Thanks very much. See below. I do not have a zaptel.conf
**********************************
Extensions.conf
[globals]
[extensions]
;directory app
exten => 9,1,Directory(extensions)
; echo latency test
exten => 10,1,Playback(demo-echotest)
exten => 10,2,Playback(beep)
exten => 10,3,Echo
exten => 10,4,Playback(demo-echodone)
exten => 10,5,Hangup
; 1000Hz tone test
exten => 11,1,Milliwatt()
exten => 11,2,Hangup
; exten for recording greetings/menus
exten => 12,1,Authenticate(12|)
exten => 12,1,Wait(2)
exten => 12,2,Record(/var/lib/asterisk/sounds/swelcome:gsm)
exten => 12,3,Wait(2)
exten => 12,4,Playback(/var/lib/asterisk/sounds/swelcome)
exten => 12,5,Wait(2)
exten => 12,6,Hangup
; date and time check
exten => 13,1,DateTime()
exten => 13,2,Wait(1)
exten => 13,3,DateTime()
exten => 13,4,Hangup
; extension check
exten => 14,1,Wait(1)
exten => 14,2,SayDigits(${CALLERIDNUM})
exten => 14,3,Wait(1)
exten => 14,4,SayDigits(${CALLERIDNUM})
exten => 14,5,Hangup
; user's voicemail
exten => 15,1,VoicemailMain
exten => 15,2,Hangup
; SIP 5000
exten => 5000,1,Dial(SIP/5000)
exten => 5000,2,Voicemail(u${EXTEN})
exten => 5000,3,Hangup
exten => 5000,102,Voicemail(b${EXTEN})
exten => 5000,103,Hangup
; SIP 5001
exten => 5001,1,Dial(SIP/5001)
exten => 5001,2,Voicemail(u${EXTEN})
exten => 5001,3,Hangup
exten => 5001,102,Voicemail(b${EXTEN})
exten => 5001,103,Hangup
; MeetMe
exten => 200,1,Answer
exten => 200,2,Wait(1)
;exten => 200,3,Authenticate(109)
exten => 200,3,MeetMe(1|Masp)
exten => 200,4,Playback(vm-goodbye)
exten => 200,5,Hangup
[incoming]
exten => 321XXXXXXX,1,Goto(incoming,s,1)
exten => s,1,Answer
exten => s,2,DigitTimeout(10)
exten => s,3,ResponseTimeout(20)
exten => s,4,Background(swelcome)
exten => t,1,Hangup
include =>extensions
[toll-trunks];voicepulse for now
[voicepulse]
;voice over IP outgoing
exten =>
_NXXNXXXXXX,1,Dial(IAX2/usernameREMOVED@gwiaxt01.voicepulse.com/1${EXTEN
})
exten =>
_NXXNXXXXXX,102,Dial(IAX2/usernameREMOVED@gwiaxt01.voicepulse.com/1${EXT
EN})
exten =>
_1NXXNXXXXXX,1,Dial(IAX2/usernameREMOVED@gwiaxt01.voicepulse.com/${EXTEN
})
exten =>
_1NXXNXXXXXX,102,Dial(IAX2/usernameREMOVED@gwiaxt02.voicepulse.com/${EXT
EN})
exten =>
_011.,1,Dial(IAX2/usernameREMOVED@gwiaxt01.voicepulse.com/${EXTEN})
exten =>
_011.,102,Dial(IAX2/usernameREMOVED@gwiaxt01.voicepulse.com/${EXTEN})
;911
;exten => 911,1,ChanIsAvail(Zap/1)
exten => 911,1,Dial(Zap/g1/911)
exten => 911,2,Hangup()
exten => 911,102,SoftHangup (Zap/1-1)
exten => 911,103,Wait(1)
exten => 911,104,Goto(1)
;411
exten => 411,1,Dial(Zap/g1/411)
exten => 411,2,Hangup
[local-trunks]
[local-access]
ignorepat => 9
include =>extensions
include => local-trunks
include => voicepulse
[toll-access]
ignorepat => 9
include => local-access
include => toll-trunks
include => voicepulse
************************************************************************
**
SIP.CONF
[general]
port=5060
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds
to all)
externip=XXX.XXX.XXX.XXX
localnet=192.168.131.0
localmask=255.255.255.0
context=incoming
tos=lowdelay
disallow=all
allow=ulaw
context=invalid
[5001]
type=friend ; either "friend" (peer+user), "peer" or
"user"
host=dynamic
username=5001
context=toll-access
canreinvite=no
quality=300
callerid=<5001>
disallow=all
allow=ulaw
allow=alaw
mailbox=5001@extensions
nat=no
dtmfmode=rfc2833
************************************************************************
ZAPATA.CONF
[channels]
;
; Default language
;
;language=en
;
; Default context
;
context=default
;
; Switchtype: Only used for PRI.
;
; national: National ISDN 2 (default)
; dms100: Nortel DMS100
; 4ess: AT&T 4ESS
; 5ess: Lucent 5ESS
; euroisdn: EuroISDN
; ni1: Old National ISDN 1
;
switchtype=national
;
; Some switches (AT&T especially) require network specific facility IE
; supported values are currently 'none', 'sdn',
'megacom', 'accunet'
;
;nsf=none
;
; PRI Dialplan: Only RARELY used for PRI.
;
; unknown: Unknown
; private: Private ISDN
; local: Local ISDN
; national: National ISDN
; international: International ISDN
;
;pridialplan=national
;
; PRI Local Dialplan: Only RARELY used for PRI (sets the calling
number's numbering plan)
;
; unknown: Unknown
; private: Private ISDN
; local: Local ISDN
; national: National ISDN
; international: International ISDN
;
;prilocaldialplan=national
;
; Overlap dialing mode (sending overlap digits)
;
;overlapdial=yes
;
; PRI Out of band indications.
; Enable this to report Busy and Congestion on a PRI using out-of-band
; notification. Inband indication, as used by Asterisk doesn't seem to
work
; with all telcos.
;
; outofband: Signal Busy/Congestion out of band with
RELEASE/DISCONNECT
; inband: Signal Busy/Congestion using in-band tones
;
; priindication = outofband
;
; ISDN Timers
; All of the ISDN timers and counters that are used are configurable.
Specify
; the timer name, and its value (in ms for timers)
;
; pritimer => t200,1000
; pritimer => t313,4000
;
;
; Signalling method (default is fxs). Valid values:
; em: E & M
; em_w: E & M Wink
; featd: Feature Group D (The fake, Adtran style, DTMF)
; featdmf: Feature Group D (The real thing, MF (domestic, US))
; featb: Feature Group B (MF (domestic, US))
; fxs_ls: FXS (Loop Start)
; fxs_gs: FXS (Ground Start)
; fxs_ks: FXS (Kewl Start)
; fxo_ls: FXO (Loop Start)
; fxo_gs: FXO (Ground Start)
; fxo_ks: FXO (Kewl Start)
; pri_cpe: PRI signalling, CPE side
; pri_net: PRI signalling, Network side
; gr303fxoks_net: GR-303 Signalling, FXO Loopstart, Network side
; gr303fxsks_cpe: GR-303 Signalling, FXS Loopstart, CPE side
; sf: SF (Inband Tone) Signalling
; sf_w: SF Wink
; sf_featd: SF Feature Group D (The fake, Adtran style, DTMF)
; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US))
; sf_featb: SF Feature Group B (MF (domestic, US))
; The following are used for Radio interfaces:
; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the
channel bank)
; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the
channel bank)
; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the
channel bank)
; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at
the channel bank)
; em_rx: Receive audio/COR on an E&M interface (1-way)
; em_tx: Transmit audio/PTT on an E&M interface (1-way)
; em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface
(2-way)
; em_rxtx: same as em_txrx (for our dyslexic friends)
; sf_rx: Receive audio/COR on an SF interface (1-way)
; sf_tx: Transmit audio/PTT on an SF interface (1-way)
; sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface
(2-way)
; sf_rxtx: same as sf_txrx (for our dyslexic friends)
;
signalling=fxo_ls
;
; A variety of timing parameters can be specified as well
; Including:
; prewink: Pre-wink time (default 50ms)
; preflash: Pre-flash time (default 50ms)
; wink: Wink time (default 150ms)
; flash: Flash time (default 750ms)
; start: Start time (default 1500ms)
; rxwink: Receiver wink time (default 300ms)
; rxflash: Receiver flashtime (default 1250ms)
; debounce: Debounce timing (default 600ms)
;
rxwink=300 ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes
;
; Whether or not to use caller ID
;
usecallerid=yes
;
; Type of caller ID signalling in use
; bell = bell202 as used in US, v23 = v23 as used in the UK, dtmf = DTMF
as used in Denmark, Sweden and Netherlands
;
;cidsignalling=bell
;
; What signals the start of caller ID
; ring = a ring signals the start, polarity = polarity reversal signals
the start
;
;cidstart=ring
;
; Whether or not to hide outgoing caller ID (Override with *67 or *82)
;
hidecallerid=no
;
; Whether or not to enable call waiting on FXO lines
;
callwaiting=yes
;
; Whether or not restrict outgoing caller ID (will be sent as ANI only,
not available for the user)
; Mostly use with FXS ports
;
;restrictcid=no
;
; Whether or not use the caller ID presentation for the outgoing call
that the calling switch is sending
;
usecallingpres=yes
;
; Support Caller*ID on Call Waiting
;
callwaitingcallerid=yes
;
; Support three-way calling
;
threewaycalling=yes
;
; Support flash-hook call transfer (requires three way calling)
;
transfer=yes
;
; Support call forward variable
;
cancallforward=yes
;
; Whether or not to support Call Return (*69)
;
callreturn=yes
;
; Stutter dialtone support: If a mailbox is specified without a
voicemail
; context, then when voicemail is received in a mailbox in the default
; voicemail context in voicemail.conf, taking the phone off hook will
; cause a stutter dialtone instead of a normal one.
;
; If a mailbox is specified *with* a voicemail context, the same will
; result if voicemail recieved in mailbox in the specified voicemail
; context
;
; for default voicemail context, the example below is fine:
;
;mailbox=1234
;
; for any other voicemail context, the following will produce the
; stutter tone:
;
;mailbox=1234@context
;
; Enable echo cancellation
; Use either "yes", "no", or a power of two from 32 to 256
if you wish
; to actually set the number of taps of cancellation.
;
echocancel=yes
;
; Generally, it is not necessary (and in fact undesirable) to echo
cancel
; when the circuit path is entirely TDM. You may, however, reverse this
; behavior by enabling the echo cancel during pure TDM bridging below.
;
echocancelwhenbridged=yes
;
; In some cases, the echo canceller doesn't train quickly enough and
there
; is echo at the beginning of the call. Enabling echo training will
cause
; asterisk to briefly mute the channel, send an impulse, and use the
impulse
; response to pre-train the echo canceller so it can start out with a
much
; closer idea of the actual echo. Value may be "yes", "no",
or a number
of
; milliseconds to delay before training (default = 400)
;
;echotraining=yes
;echotraining=800
;
; If you are having trouble with DTMF detection, you can relax the
; DTMF detection parameters. Relaxing them may make the DTMF detector
; more likely to have "talkoff" where DTMF is detected when it
; shouldn't be.
;
;relaxdtmf=yes
;
; You may also set the default receive and transmit gains (in dB)
;
rxgain=0.0
txgain=0.0
;
; Logical groups can be assigned to allow outgoing rollover. Groups
; range from 0 to 31, and multiple groups can be specified.
;
group=1
;
; Ring groups (a.k.a. call groups) and pickup groups. If a phone is
ringing
; and it is a member of a group which is one of your pickup groups, then
; you can answer it by picking up and dialing *8#. For simple offices,
just
; make these both the same
;
callgroup=1
pickupgroup=1
;
; Specify whether the channel should be answered immediately or
; if the simple switch should provide dialtone, read digits, etc.
;
immediate=no
;
; CallerID can be set to "asreceived" or a specific number
; if you want to override it. Note that "asreceived" only
; applies to trunk interfaces.
;
;callerid=2564286000
;
; AMA flags affects the recording of Call Detail Records. If specified
; it may be 'default', 'omit', 'billing', or
'documentation'.
;
;amaflags=default
;
; Channels may be associated with an account code to ease
; billing
;
;accountcode=lss0101
;
; ADSI (Analog Display Services Interface) can be enabled on a
per-channel
; basis if you have (or may have) ADSI compatible CPE equipment
;
;adsi=yes
;
; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group
D
; etc, it can be useful to perform busy detection either in an effort to
; detect hangup or for detecting busies
;
;busydetect=yes
;
; If busydetect is enabled, is also possible to specify how many
; busy tones to wait before hanging up. The default is 4, but
; better results can be achieved if set to 6 or even 8. Mind that
; higher the number, more time is needed to hangup a channel, but
; lower is probability to get random hangups
;
;busycount=4
;
; In some countries, a polarity reversal is used to signal the
disconnect
; of a phone line. If the hanguponpolarityswitch option is selected,
the
; call will be considered "hung up" on a polarity reversal
;
;hanguponpolarityswitch
;
; On trunk interfaces (FXS) it can be useful to attempt to follow the
progress
; of a call through RINGING, BUSY, and ANSWERING. If turned on, call
; progress attempts to determine answer, busy, and ringing on phone
lines.
; This feature is HIGHLY EXPERIMENTAL and can easily detect false
answers,
; so don't count on it being very accurate.
;
; Few zones are supported at the time of this writing, but may
; be selected with "progzone"
;
; This feature can also easily detect false hangups. The symptoms of
this
; is being disconnected in the middle of a call for no reason.
;
;callprogress=yes
;progzone=us
;
; For FXO (FXS signalled) devices, whether to use pulse dial instead of
DTMF
;
;pulsedial=yes
;
; For fax detection, uncomment one of the following lines. The default
is *OFF*
;
;faxdetect=both
;faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
;
; Select which class of music to use for music on hold. If not
specified
; then the default will be used.
;
;musiconhold=default
;
; PRI channels can have an idle extension and a minunused number. So
long
; as at least "minunused" channels are idle, chan_zap will try to call
; "idledial" on them, and then dump them into the PBX in the
"idleext"
; extension (which is of the form exten@context). When channels are
needed
; the "idle" calls are disconnected (so long as there are at least
"minidle"
; calls still running, of course) to make more channels available. The
; primary use of this is to create a dynamic service, where idle
channels
; are bundled through multilink PPP, thus more efficiently utilizing
; combined voice/data services than conventional fixed mappings/muxings.
;
;idledial=6999
;idleext=6999@dialout
;minunused=2
;minidle=1
;
; Configure jitter buffers in zapata (each one is 20ms, default is 4)
;
;jitterbuffers=4
;
; You can define your own custom ring cadences here. You can define up
to
; 8 pairs. If the silence is negative, it indicates where the callerid
; spill is to be placed. Also, if you define any custom cadences, the
; default cadences will be turned off.
;
; Syntax is: cadence=ring,silence[,ring,silence[...]]
;
; These are the default cadences:
;
;cadence=125,125,2000,-4000
;cadence=250,250,500,1000,250,250,500,-4000
;cadence=125,125,125,125,125,-4000
;cadence=1000,500,2500,-5000
;
; Each channel consists of the channel number or range. It
; inherits the parameters that were specified above its declaration
;
; For GR-303, CRV's are created like channels except they must start
; with the trunk group followed by a colon, e.g.:
;
; crv => 1:1
; crv => 2:1-2,5-8
;
;
;callerid="Green Phone"<(256) 428-6121>
;channel => 1
;callerid="Black Phone"<(256) 428-6122>
;channel => 2
;callerid="CallerID Phone" <(256) 428-6123>
;callerid="CallerID Phone" <(630) 372-1564>
;callerid="CallerID Phone" <(256) 704-4666>
;channel => 3
;callerid="Pac Tel Phone" <(256) 428-6124>
;channel => 4
;callerid="Uniden Dead" <(256) 428-6125>
;channel => 5
;callerid="Cortelco 2500" <(256) 428-6126>
;channel => 6
;callerid="Main TA 750" <(256) 428-6127>
;channel => 44
;
; For example, maybe we have some other channels
; which start out in a different context and use
; E & M signalling instead.
;
;context=remote
;sigalling=em
;channel => 15
;channel => 16
;signalling=em_w
;
; All those in group 0 I'll use for outgoing calls
;
; Strip most significant digit (9) before sending
;
;stripmsd=1
;callerid=asreceived
;group=0
;signalling=fxs_ls
;channel => 45
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Noah
Miller
Sent: Friday, December 03, 2004 4:34 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Why, why, why???
> Help.
>
> Why is it that I can call out from my GSBudgetone SIP phone but the
> audio is "one-way'?
>
> Why is it that when I call my asterisk phone number, I get a fast
> busy?
Can we have a looksie at your config files? sip.conf, extensions.conf,
zapata.conf, zaptel.conf.
If you start asterisk with enough verbosity:
asterisk -vvvvvvgc
what does the console say when you make an incoming call to your
asterisk box and get the fast busy?
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