${GMNETPHONE4}&${GMNETPHONE5}&${GMNETPHONE6}&${GMNETPHONE7}|d)
exten => 59,3, Hangup

22 thoughts on - help with dialplan

  • Todd

    How do you have the context in the phones sip configs set?

    Bryant

    href=”mailto:treese65@gmail.com”>treese65@gmail.com

    Hi all,

    I’ve been have problems with getting this system on line and would like
    to acquire some help with the extensions.conf.

    My current problem is that the phones won’t dialout.on the VOIP lines
    listed as dialout1, dialout2, dialout3. This version of asterisk is
    1.6.2.11. Below is the extensions.conf file.

    [globals]

    QPHONE0=SIP/10
    QPHONE1=SIP/11
    QPHONE2=SIP/12
    QPHONE3=SIP/13
    QPHONE4=SIP/14
    QPHONE5=SIP/15
    QPHONE6=SIP/16
    QPHONE7=SIP/17

    ACAPHONE0=SIP/20
    ACAPHONE1=SIP/21
    ACAPHONE2=SIP/22
    ACAPHONE3=SIP/23
    ACAPHONE4=SIP/24
    ACAPHONE5=SIP/25
    ACAPHONE6=SIP/26
    ACAPHONE7=SIP/27

    GMNETPHONE0=SIP/30
    GMNETPHONE1=SIP/31
    GMNETPHONE2=SIP/32
    GMNETPHONE3=SIP/33
    GMNETPHONE4=SIP/34
    GMNETPHONE5=SIP/35
    GMNETPHONE6=SIP/36
    GMNETPHONE7=SIP/37

    EXTERNPHONE0=SIP/150

    CPHONE1=SIP/16780000000
    CPHONE2=SIP/17700000000

    EMERGENCY=0
    EMERGENCY_TRUNK=DAHDI/G1
    ; Change this for production use:
    EMERGENCY_NUM=6789542133

    [from-pstn]
    exten => s,1,Set(FROM_DID=”6780000000)
    exten => s,n,NoOp(id is ${FROM_DID})
    exten => s,n,Goto(incoming1,s,1)

    [from-pstn1]
    exten => s,1,Set(FROM_DID=”6780000000)
    exten => s,n,NoOp(id is ${FROM_DID})
    exten => s,n,Goto(incoming1,s,1)

    [from-pstn2]
    exten => s,1,Set(FROM_DID=”6780000000)
    exten => s,n,NoOp(id is ${FROM_DID})
    exten => s,n,Goto(incoming1,s,1)

    [from-pstn3]
    exten => s,1,Set(FROM_DID=”6780000000)
    exten => s,n,NoOp(id is ${FROM_DID})
    exten => s,n,Goto(incoming1,s,1)

    [from-pstn4]
    exten => s,1,Set(FROM_DID=”6780000000)
    exten => s,n,NoOp(id is ${FROM_DID})
    exten => s,n,Goto(incoming3,s,1)

    [from-pstn5]
    exten => s,1,Set(FROM_DID=”6780000000)
    exten => s,n,NoOp(id is ${FROM_DID})
    exten => s,n,Goto(incoming2,s,1)

    [from-pstn6]
    exten => s,1,Set(FROM_DID=”6780000000)
    exten => s,n,NoOp(id is ${FROM_DID})
    exten => s,n,Goto(incoming1,s,1)

    [from-pstn7]
    exten => s,1,Set(FROM_DID=”6780000000)
    exten => s,n,NoOp(id is ${FROM_DID})
    exten => s,n,Goto(incoming1,s,1)

    [from-pstn8]
    exten => s,1,Set(FROM_DID=”6780000000)
    exten => s,n,NoOp(id is ${FROM_DID})
    exten => s,n,Goto(incoming1,s,1)

    [incoming1]
    include => from-internal
    include => parkedcalls
    exten => s,1,Answer
    exten => s,n,Wait(1)
    exten => s,n,Set(CHANNEL(musicclass)=QCI)
    exten => s,n,Set(TIMEOUT(digit)=5)
    exten => s,n,Set(TIMEOUT(response)=10)
    exten => s,n,Background(thank-you-for-calling)
    exten =>
    s,n,Dial(${QPHONE0}&${QPHONE1}&${QPHONE2}&${QPHONE3}&${QPHONE4}&${QPHONE5}&$
    {QPHONE6}&${QPHONE7},40,Ttr)
    exten => s,n,Hangup

    [incoming2]
    include => from-internal
    include => parkedcalls
    exten => s,1,Answer
    exten => s,n,Wait(1)
    exten => s,n,Set(CHANNEL(musicclass)=QCI)
    exten => s,n,Set(TIMEOUT(digit)=5)
    exten => s,n,Set(TIMEOUT(response)=10)
    exten => s,n,Background(thank-you-for-calling)
    exten =>
    s,n,Dial(${ACAPHONE0}&${ACAPHONE1}&${ACAPHONE2}&${ACAPHONE3}&${ACAPHONE4}&${
    ACAPHONE5}&${ACAPHONE6}&${ACAPHONE7},40,TTr)
    exten => s,n,Hangup

    [incoming3]
    include => from-internal
    include => parkedcalls
    exten => s,1,Answer
    exten => s,n,Wait(1)
    exten => s,n,Set(CHANNEL(musicclass)=QCI)
    exten => s,n,Set(TIMEOUT(digit)=5)
    exten => s,n,Set(TIMEOUT(response)=10)
    exten => s,n,Background(thank-you-for-calling)
    exten => s,n,GotoIfTime(17:01-09:00|*|*|*?closed,s,1)
    exten =>
    s,n,Dial(${GMNETPHONE0}&${GMNETPHONE1}&${GMNETPHONE2}&${GMNETPHONE3}&${GMNET
    PHONE4}&${GMNETPHONE5}&${GMNETPHONE6}&${GMNETPHONE7},40,Ttr)
    exten => s,n,Hangup

    [from-interal]
    include => dialout1
    include => dialout2
    include => dialout3
    include => parkedcalls
    include => intercom

    exten => 10,1,Macro(oneline,${QPHONE0})
    exten => 11,1,Macro(oneline,${QPHONE1})
    exten => 12,1,Macro(oneline,${QPHONE2})
    exten => 13,1,Macro(oneline,${QPHONE3})
    exten => 14,1,Macro(oneline,${QPHONE4})
    exten => 15,1,Macro(oneline,${QPHONE5})
    exten => 16,1,Macro(oneline,${QPHONE6})
    exten => 17,1,Macro(oneline,${QPHONE7})

    exten => 20,1,Macro(oneline,${ACAPHONE0})
    exten => 21,1,Macro(oneline,${ACAPHONE1})
    exten => 22,1,Macro(oneline,${ACAPHONE2})
    exten => 23,1,Macro(oneline,${ACAPHONE3})
    exten => 24,1,Macro(oneline,${ACAPHONE4})
    exten => 25,1,Macro(oneline,${ACAPHONE5})
    exten => 26,1,Macro(oneline,${ACAPHONE6})
    exten => 27,1,Macro(oneline,${ACAPHONE7})

    exten => 30,1,Macro(oneline,${GMNETPHONE0})
    exten => 31,1,Macro(oneline,${GMNETPHONE1})
    exten => 32,1,Macro(oneline,${GMNETPHONE2})
    exten => 33,1,Macro(oneline,${GMNETPHONE3})
    exten => 34,1,Macro(oneline,${GMNETPHONE4})
    exten => 35,1,Macro(oneline,${GMNETPHONE5})
    exten => 36,1,Macro(oneline,${GMNETPHONE6})
    exten => 37,1,Macro(oneline,${GMNETPHONE7})

    exten => 40,1,Macro(oneline,${QPHONE0})
    exten => 41,1,Macro(oneline,${QPHONE1})
    exten => 42,1,Macro(oneline,${QPHONE2})
    exten => 43,1,Macro(oneline,${QPHONE3})
    exten => 44,1,Macro(oneline,${QPHONE4})
    exten => 45,1,Macro(oneline,${QPHONE5})
    exten => 46,1,Macro(oneline,${QPHONE6})
    exten => 47,1,Macro(oneline,${QPHONE7})

    exten => 150,1,Macro(oneline,${EXTERNPHONE0})

    [macro-oneline]
    exten => s,1,Set(CHANNEL(musicclass)=default)
    exten => s,n,Dial(${ARG1},20,Ttr)
    exten => s,n,Voicemail(${MACRO_EXTEN})
    exten => s,n,Hangup
    exten => s,102,Voicemail(${MACRO_EXTEN})
    exten => s,103,Hangup

    [dialout1]
    include => from-internal
    include => 411
    exten => 911,1,Goto(nineoneone,s,1)
    exten => _1NXXNXXXXXX,n,Dial(SIP/v6781234567/${EXTEN},40,Ttr)
    exten => _NXXNXXXXXX,n,Dial(SIP/v6781234567/${EXTEN},40,Ttr)

    [dialout2]
    include => from-internal
    include => 411
    exten => 911,1,Goto(nineoneone,s,1)
    exten => _1NXXNXXXXXX,n,Dial(SIP/voipdialACA/${EXTEN},40,Ttr)
    exten => _NXXNXXXXXX,n,Dial(SIP/voipdialACA/${EXTEN},40,Ttr)

    [dialout3]
    include => from-internal
    include => 411
    exten => 911,1,Goto(nineoneone,s,1)
    exten => _1NXXNXXXXXX,n,Dial(SIP/v6781234567/${EXTEN},40,Ttr)
    exten => _1NXXNXXXXXX,n,Dial(SIP/v6781234567/${EXTEN},40,Ttr)

    [voicemail]
    exten => 8500,1,VoicemailMain
    exten => 8500,2,Hangup

    [411]
    exten => 411,1,Dial(SIP/v6781234567/18004664411,,Ttr)

    [nineoneone]
    exten => s,1,Set(SET_EMERG_FLAG=0)
    exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
    exten => s,n,Set(EMERGENCY=1,g)
    exten => s,n,Set(SET_EMERG_FLAG=1)
    exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})
    exten => s,s+2(trunkbusy),GotoIf($[${EMERGENCY} = 1]?inprogress)
    exten => s,n,SoftHangup(${EMERGENCY_TRUNK}-1)
    exten => s,n,Wait(12)
    exten => s,n,Goto(checkavail)
    exten => s,s+2(inprogress),Congestion
    exten => s,checkavail+101(notavail),Goto(trunkbusy)
    exten => h,1,GotoIf($[${SET_EMERG_FLAG} = 1]?3)
    exten => h,3,Set(EMERGENCY=0,g)

    [closed]
    exten => s,n,Dial(Dial(SIP/v6781234567/${CPHONE1},40,Ttr)
    exten => s,n,Hangup

    [intercom]
    exten => 59,1,SIPAddHeader(Call-Info: answer-after=0)
    exten =>
    59,2,Page(${GMNETPHONE0}&${GMNETPHONE1}&${GMNETPHONE2}&${GMNETPHONE3}&${GMNE
    TPHONE4}&${GMNETPHONE5}&${GMNETPHONE6}&${GMNETPHONE7}|d)
    exten => 59,3, Hangup

  • Here is the sip.conf portion for extension 150

    [150]
    deny=0.0.0.0/0.0.0.0
    type=friend
    secret=1234567890
    qualify=yes
    port=5060
    pickupgroup=
    permit=0.0.0.0/0.0.0.0
    nat=yes
    host=dynamic
    dtmfmode=rfc2833
    dial=SIP/150
    context=from-trunk
    canreinvite=no
    callgroup=
    callerid=device <150>
    accountcode=
    call-limit=50

    href=”mailto:treese65@gmail.com”>treese65@gmail.com

  • Todd

    Your context must be set to where you want your extension to start each
    time it dials out. Without getting into your dialplan code too much try
    changing the context to point to dialout1

    context=dialout1

    If dialout1 is working you should be able to dial.

    The best way to handle this is to create a context that when you dial from
    your phones it decieds if you have dialed an extension or an external
    number and then routes the call correclty. This way you can pickup an
    extension and dial either and get the desired results.

    Bryant

  • Here’s a debug for extension 150

    [Aug 30 11:34:53] VERBOSE[2099] config.c: == Parsing
    ‘/etc/asterisk/logger.conf’: [Aug 30 11:34:53] DEBUG[2099] config.c:
    Parsing /etc/asterisk/logger.conf
    [Aug 30 11:34:53] VERBOSE[2099] config.c: == Found
    [Aug 30 11:34:53] VERBOSE[2099] logger.c: Asterisk Event Logger restarted
    [Aug 30 11:34:53] VERBOSE[2099] logger.c: Asterisk Queue Logger restarted
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 0 [ 38]: OPTIONS
    sip:76.122.117.31:5060 SIP/2.0
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 1 [ 44]: Via:
    SIP/2.0/UDP 64.34.245.174:5060;branch=0
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 2 [ 38]: From:
    sip:pinger@voip.com;tag=7c9c6206
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 3 [ 26]: To:
    sip:76.122.117.31:5060
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 4 [ 47]: Call-ID:

    href=”mailto:89c833e4-9c8f2516-5bb7e3@64.34.245.174″>89c833e4-9c8f2516-5bb7e3@64.34.245.174
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 5 [ 15]: CSeq: 1 OPTIONS
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 6 [ 17]:
    Content-Length: 0
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 7 [ 0]:
    [Aug 30 11:34:53] DEBUG[2079] acl.c: Found IP address for this socket
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Setting SIP_TRANSPORT_UDP with
    address 10.0.1.102:5060
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Allocating new SIP dialog for

    href=”mailto:89c833e4-9c8f2516-5bb7e3@64.34.245.174″>89c833e4-9c8f2516-5bb7e3@64.34.245.174 – OPTIONS (No RTP)
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: **** Received OPTIONS (3) –
    Command in SIP OPTIONS
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Trying to put ‘SIP/2.0 404’
    onto UDP socket destined for 64.34.245.174:5060
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: SIP message could not be
    handled, bad request:
    href=”mailto:89c833e4-9c8f2516-5bb7e3@64.34.245.174″>89c833e4-9c8f2516-5bb7e3@64.34.245.174
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 0 [ 38]: OPTIONS
    sip:76.122.117.31:5060 SIP/2.0
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 1 [ 44]: Via:
    SIP/2.0/UDP 64.34.245.174:5060;branch=0
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 2 [ 38]: From:
    sip:pinger@voip.com;tag=1f9c6206
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 3 [ 26]: To:
    sip:76.122.117.31:5060
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 4 [ 47]: Call-ID:

    href=”mailto:89c833e4-3f8f2516-5bb7e3@64.34.245.174″>89c833e4-3f8f2516-5bb7e3@64.34.245.174
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 5 [ 15]: CSeq: 1 OPTIONS
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 6 [ 17]:
    Content-Length: 0
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 7 [ 0]:
    [Aug 30 11:34:53] DEBUG[2079] acl.c: Found IP address for this socket
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Setting SIP_TRANSPORT_UDP with
    address 10.0.1.102:5060
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Allocating new SIP dialog for

    href=”mailto:89c833e4-3f8f2516-5bb7e3@64.34.245.174″>89c833e4-3f8f2516-5bb7e3@64.34.245.174 – OPTIONS (No RTP)
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: **** Received OPTIONS (3) –
    Command in SIP OPTIONS
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Trying to put ‘SIP/2.0 404’
    onto UDP socket destined for 64.34.245.174:5060
    [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: SIP message could not be
    handled, bad request:
    href=”mailto:89c833e4-3f8f2516-5bb7e3@64.34.245.174″>89c833e4-3f8f2516-5bb7e3@64.34.245.174
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Auto destroying SIP dialog
    ’89c833e4-806e2516-79b7e3@64.34.245.174′
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Destroying SIP dialog

    href=”mailto:89c833e4-806e2516-79b7e3@64.34.245.174″>89c833e4-806e2516-79b7e3@64.34.245.174
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Auto destroying SIP dialog
    ’89c833e4-236e2516-79b7e3@64.34.245.174′
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Destroying SIP dialog

    href=”mailto:89c833e4-236e2516-79b7e3@64.34.245.174″>89c833e4-236e2516-79b7e3@64.34.245.174
    [Aug 30 11:34:54] VERBOSE[2079] chan_sip.c:
    < --- SIP read from UDP:97.80.176.231:5060 --->

    < ------------->
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Header 0 [ 0]:
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Body 0 [ 0]:
    [Aug 30 11:34:54] VERBOSE[2079] chan_sip.c:
    < --- SIP read from UDP:97.80.176.231:5060 --->
    INVITE sip:6789542133@qci.homeip.net SIP/2.0
    Via: SIP/2.0/UDP 10.11.17.24:5060;branch=z9hG4bK82d40955a28df1af
    Contact:
    Supported: replaces, timer, path
    P-Early-Media: Supported
    Call-ID:
    href=”mailto:62f35b2ee0ada782@10.11.17.24″>62f35b2ee0ada782@10.11.17.24
    CSeq: 21395 INVITE
    User-Agent: Grandstream GXP2000 1.2.3.5
    Max-Forwards: 70
    Allow:
    INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
    Content-Type: application/sdp
    Content-Length: 345

    v=0
    o=150 8000 8000 IN IP4 10.11.17.24
    s=SIP Call
    c=IN IP4 10.11.17.24
    t=0 0
    m=audio 5050 RTP/AVP 0 8 4 18 2 97 9 3
    a=sendrecv
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:4 G723/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:2 G726-32/8000
    a=rtpmap:97 iLBC/8000
    a=fmtp:97 mode=20
    a=rtpmap:9 G722/8000
    a=rtpmap:3 GSM/8000
    a=ptime:20

    < ------------->
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Header 0 [ 44]: INVITE
    sip:6789542133@qci.homeip.net SIP/2.0
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Header 1 [ 64]: Via:
    SIP/2.0/UDP 10.11.17.24:5060;branch=z9hG4bK82d40955a28df1af
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Header 2 [ 58]: From: “ATAP”
    ;tag=ee0cedf5f71d40f9
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Header 3 [ 35]: To:

    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Header 4 [ 49]: Contact:

    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Header 5 [ 32]: Supported:
    replaces, timer, path
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Header 6 [ 24]:
    P-Early-Media: Supported
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Header 7 [ 37]: Call-ID:

    href=”mailto:62f35b2ee0ada782@10.11.17.24″>62f35b2ee0ada782@10.11.17.24
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Header 8 [ 18]: CSeq: 21395
    INVITE
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Header 9 [ 39]: User-Agent:
    Grandstream GXP2000 1.2.3.5
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Header 11 [ 85]: Allow:
    INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Header 12 [ 29]:
    Content-Type: application/sdp
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Header 13 [ 19]:
    Content-Length: 345
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Header 14 [ 0]:
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Body 0 [ 3]: v=0
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Body 1 [ 34]: o=150 8000
    8000 IN IP4 10.11.17.24
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Body 2 [ 10]: s=SIP Call
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Body 3 [ 20]: c=IN IP4
    10.11.17.24
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Body 4 [ 5]: t=0 0
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Body 5 [ 38]: m=audio 5050
    RTP/AVP 0 8 4 18 2 97 9 3
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Body 6 [ 10]: a=sendrecv
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Body 7 [ 20]: a=rtpmap:0
    PCMU/8000
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Body 8 [ 20]: a=rtpmap:8
    PCMA/8000
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Body 9 [ 20]: a=rtpmap:4
    G723/8000
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Body 10 [ 21]: a=rtpmap:18
    G729/8000
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Body 11 [ 23]: a=rtpmap:2
    G726-32/8000
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Body 12 [ 21]: a=rtpmap:97
    iLBC/8000
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Body 13 [ 17]: a=fmtp:97
    mode=20
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Body 14 [ 20]: a=rtpmap:9
    G722/8000
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Body 15 [ 19]: a=rtpmap:3
    GSM/8000
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Body 16 [ 10]: a=ptime:20
    [Aug 30 11:34:54] VERBOSE[2079] chan_sip.c: — (14 headers 17 lines) —
    [Aug 30 11:34:54] DEBUG[2079] acl.c: Found IP address for this socket
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Setting SIP_TRANSPORT_UDP with
    address 10.0.1.102:5060
    [Aug 30 11:34:54] VERBOSE[2079] netsock.c: == Using SIP RTP TOS bits 184
    [Aug 30 11:34:54] VERBOSE[2079] netsock.c: == Using SIP RTP CoS mark 5
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Setting NAT on RTP to On
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Allocating new SIP dialog for

    href=”mailto:62f35b2ee0ada782@10.11.17.24″>62f35b2ee0ada782@10.11.17.24 – INVITE (With RTP)
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: **** Received INVITE (5) –
    Command in SIP INVITE
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Begin: parsing SIP “Supported:
    replaces, timer, path”
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Found SIP option: -replaces-
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Matched SIP option: replaces
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Found SIP option: -timer-
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Matched SIP option: timer
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Found SIP option: -path-
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Matched SIP option: path
    [Aug 30 11:34:54] VERBOSE[2079] chan_sip.c: Sending to 97.80.176.231 :
    5060 (NAT)
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Initializing initreq for
    method INVITE – callid
    href=”mailto:62f35b2ee0ada782@10.11.17.24″>62f35b2ee0ada782@10.11.17.24
    [Aug 30 11:34:54] VERBOSE[2079] chan_sip.c: Using INVITE request as
    basis request –
    href=”mailto:62f35b2ee0ada782@10.11.17.24″>62f35b2ee0ada782@10.11.17.24
    [Aug 30 11:34:54] VERBOSE[2079] chan_sip.c: Found peer ‘150’ for ‘150’
    from 97.80.176.231:5060
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Setting NAT on RTP to On
    [Aug 30 11:34:54] VERBOSE[2079] chan_sip.c:
    < --- Reliably Transmitting (NAT) to 97.80.176.231:5060 --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP
    10.11.17.24:5060;branch=z9hG4bK82d40955a28df1af;received=97.80.176.231
    Call-ID:
    href=”mailto:62f35b2ee0ada782@10.11.17.24″>62f35b2ee0ada782@10.11.17.24
    CSeq: 21395 INVITE
    Server: Asterisk PBX 1.6.2.11
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    WWW-Authenticate: Digest algorithm=MD5, realm=”asterisk”, nonce=”449798ce”
    Content-Length: 0

    < ------------>
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: *** SIP TIMER: Initializing
    retransmit timer on packet: Id #1721
    [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Trying to put ‘SIP/2.0 401’
    onto UDP socket destined for 97.80.176.231:5060
    [Aug 30 11:34:54] VERBOSE[2079] chan_sip.c: Scheduling destruction of
    SIP dialog ’62f35b2ee0ada782@10.11.17.24′ in 6400 ms (Method: INVITE)
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c:
    < --- SIP read from UDP:97.80.176.231:5060 --->
    ACK sip:6789542133@qci.homeip.net SIP/2.0
    Via: SIP/2.0/UDP 10.11.17.24:5060;branch=z9hG4bK82d40955a28df1af
    Contact:
    Supported: path
    Call-ID:
    href=”mailto:62f35b2ee0ada782@10.11.17.24″>62f35b2ee0ada782@10.11.17.24
    CSeq: 21395 ACK
    User-Agent: Grandstream GXP2000 1.2.3.5
    Max-Forwards: 70
    Allow:
    INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
    Content-Length: 0

    < ------------->
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 0 [ 41]: ACK
    sip:6789542133@qci.homeip.net SIP/2.0
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 1 [ 64]: Via:
    SIP/2.0/UDP 10.11.17.24:5060;branch=z9hG4bK82d40955a28df1af
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 2 [ 58]: From: “ATAP”
    ;tag=ee0cedf5f71d40f9
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 3 [ 50]: To:
    ;tag=as277ae45d
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 4 [ 49]: Contact:

    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 5 [ 15]: Supported: path
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 6 [ 37]: Call-ID:

    href=”mailto:62f35b2ee0ada782@10.11.17.24″>62f35b2ee0ada782@10.11.17.24
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 7 [ 15]: CSeq: 21395 ACK
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 8 [ 39]: User-Agent:
    Grandstream GXP2000 1.2.3.5
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 9 [ 16]: Max-Forwards: 70
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 10 [ 85]: Allow:
    INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 11 [ 17]:
    Content-Length: 0
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 12 [ 0]:
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: — (12 headers 0 lines) —
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: **** Received ACK (6) –
    Command in SIP ACK
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: ** SIP TIMER: Cancelling
    retransmit of packet (reply received) Retransid #1721
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Stopping retransmission on
    ’62f35b2ee0ada782@10.11.17.24′ of Response 21395: Match Found
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c:
    < --- SIP read from UDP:97.80.176.231:5060 --->
    INVITE sip:6789542133@qci.homeip.net SIP/2.0
    Via: SIP/2.0/UDP 10.11.17.24:5060;branch=z9hG4bK45bf7291c591a19a
    Contact:

    Supported: replaces, timer, path
    P-Early-Media: Supported
    Authorization: Digest username=”150″, realm=”asterisk”, algorithm=MD5,
    uri=”sip:6789542133@qci.homeip.net”, nonce=”449798ce”,
    response=”237f1b41f316074f60903086366682b4″
    Call-ID:
    href=”mailto:62f35b2ee0ada782@10.11.17.24″>62f35b2ee0ada782@10.11.17.24
    CSeq: 21396 INVITE
    User-Agent: Grandstream GXP2000 1.2.3.5
    Max-Forwards: 70
    Allow:
    INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
    Content-Type: application/sdp
    Content-Length: 345

    v=0
    o=150 8000 8001 IN IP4 10.11.17.24
    s=SIP Call
    c=IN IP4 10.11.17.24
    t=0 0
    m=audio 5050 RTP/AVP 0 8 4 18 2 97 9 3
    a=sendrecv
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:4 G723/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:2 G726-32/8000
    a=rtpmap:97 iLBC/8000
    a=fmtp:97 mode=20
    a=rtpmap:9 G722/8000
    a=rtpmap:3 GSM/8000
    a=ptime:20

    < ------------->
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 0 [ 44]: INVITE
    sip:6789542133@qci.homeip.net SIP/2.0
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 1 [ 64]: Via:
    SIP/2.0/UDP 10.11.17.24:5060;branch=z9hG4bK45bf7291c591a19a
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 2 [ 58]: From: “ATAP”
    ;tag=ee0cedf5f71d40f9
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 3 [ 35]: To:

    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 4 [ 49]: Contact:

    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 5 [ 32]: Supported:
    replaces, timer, path
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 6 [ 24]:
    P-Early-Media: Supported
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 7 [169]:
    Authorization: Digest username=”150″, realm=”asterisk”, algorithm=MD5,
    uri=”sip:6789542133@qci.homeip.net”, nonce=”449798ce”,
    response=”237f1b41f316074f60903086366682b4″
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 8 [ 37]: Call-ID:

    href=”mailto:62f35b2ee0ada782@10.11.17.24″>62f35b2ee0ada782@10.11.17.24
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 9 [ 18]: CSeq: 21396
    INVITE
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 10 [ 39]: User-Agent:
    Grandstream GXP2000 1.2.3.5
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 11 [ 16]: Max-Forwards: 70
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 12 [ 85]: Allow:
    INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 13 [ 29]:
    Content-Type: application/sdp
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 14 [ 19]:
    Content-Length: 345
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 15 [ 0]:
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Body 0 [ 3]: v=0
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Body 1 [ 34]: o=150 8000
    8001 IN IP4 10.11.17.24
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Body 2 [ 10]: s=SIP Call
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Body 3 [ 20]: c=IN IP4
    10.11.17.24
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Body 4 [ 5]: t=0 0
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Body 5 [ 38]: m=audio 5050
    RTP/AVP 0 8 4 18 2 97 9 3
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Body 6 [ 10]: a=sendrecv
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Body 7 [ 20]: a=rtpmap:0
    PCMU/8000
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Body 8 [ 20]: a=rtpmap:8
    PCMA/8000
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Body 9 [ 20]: a=rtpmap:4
    G723/8000
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Body 10 [ 21]: a=rtpmap:18
    G729/8000
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Body 11 [ 23]: a=rtpmap:2
    G726-32/8000
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Body 12 [ 21]: a=rtpmap:97
    iLBC/8000
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Body 13 [ 17]: a=fmtp:97
    mode=20
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Body 14 [ 20]: a=rtpmap:9
    G722/8000
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Body 15 [ 19]: a=rtpmap:3
    GSM/8000
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Body 16 [ 10]: a=ptime:20
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: — (15 headers 17 lines) —
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: **** Received INVITE (5) –
    Command in SIP INVITE
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: Sending to 97.80.176.231 :
    5060 (NAT)
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Initializing initreq for
    method INVITE – callid
    href=”mailto:62f35b2ee0ada782@10.11.17.24″>62f35b2ee0ada782@10.11.17.24
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: Using INVITE request as
    basis request –
    href=”mailto:62f35b2ee0ada782@10.11.17.24″>62f35b2ee0ada782@10.11.17.24
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: Found peer ‘150’ for ‘150’
    from 97.80.176.231:5060
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Setting NAT on RTP to On
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Processing session-level SDP
    v=0… UNSUPPORTED.
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Processing session-level SDP
    o=150 8000 8001 IN IP4 10.11.17.24… UNSUPPORTED.
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Processing session-level SDP
    s=SIP Call… UNSUPPORTED.
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Processing session-level SDP
    c=IN IP4 10.11.17.24… OK.
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Processing session-level SDP
    t=0 0… UNSUPPORTED.
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: Found RTP audio format 0
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: Found RTP audio format 8
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: Found RTP audio format 4
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: Found RTP audio format 18
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: Found RTP audio format 2
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: Found RTP audio format 97
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: Found RTP audio format 9
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: Found RTP audio format 3
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Processing media-level (audio)
    SDP a=sendrecv… OK.
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: Found audio description
    format PCMU for ID 0
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Processing media-level (audio)
    SDP a=rtpmap:0 PCMU/8000… OK.
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: Found audio description
    format PCMA for ID 8
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Processing media-level (audio)
    SDP a=rtpmap:8 PCMA/8000… OK.
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: Found audio description
    format G723 for ID 4
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Processing media-level (audio)
    SDP a=rtpmap:4 G723/8000… OK.
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: Found audio description
    format G729 for ID 18
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Processing media-level (audio)
    SDP a=rtpmap:18 G729/8000… OK.
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: Found audio description
    format G726-32 for ID 2
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Processing media-level (audio)
    SDP a=rtpmap:2 G726-32/8000… OK.
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: Found audio description
    format iLBC for ID 97
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Processing media-level (audio)
    SDP a=rtpmap:97 iLBC/8000… OK.
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Processing media-level (audio)
    SDP a=fmtp:97 mode=20… UNSUPPORTED.
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: Found audio description
    format G722 for ID 9
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Processing media-level (audio)
    SDP a=rtpmap:9 G722/8000… OK.
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: Found audio description
    format GSM for ID 3
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Processing media-level (audio)
    SDP a=rtpmap:3 GSM/8000… OK.
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Processing media-level (audio)
    SDP a=ptime:20… OK.
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: Capabilities: us – 0x4
    (ulaw), peer – audio=0x1d0f
    (g723|gsm|ulaw|alaw|g726|g729|ilbc|g722)/video=0x0 (nothing)/text=0x0
    (nothing), combined – 0x4 (ulaw)
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: Non-codec capabilities
    (dtmf): us – 0x1 (telephone-event), peer – 0x0 (nothing), combined – 0x0
    (nothing)
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: Peer audio RTP is at port
    10.11.17.24:5050
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: We’re settling with these
    formats: 0x4 (ulaw)
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Checking SIP call limits for
    device 150
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Updating call counter for
    incoming call
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Call from peer ‘150’ is 1 out
    of 50
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: Looking for 6789542133 in
    extensions.conf (domain qci.homeip.net)
    [Aug 30 11:34:55] DEBUG[2065] devicestate.c: No provider found, checking
    channel drivers for SIP – 150
    [Aug 30 11:34:55] DEBUG[2065] chan_sip.c: Checking device state for peer 150
    [Aug 30 11:34:55] DEBUG[2065] devicestate.c: Changing state for SIP/150
    – state 2 (In use)
    [Aug 30 11:34:55] DEBUG[2065] devicestate.c: device ‘SIP/150’ state ‘2’
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c:
    < --- Reliably Transmitting (NAT) to 97.80.176.231:5060 --->
    SIP/2.0 404 Not Found
    Via: SIP/2.0/UDP
    10.11.17.24:5060;branch=z9hG4bK45bf7291c591a19a;received=97.80.176.231
    Call-ID:
    href=”mailto:62f35b2ee0ada782@10.11.17.24″>62f35b2ee0ada782@10.11.17.24
    CSeq: 21396 INVITE
    Server: Asterisk PBX 1.6.2.11
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    Content-Length: 0

    < ------------>
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: *** SIP TIMER: Initializing
    retransmit timer on packet: Id #1723
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Trying to put ‘SIP/2.0 404’
    onto UDP socket destined for 97.80.176.231:5060
    [Aug 30 11:34:55] DEBUG[2074] app_queue.c: Device ‘SIP/150’ changed to
    state ‘2’ (In use) but we don’t care because they’re not a member of any
    queue.
    [Aug 30 11:34:55] NOTICE[2079] chan_sip.c: Call from ‘150’ to extension
    ‘6789542133’ rejected because extension not found in context
    ‘extensions.conf’.
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Updating call counter for
    incoming call
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Call from peer ‘150’ removed
    from call limit 50
    [Aug 30 11:34:55] DEBUG[2065] devicestate.c: No provider found, checking
    channel drivers for SIP – 150
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: Scheduling destruction of
    SIP dialog ’62f35b2ee0ada782@10.11.17.24′ in 6400 ms (Method: INVITE)
    [Aug 30 11:34:55] DEBUG[2065] chan_sip.c: Checking device state for peer 150
    [Aug 30 11:34:55] DEBUG[2065] devicestate.c: Changing state for SIP/150
    – state 1 (Not in use)
    [Aug 30 11:34:55] DEBUG[2065] devicestate.c: device ‘SIP/150’ state ‘1’
    [Aug 30 11:34:55] DEBUG[2074] app_queue.c: Device ‘SIP/150’ changed to
    state ‘1’ (Not in use) but we don’t care because they’re not a member of
    any queue.
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c:
    < --- SIP read from UDP:97.80.176.231:5060 --->
    ACK sip:6789542133@qci.homeip.net SIP/2.0
    Via: SIP/2.0/UDP 10.11.17.24:5060;branch=z9hG4bK45bf7291c591a19a
    Contact:
    Supported: path
    Authorization: Digest username=”150″, realm=”asterisk”, algorithm=MD5,
    uri=”sip:6789542133@qci.homeip.net”, nonce=”449798ce”,
    response=”237f1b41f316074f60903086366682b4″
    Call-ID:
    href=”mailto:62f35b2ee0ada782@10.11.17.24″>62f35b2ee0ada782@10.11.17.24
    CSeq: 21396 ACK
    User-Agent: Grandstream GXP2000 1.2.3.5
    Max-Forwards: 70
    Allow:
    INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
    Content-Length: 0

    < ------------->
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 0 [ 41]: ACK
    sip:6789542133@qci.homeip.net SIP/2.0
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 1 [ 64]: Via:
    SIP/2.0/UDP 10.11.17.24:5060;branch=z9hG4bK45bf7291c591a19a
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 2 [ 58]: From: “ATAP”
    ;tag=ee0cedf5f71d40f9
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 3 [ 50]: To:
    ;tag=as277ae45d
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 4 [ 49]: Contact:

    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 5 [ 15]: Supported: path
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 6 [169]:
    Authorization: Digest username=”150″, realm=”asterisk”, algorithm=MD5,
    uri=”sip:6789542133@qci.homeip.net”, nonce=”449798ce”,
    response=”237f1b41f316074f60903086366682b4″
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 7 [ 37]: Call-ID:

    href=”mailto:62f35b2ee0ada782@10.11.17.24″>62f35b2ee0ada782@10.11.17.24
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 8 [ 15]: CSeq: 21396 ACK
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 9 [ 39]: User-Agent:
    Grandstream GXP2000 1.2.3.5
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 11 [ 85]: Allow:
    INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 12 [ 17]:
    Content-Length: 0
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Header 13 [ 0]:
    [Aug 30 11:34:55] VERBOSE[2079] chan_sip.c: — (13 headers 0 lines) —
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: **** Received ACK (6) –
    Command in SIP ACK
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: ** SIP TIMER: Cancelling
    retransmit of packet (reply received) Retransid #1723
    [Aug 30 11:34:55] DEBUG[2079] chan_sip.c: Stopping retransmission on
    ’62f35b2ee0ada782@10.11.17.24′ of Response 21396: Match Found
    [Aug 30 11:35:01] DEBUG[2079] chan_sip.c: Auto destroying SIP dialog
    ‘447f358d384b5f3d194e157e19053d7d@10.0.1.102’
    [Aug 30 11:35:01] DEBUG[2079] chan_sip.c: Destroying SIP dialog

    href=”mailto:447f358d384b5f3d194e157e19053d7d@10.0.1.102″>447f358d384b5f3d194e157e19053d7d@10.0.1.102
    [Aug 30 11:35:01] DEBUG[2079] chan_sip.c: Auto destroying SIP dialog
    ’08bfa4c101c769a876067caa779efca5@10.0.1.102′
    [Aug 30 11:35:01] DEBUG[2079] chan_sip.c: Destroying SIP dialog

    href=”mailto:08bfa4c101c769a876067caa779efca5@10.0.1.102″>08bfa4c101c769a876067caa779efca5@10.0.1.102
    [Aug 30 11:35:01] DEBUG[2079] chan_sip.c: Auto destroying SIP dialog
    ’66cafdb06008db6d566d39fc3477a659@10.0.1.102′
    [Aug 30 11:35:01] DEBUG[2079] chan_sip.c: Destroying SIP dialog

    href=”mailto:66cafdb06008db6d566d39fc3477a659@10.0.1.102″>66cafdb06008db6d566d39fc3477a659@10.0.1.102
    [Aug 30 11:35:01] DEBUG[2079] chan_sip.c: Auto destroying SIP dialog
    ‘1fe133904ae2c3f9229778514394e48d@10.0.1.102’
    [Aug 30 11:35:01] DEBUG[2079] chan_sip.c: Destroying SIP dialog

    href=”mailto:1fe133904ae2c3f9229778514394e48d@10.0.1.102″>1fe133904ae2c3f9229778514394e48d@10.0.1.102
    [Aug 30 11:35:01] DEBUG[2079] chan_sip.c: Auto destroying SIP dialog
    ’62f35b2ee0ada782@10.11.17.24′
    [Aug 30 11:35:01] DEBUG[2079] chan_sip.c: Destroying SIP dialog

    href=”mailto:62f35b2ee0ada782@10.11.17.24″>62f35b2ee0ada782@10.11.17.24
    [Aug 30 11:35:01] VERBOSE[2079] chan_sip.c: Really destroying SIP dialog
    ’62f35b2ee0ada782@10.11.17.24′ Method: ACK
    [Aug 30 11:35:01] DEBUG[2079] chan_sip.c: Auto destroying SIP dialog
    ’05b171257bbe690373de757163fdbb6c@10.0.1.102′
    [Aug 30 11:35:01] DEBUG[2079] chan_sip.c: Destroying SIP dialog

    href=”mailto:05b171257bbe690373de757163fdbb6c@10.0.1.102″>05b171257bbe690373de757163fdbb6c@10.0.1.102
    [Aug 30 11:35:01] DEBUG[2079] chan_sip.c: Auto destroying SIP dialog
    ’48d5fa817d96bb632813ff2943107888@10.0.1.102′
    [Aug 30 11:35:01] DEBUG[2079] chan_sip.c: Destroying SIP dialog

    href=”mailto:48d5fa817d96bb632813ff2943107888@10.0.1.102″>48d5fa817d96bb632813ff2943107888@10.0.1.102
    [Aug 30 11:35:11] DEBUG[2079] chan_sip.c: Allocating new SIP dialog for

    href=”mailto:01c252e3142ee4a01cffcb0a3f854026@10.0.1.102″>01c252e3142ee4a01cffcb0a3f854026@10.0.1.102 – OPTIONS (No RTP)
    [Aug 30 11:35:11] DEBUG[2079] acl.c: Found IP address for this socket
    [Aug 30 11:35:11] DEBUG[2079] chan_sip.c: Setting SIP_TRANSPORT_UDP with
    address 10.0.1.102:5060
    [Aug 30 11:35:11] DEBUG[2079] chan_sip.c: Initializing initreq for
    method OPTIONS – callid
    href=”mailto:29bd1b0b32dd25b13a44c3236c030727@10.0.1.102″>29bd1b0b32dd25b13a44c3236c030727@10.0.1.102
    [Aug 30 11:35:11] DEBUG[2079] chan_sip.c: Trying to put ‘OPTIONS sip’
    onto UDP socket destined for 10.0.1.132:5060
    [Aug 30 11:35:11] DEBUG[2079] chan_sip.c: Stopping retransmission on
    ’29bd1b0b32dd25b13a44c3236c030727@10.0.1.102′ of Request 102: Match Found
    [Aug 30 11:35:11] DEBUG[2079] chan_sip.c: Destroying SIP dialog

    href=”mailto:29bd1b0b32dd25b13a44c3236c030727@10.0.1.102″>29bd1b0b32dd25b13a44c3236c030727@10.0.1.102
    [Aug 30 11:35:13] DEBUG[2079] chan_sip.c: Allocating new SIP dialog for

    href=”mailto:1be7697f05f93db02d5286ca2671685d@10.0.1.102″>1be7697f05f93db02d5286ca2671685d@10.0.1.102 – OPTIONS (No RTP)
    [Aug 30 11:35:13] DEBUG[2079] acl.c: Found IP address for this socket
    [Aug 30 11:35:13] DEBUG[2079] chan_sip.c: Setting SIP_TRANSPORT_UDP with
    address 10.0.1.102:5060
    [Aug 30 11:35:13] DEBUG[2079] chan_sip.c: Initializing initreq for
    method OPTIONS – callid
    href=”mailto:67f02b79570a1edf0525fb6e0ffdeef2@10.0.1.102″>67f02b79570a1edf0525fb6e0ffdeef2@10.0.1.102
    [Aug 30 11:35:13] DEBUG[2079] chan_sip.c: Trying to put ‘OPTIONS sip’
    onto UDP socket destined for 10.0.1.132:5062
    [Aug 30 11:35:13] DEBUG[2079] chan_sip.c: Stopping retransmission on
    ’67f02b79570a1edf0525fb6e0ffdeef2@10.0.1.102′ of Request 102: Match Found
    [Aug 30 11:35:13] DEBUG[2079] chan_sip.c: Destroying SIP dialog

    href=”mailto:67f02b79570a1edf0525fb6e0ffdeef2@10.0.1.102″>67f02b79570a1edf0525fb6e0ffdeef2@10.0.1.102
    [Aug 30 11:35:15] DEBUG[2079] chan_sip.c: Allocating new SIP dialog for

    href=”mailto:7088bae963d26a31717a8cc3476d1196@10.0.1.102″>7088bae963d26a31717a8cc3476d1196@10.0.1.102 – OPTIONS (No RTP)
    [Aug 30 11:35:15] DEBUG[2079] acl.c: Found IP address for this socket
    [Aug 30 11:35:15] DEBUG[2079] chan_sip.c: Setting SIP_TRANSPORT_UDP with
    address 10.0.1.102:5060
    [Aug 30 11:35:15] DEBUG[2079] chan_sip.c: Initializing initreq for
    method OPTIONS – callid
    href=”mailto:76b217753bdfbedd260a49f378f09993@10.0.1.102″>76b217753bdfbedd260a49f378f09993@10.0.1.102
    [Aug 30 11:35:15] DEBUG[2079] chan_sip.c: Trying to put ‘OPTIONS sip’
    onto UDP socket destined for 10.0.1.132:5064
    [Aug 30 11:35:15] DEBUG[2079] chan_sip.c: Stopping retransmission on
    ’76b217753bdfbedd260a49f378f09993@10.0.1.102′ of Request 102: Match Found
    [Aug 30 11:35:15] DEBUG[2079] chan_sip.c: Destroying SIP dialog

    href=”mailto:76b217753bdfbedd260a49f378f09993@10.0.1.102″>76b217753bdfbedd260a49f378f09993@10.0.1.102
    [Aug 30 11:35:16] DEBUG[2079] acl.c: Found IP address for this socket
    [Aug 30 11:35:16] DEBUG[2079] chan_sip.c: Setting SIP_TRANSPORT_UDP with
    address 10.0.1.102:5060
    [Aug 30 11:35:16] DEBUG[2079] chan_sip.c: Allocating new SIP dialog for

    href=”mailto:89c833e4-75603516-bcb7e3@64.34.245.174″>89c833e4-75603516-bcb7e3@64.34.245.174 – OPTIONS (No RTP)
    [Aug 30 11:35:16] DEBUG[2079] chan_sip.c: **** Received OPTIONS (3) –
    Command in SIP OPTIONS
    [Aug 30 11:35:16] DEBUG[2079] chan_sip.c: Trying to put ‘SIP/2.0 404’
    onto UDP socket destined for 64.34.245.174:5060
    [Aug 30 11:35:16] DEBUG[2079] chan_sip.c: SIP message could not be
    handled, bad request:
    href=”mailto:89c833e4-75603516-bcb7e3@64.34.245.174″>89c833e4-75603516-bcb7e3@64.34.245.174
    [Aug 30 11:35:16] DEBUG[2079] acl.c: Found IP address for this socket
    [Aug 30 11:35:16] DEBUG[2079] chan_sip.c: Setting SIP_TRANSPORT_UDP with
    address 10.0.1.102:5060
    [Aug 30 11:35:16] DEBUG[2079] chan_sip.c: Allocating new SIP dialog for

    href=”mailto:89c833e4-98603516-bcb7e3@64.34.245.174″>89c833e4-98603516-bcb7e3@64.34.245.174 – OPTIONS (No RTP)
    [Aug 30 11:35:16] DEBUG[2079] chan_sip.c: **** Received OPTIONS (3) –
    Command in SIP OPTIONS
    [Aug 30 11:35:16] DEBUG[2079] chan_sip.c: Trying to put ‘SIP/2.0 404’
    onto UDP socket destined for 64.34.245.174:5060
    [Aug 30 11:35:16] DEBUG[2079] chan_sip.c: SIP message could not be
    handled, bad request:
    href=”mailto:89c833e4-98603516-bcb7e3@64.34.245.174″>89c833e4-98603516-bcb7e3@64.34.245.174
    [Aug 30 11:35:17] DEBUG[2079] chan_sip.c: Allocating new SIP dialog for

    href=”mailto:0d3bb3e908ff7e923527a72d3b7a5ee2@10.0.1.102″>0d3bb3e908ff7e923527a72d3b7a5ee2@10.0.1.102 – OPTIONS (No RTP)
    [Aug 30 11:35:17] DEBUG[2079] acl.c: Found IP address for this socket
    [Aug 30 11:35:17] DEBUG[2079] chan_sip.c: Setting SIP_TRANSPORT_UDP with
    address 10.0.1.102:5060
    [Aug 30 11:35:17] DEBUG[2079] chan_sip.c: Initializing initreq for
    method OPTIONS – callid
    href=”mailto:6be9fbbb285c466f630fa66b480c8b29@10.0.1.102″>6be9fbbb285c466f630fa66b480c8b29@10.0.1.102
    [Aug 30 11:35:17] DEBUG[2079] chan_sip.c: Trying to put ‘OPTIONS sip’
    onto UDP socket destined for 10.0.1.132:5066
    [Aug 30 11:35:17] DEBUG[2079] chan_sip.c: Stopping retransmission on
    ‘6be9fbbb285c466f630fa66b480c8b29@10.0.1.102’ of Request 102: Match Found
    [Aug 30 11:35:17] DEBUG[2079] chan_sip.c: Destroying SIP dialog

    href=”mailto:6be9fbbb285c466f630fa66b480c8b29@10.0.1.102″>6be9fbbb285c466f630fa66b480c8b29@10.0.1.102
    [Aug 30 11:35:17] DEBUG[2079] chan_sip.c: Auto destroying SIP dialog
    ’89c833e4-993f2516-dab7e3@64.34.245.174′
    [Aug 30 11:35:17] DEBUG[2079] chan_sip.c: Destroying SIP dialog

    href=”mailto:89c833e4-993f2516-dab7e3@64.34.245.174″>89c833e4-993f2516-dab7e3@64.34.245.174
    [Aug 30 11:35:17] DEBUG[2079] chan_sip.c: Auto destroying SIP dialog
    ’89c833e4-bc3f2516-dab7e3@64.34.245.174′
    [Aug 30 11:35:17] DEBUG[2079] chan_sip.c: Destroying SIP dialog

    href=”mailto:89c833e4-bc3f2516-dab7e3@64.34.245.174″>89c833e4-bc3f2516-dab7e3@64.34.245.174
    [Aug 30 11:35:18] DEBUG[2079] acl.c: Found IP address for this socket
    [Aug 30 11:35:18] DEBUG[2079] chan_sip.c: Setting SIP_TRANSPORT_UDP with
    address 10.0.1.102:5060
    [Aug 30 11:35:18] DEBUG[2079] chan_sip.c: Allocating new SIP dialog for

    href=”mailto:89c833e4-fa703516-dcb7e3@64.34.245.174″>89c833e4-fa703516-dcb7e3@64.34.245.174 – OPTIONS (No RTP)
    [Aug 30 11:35:18] DEBUG[2079] chan_sip.c: **** Received OPTIONS (3) –
    Command in SIP OPTIONS
    [Aug 30 11:35:18] DEBUG[2079] chan_sip.c: Trying to put ‘SIP/2.0 404’
    onto UDP socket destined for 64.34.245.174:5060
    [Aug 30 11:35:18] DEBUG[2079] chan_sip.c: SIP message could not be
    handled, bad request:
    href=”mailto:89c833e4-fa703516-dcb7e3@64.34.245.174″>89c833e4-fa703516-dcb7e3@64.34.245.174
    [Aug 30 11:35:19] DEBUG[2079] chan_sip.c: Auto destroying SIP dialog
    ’89c833e4-1f4f2516-fab7e3@64.34.245.174′
    [Aug 30 11:35:19] DEBUG[2079] chan_sip.c: Destroying SIP dialog

    href=”mailto:89c833e4-1f4f2516-fab7e3@64.34.245.174″>89c833e4-1f4f2516-fab7e3@64.34.245.174

  • Unfortunately, that didn’t work. The phone is still giving me a 404
    error.

    I have my own system that is 1.6.2.7 with Grandstream phones that works
    fine. Using it as a guide, I built this server for a client which also
    has Grandstream phones.

    Last week, it dialed out fine. Since the weekend, no dialing at all.

    href=”mailto:asterisk-users@lists.digium.com”>asterisk-users@lists.digium.com
    href=”mailto:treese65@gmail.com”>treese65@gmail.com

  • In the future, simply attach your debug log to your email. Here is
    your problem:

    [Aug 30 11:34:55] NOTICE[2079] chan_sip.c: Call from ‘150’ to extension
    ‘6789542133’ rejected because extension not found in context
    ‘extensions.conf’.

  • I actually found that one and corrected it. I have replaced the
    context with the from-internal, remote, and dialout1. Each has produced
    the same results of a 404 error.

  • [Aug 30 15:27:41] NOTICE[11568] chan_sip.c: Call from ‘150’ to
    extension ‘6789542133’ rejected because extension not found in context
    ‘remote’.

    So, again, a context problem. You can confirm by entering:

    *CLI> dialplan show 6789542133@remote

  • asterisk*CLI> dialplan show 6789542133@remote
    There is no existence of ‘remote’ context
    Command ‘dialplan show 6789542133@remote’ failed.
    asterisk*CLI>

  • From extensions.conf

    [remote]
    include => from-internal
    include => dialout1
    include => dialout2
    include => dialout3
    include => intercom
    exten => 150,1,Macro(oneline,${EXTERNPHONE0})

    [dialout1]
    include => from-internal
    include => 411
    include => remote
    exten => 911,1,Goto(nineoneone,s,1)
    exten => _1NXXNXXXXXX,n,Dial(SIP/v6787500600/${EXTEN},40,Ttr)
    [dialout2]
    include => from-internal
    include => 411
    include => remote
    exten => 911,1,Goto(nineoneone,s,1)
    exten => _1NXXNXXXXXX,n,Dial(SIP/voip.comACA/${EXTEN},40,Ttr)
    [dialout3]
    include => from-internal
    include => 411
    include => remote
    exten => 911,1,Goto(nineoneone,s,1)
    exten => _1NXXNXXXXXX,n,Dial(SIP/v6783747049/${EXTEN},40,Ttr)

  • Todd–

    There is probably some nifty anti-infinite-recursion code in the
    extensions.conf parser,
    to keep asterisk from going into infinite loops trying to descend into the
    right context.

    In your dialplan, [remote] includes dialout1, dialout2, dialout3, and each
    of those
    include remote.

    Straighten out that mess and maybe things might work. Just a guess, but
    worth a try!

    murf

  • Interesting things going on herel.

    After your suggestions, Steve. I reran the dialplan show
    16789542133@remote command with the below results.

    Phone calls are geting the 404 error and the NOTICE on the console.
    [Aug 31 11:06:46] NOTICE[11884]: chan_sip.c:20161 handle_request_invite:
    Call from ‘150’ to extension ‘16789542133’ rejected because extension
    not found in context ‘remote’.

    asterisk*CLI> dialplan show 16789542133@remote
    [ Included context ‘dialout1’ created by ‘pbx_config’ ]
    ‘_1NXXNXXXXXX’ => -2. Dial(SIP/v6787500600/${EXTEN},40,Ttr)
    [pbx_config]

    [ Included context ‘dialout1’ created by ‘pbx_config’ ]
    ‘_1NXXNXXXXXX’ => -2. Dial(SIP/v6787500600/${EXTEN},40,Ttr)
    [pbx_config]

    [ Included context ‘dialout2’ created by ‘pbx_config’ ]
    ‘_1NXXNXXXXXX’ => -2. Dial(SIP/voip.comACA/${EXTEN},40,Ttr)
    [pbx_config]

    [ Included context ‘dialout3’ created by ‘pbx_config’ ]
    ‘_1NXXNXXXXXX’ => -2. Dial(SIP/v6783747049/${EXTEN},40,Ttr)
    [pbx_config]

    [ Included context ‘dialout1’ created by ‘pbx_config’ ]
    ‘_1NXXNXXXXXX’ => -2. Dial(SIP/v6787500600/${EXTEN},40,Ttr)
    [pbx_config]

    [ Included context ‘dialout2’ created by ‘pbx_config’ ]
    ‘_1NXXNXXXXXX’ => -2. Dial(SIP/voip.comACA/${EXTEN},40,Ttr)
    [pbx_config]

    [ Included context ‘dialout3’ created by ‘pbx_config’ ]
    ‘_1NXXNXXXXXX’ => -2. Dial(SIP/v6783747049/${EXTEN},40,Ttr)
    [pbx_config]

    -= 7 extensions (7 priorities) in 7 contexts. =-
    [Aug 31 11:03:38] WARNING[11903]: pbx.c:5741 show_dialplan_helper:
    Avoiding circular include of from-internal within remote

  • Well, that could be done, and probably would be a good tactic if you have
    lots of DID’s
    and want to do db lookup or something to direct the next call leg.

    But, if you only have one or two DID’s, all the machinery and programming
    seem
    a bit overkill.

    murf