[Asterisk-Users] (Asterisk-Users] Affordable SIP Phone - Stiil a Myth?
Kanuri, Seshu
seshu.kanuri at citigroup.com
Mon Jul 19 15:19:01 MST 2004
Steve,
Here is the config, I pulled from my server, that works with D'Link Phones:
Main Menu
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SIP.CONF
[general]
port = 5060 ; Port to bind to (SIP is 5060)
;bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
bindaddr = 67.109.153.236
disallow=all
;allow=ilbc
allow=gsm
allow=ulaw
allow=alaw
allow=g729
;allow=g723
jitterbuffer=no
localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
context = bogon-calls ; Send SIP callers that we don't know about here
[2000]
type=friend ; This device takes and makes calls
username=2000 ; Username on device
secret=2000 ; Password for device
host=dynamic ; This host is not on the same IP addr every time
context=from-sip ; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
;dtmfmod=inband ; voicemailbox has messages in it
reinvite=no
canreinvite=no
nat=yes
qualify=4000
callerid=Mr. Mirchandani <2000>
[2001] ; Duplicate of 2000, except with different auth data
type=friend
username=2001
secret=2001
host=dynamic
context=from-sip
mailbox=101
;dtmfmod=inband
nat=yes
reinvite=no
canreinvite=no
callerid=Mr. Mandar <2001>
[2002]
type=friend
host=dynamic
callerid=William Suffill <2002>
username=2002
secret=2002
context=from-sip
nat=yes
mailbox=2002
[2003] ; Duplicate of 2000, except with different auth data
type=friend
username=2003
secret=2003
host=dynamic
context=from-sip
mailbox=103
;dtmfmod=inband
reinvite=no
canreinvite=no
callerid=Mr.Seshu <2003>
[2004] ; Duplicate of 2000, except with different auth data
type=friend
username=2004
;secret=2004
secret=2004
host=dynamic
context=from-sip
mailbox=103
;dtmfmod=inband
reinvite=no
canreinvite=no
nat=yes
callerid=D-link ID1 <2004>
[2005] ; Duplicate of 2000, except with different auth data
type=friend
username=2005
;secret=2005
secret=2005
host=dynamic
context=from-sip
mailbox=104
;dtmfmod=inband
reinvite=no
canreinvite=no
nat=yes
callerid=D-link ID2 <2005>
[2006] ; Duplicate of 2000, except with different auth data
type=friend
username=2006
secret=2006
host=dynamic
context=from-sip
mailbox=105
;dtmfmod=inband
reinvite=no
canreinvite=no
;nat=yes
callerid=Pacenet ID1 <2006>
[2007] ; Duplicate of 2000, except with different auth data
type=friend
username=2007
secret=2007
host=dynamic
context=from-sip
mailbox=106
;dtmfmod=inband
reinvite=no
canreinvite=no
;nat=yes
callerid=Pacenet ID1 <2007>
[2008] ; Duplicate of 2000, except with different auth data
type=friend
username=2008
;secret=2008
secret=2008
host=dynamic
context=from-sip
mailbox=107
;dtmfmod=inband
reinvite=no
canreinvite=no
;nat=yes
callerid=D-link ID3 <2008>
[2009] ; Duplicate of 2000, except with different auth data
type=friend
username=2009
secret=2009
host=dynamic
context=from-sip
mailbox=108
;dtmfmod=inband
reinvite=no
canreinvite=no
;nat=yes
callerid=D-link ID4 <2009>
[2010] ; Duplicate of 2000, except with different auth data
type=friend
username=2010
secret=2010
host=dynamic
context=from-sip
mailbox=109
;dtmfmod=inband
reinvite=no
canreinvite=no
;nat=yes
callerid=USA-ID10<2010>
[2011] ; Duplicate of 2000, except with different auth data
type=friend
username=2011
secret=2011
host=dynamic
context=from-sip
mailbox=110
;dtmfmod=inband
reinvite=no
canreinvite=no
;nat=yes
callerid=USA-ID11<2011>
[3001] ; Duplicate of 3000, except with different auth data
type=friend
username=3001
secret=3001
host=dynamic
context=for-dlink
mailbox=109
;dtmfmod=inband
reinvite=no
canreinvite=no
;nat=yes
callerid=D-link Contact1 <3001>
[3002] ; Duplicate of 3000, except with different auth data
type=friend
username=3002
secret=3002
host=dynamic
context=for-dlink
mailbox=110
;dtmfmod=inband
reinvite=no
canreinvite=no
;nat=yes
callerid=D-link Contact2 <3002>
[3003] ; Duplicate of 3000, except with different auth data
type=friend
username=3003
secret=3003
host=dynamic
context=for-dlink
mailbox=111
;dtmfmod=inband
reinvite=no
canreinvite=no
;nat=yes
callerid=D-link Contact3 <3003>
[3004] ; Duplicate of 3000, except with different auth data
type=friend
username=3004
secret=3004
host=dynamic
context=for-dlink
mailbox=112
;dtmfmod=inband
reinvite=no
canreinvite=no
;nat=yes
callerid=D-link Contact4 <3004>
[4001] ; Duplicate of 3000, except with different auth data
type=friend
username=4001
secret=4001
host=dynamic
context=for-NetWeb
mailbox=109
;dtmfmod=inband
reinvite=no
canreinvite=no
;nat=yes
callerid=NetWeb1 <4001>
[4002] ; Duplicate of 3000, except with different auth data
type=friend
username=4002
secret=4002
host=dynamic
context=for-NetWeb
mailbox=110
;dtmfmod=inband
reinvite=no
canreinvite=no
nat=yes
callerid=NetWeb2 <4002>
[4003] ; Duplicate of 3000, except with different auth data
type=friend
username=4003
secret=4003
host=dynamic
context=for-NetWeb
mailbox=111
;dtmfmod=inband
reinvite=no
canreinvite=no
nat=yes
callerid=NetWeb3 <4003>
[4004] ; Duplicate of 3000, except with different auth data
type=friend
username=4004
secret=4004
host=dynamic
context=for-NetWeb
mailbox=112
;dtmfmod=inband
reinvite=no
canreinvite=no
nat=yes
callerid=NetWeb4 <4004>
[8612312341] ; Duplicate of 2000, except with different auth data
type=friend
username=8612312341
secret=4321
;secret=dlink005
host=dynamic
context=from-sip-1
mailbox=105
;dtmfmod=inband
reinvite=no
canreinvite=no
nat=yes
callerid=Mr. Seshu <8612312341>
[8612312342] ; Duplicate of 2000, except with different auth data
type=friend
username=8612312342
secret=netweb
host=dynamic
context=from-sip-1
mailbox=106
;dtmfmod=inband
reinvite=no
canreinvite=no
nat=yes
callerid=Mr. Mandar <8612312342>
[8612312343] ; Duplicate of 2000, except with different auth data
type=friend
username=8612312343
secret=4321
host=dynamic
context=from-sip-1
mailbox=107
;dtmfmod=inband
reinvite=no
canreinvite=no
nat=yes
callerid=Mr. Gary <8612312343>
[8612312344] ; Duplicate of 2000, except with different auth data
type=friend
username=8612312344
secret=4321
;host=dynamic
context=from-sip-1
mailbox=108
;dtmfmod=inband
reinvite=no
canreinvite=no
nat=yes
callerid=Mr. NoName <8612312344>
[8612312345] ; Duplicate of 2000, except with different auth data
type=friend
username=8612312345
secret=4321
;host=dynamic
context=from-sip-1
mailbox=108
;dtmfmod=inband
reinvite=no
canreinvite=no
nat=yes
callerid=Mr. NoName <8612312345>
[8612312346] ; Duplicate of 2000, except with different auth data
type=friend
username=8612312346
secret=54321
;host=dynamic
context=for-NetWeb
mailbox=108
;dtmfmod=inband
reinvite=no
canreinvite=no
nat=yes
callerid=Mr. AAA <8612312346>
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ZAPATA.CONF
--------------------------------------------------------------------------------
;
; Zapata telephony interface
;
; Configuration file
[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
;
; PRI Dialplan: Only RARELY used for PRI.
;
; unknown: Unknown
; private: Private ISDN
; local: Local ISDN
; national: National ISDN
; international: International ISDN
;
;pridialplan=national
;
; Overlap dialing mode (sending overlap digits)
;
;overlapdial=yes
;
; 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
; 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
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes
;
; Whether or not to use caller ID
;
usecallerid=yes
;
; 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, then when voicemail
; is received in that mailbox, taking the phone off hook will cause
; a stutter dialtone instead of a normal one
;
;mailbox=1234
;
; 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.
;
;echotraining=yes
;
; 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
;
; 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. Also, it is ONLY configured for
; standard U.S. tones. 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
;
; 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 at 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 at dialout
;minunused=2
;minidle=1
;
; Configure jitter buffers in zapata (each one is 20ms, default is 4)
;
;jitterbuffers=4
;
; Each channel consists of the channel number or range. It
; inherits the parameters that were specified above its declaration
;
;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
;signalling=fxo_ls
;group=1
;callerid="Joe Schmoe" <(256) 428-6131>
;channel => 25
;callerid="Megan May" <(256) 428-6132>
;channel => 26
;callerid="Suzy Queue" <(256) 428-6233>
;channel => 27
;callerid="Larry Moe" <(256) 428-6234>
;channel => 28
;
; Sample PRI (CPE) config: Specify the switchtype, the signalling as
; either pri_cpe or pri_net for CPE or Network termination, and generally
; you will want to create a single "group" for all channels of the PRI.
;
; switchtype = national
; signalling = pri_cpe
; group = 2
; channel => 1-23
;
; Used for distintive ring support for x100p.
; You can see the dringX patterns is to set any one of the dringXcontext fields
; and they will be printed on the console when an inbound call comes in.
;
;dring1=95,0,0
;dring1context=internal1
;dring2=325,95,0
;dring2context=internal2
; If no pattern is matched here is where we go.
;context=default
;channel => 1
--------------------------------------------------------------------------------
EXTENSIONS.CONF
--------------------------------------------------------------------------------
[general]
static=yes ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.
VM=216.162.116.46; ip address of VOICE master
[bogon-calls]
;
; Take unknown callers that may have found
; our system, and send them to a re-order tone.
; The string "_." matches any dialed sequence, so all
; calls will result in the Congestion tone application
; being called. They'll get bored and hang up eventually.
;
exten => _.,1,Congestion
[from-sip]
;
; If the number dialed by the calling party was "2000", then
; Dial the user "2000" via the SIP channel driver. Let the number
; ring for 20 seconds, and if no answer, proceed to priority 2.
; If the number gives a "busy" result, then jump to priority 102
;
exten => 2000,1,Dial(SIP/2000,20)
;
; Priority 2 send the caller to voicemail, and gives the "u"navailable
; message for user 2000, as recorded previously. The only way out
; of voicemail in this instance is to hang up, so we have reached
; the end of our priority list.
;
exten => 2000,2,Voicemail(u2000)
;
; If the Dialed number in priority 1 above results in
; a "busy" code, then Dial will jump to 101 + (current priority)
; which in our case will be 101+1=102. This +101 jump is built
; into Asterisk and does not need to be defined.
;
exten => 2000,102,Voicemail(b2000)
exten => 2000,103,Hangup
;
; Now, what if the number dialed was "2001"?
;
exten => 2001,1,Dial(SIP/2001,20)
exten => 2001,2,Voicemail(u2001)
exten => 2001,102,Voicemail(b2001)
exten => 2001,103,Hangup
;New one
exten => 2002,1,Dial(SIP/2002,20)
exten => 2002,2,Voicemail(u2002)
exten => 2002,102,Voicemail(b2002)
exten => 2002,103,Hangup
;New one
exten => 2003,1,Dial(SIP/2003,20)
exten => 2003,2,Voicemail(u2003)
exten => 2003,102,Voicemail(b2003)
exten => 2003,103,Hangup
;New one
exten => 2004,1,Dial(SIP/2004,20)
exten => 2004,2,Voicemail(u2004)
exten => 2004,102,Voicemail(b2004)
exten => 2004,103,Hangup
;New one
exten => 2005,1,Dial(SIP/2005,20)
exten => 2005,2,Voicemail(u2005)
exten => 2005,102,Voicemail(b2005)
exten => 2005,103,Hangup
;New one
exten => 2006,1,Dial(SIP/2006,20)
exten => 2006,2,Voicemail(u2006)
exten => 2006,102,Voicemail(b2006)
exten => 2006,103,Hangup
;New one
exten => 2007,1,Dial(SIP/2007,20)
exten => 2007,2,Voicemail(u2007)
exten => 2007,102,Voicemail(b2007)
exten => 2007,103,Hangup
;New one
exten => 2008,1,Dial(SIP/2008,20)
exten => 2008,2,Voicemail(u2008)
exten => 2008,102,Voicemail(b2008)
exten => 2008,103,Hangup
;New one
exten => 2009,1,Dial(SIP/2009,20)
exten => 2009,2,Voicemail(u2009)
exten => 2009,102,Voicemail(b2009)
exten => 2009,103,Hangup
;New one
exten => 2010,1,Dial(SIP/2010,20)
exten => 2010,2,Voicemail(u2010)
exten => 2010,102,Voicemail(b2010)
exten => 2010,103,Hangup
;New one
exten => 2011,1,Dial(SIP/2011,20)
exten => 2011,2,Voicemail(u2011)
exten => 2011,102,Voicemail(b2011)
exten => 2011,103,Hangup
;
; Define a way so that users can dial a number to reach
; voicemail. Call the VoicemailMain application with the
; number of the caller already passed as a variable, so
; all the user needs to do is type in the password.
;
exten => 2999,1,VoicemailMain(${CALLERIDNUM})
include => from-sip-1
[for-dlink]
;New one
exten => 3001,1,Dial(SIP/3001,20)
exten => 3001,2,Voicemail(u3001)
exten => 3001,102,Voicemail(b3001)
exten => 3001,103,Hangup
;New one
exten => 3002,1,Dial(SIP/3002,20)
exten => 3002,2,Voicemail(u3002)
exten => 3002,102,Voicemail(b3002)
exten => 3002,103,Hangup
;New one
exten => 3003,1,Dial(SIP/3003,20)
exten => 3003,2,Voicemail(u3003)
exten => 3003,102,Voicemail(b3003)
exten => 3003,103,Hangup
;New one
exten => 3004,1,Dial(SIP/3004,20)
exten => 3004,2,Voicemail(u3004)
exten => 3004,102,Voicemail(b3004)
exten => 3004,103,Hangup
exten => 3999,1,VoicemailMain(${CALLERIDNUM})
[for-NetWeb]
;New one
exten => 4001,1,Dial(SIP/4001,20)
exten => 4001,2,Voicemail(u4001)
exten => 4001,102,Voicemail(b4001)
exten => 4001,103,Hangup
;New one
exten => 4002,1,Dial(SIP/4002,20)
exten => 4002,2,Voicemail(u4002)
exten => 4002,102,Voicemail(b4002)
exten => 4002,103,Hangup
;New one
exten => 4003,1,Dial(SIP/4003,20)
exten => 4003,2,Voicemail(u4003)
exten => 4003,102,Voicemail(b4003)
exten => 4003,103,Hangup
;New one
exten => 4004,1,Dial(SIP/4004,20)
exten => 4004,2,Voicemail(u4004)
exten => 4004,102,Voicemail(b4004)
exten => 4004,103,Hangup
;New one
exten => 8612312346,1,Dial(SIP/8612312346,20)
exten => 8612312346,2,Voicemail(u8612312346)
exten => 8612312346,102,Voicemail(b8612312346)
exten => 8612312346,103,Hangup
exten => 4999,1,VoicemailMain(${CALLERIDNUM})
[from-sip-1]
;New one
exten => 8612312341,1,Dial(SIP/8612312341,20)
exten => 8612312341,2,Voicemail(u8612312341)
exten => 8612312341,102,Voicemail(b8612312341)
exten => 8612312341,103,Hangup
;New one
exten => 8612312342,1,Dial(SIP/8612312342,20)
exten => 8612312342,2,Voicemail(u8612312342)
exten => 8612312342,102,Voicemail(b8612312342)
exten => 8612312342,103,Hangup
;New one
exten => 8612312343,1,Dial(SIP/8612312343,20)
exten => 8612312343,2,Voicemail(u8612312343)
exten => 8612312343,102,Voicemail(b8612312343)
exten => 8612312343,103,Hangup
;New one
exten => 8612312344,1,Dial(SIP/17323874133 at 216.162.116.46,20)
exten => 8612312344,2,Voicemail(u8612312344)
exten => 8612312344,102,Voicemail(b8612312344)
exten => 8612312344,103,Hangup
;New one
exten => 8612312345,1,MeetMe,1234
exten => 8612312345,2,Voicemail(u8612312345)
exten => 8612312345,102,Voicemail(b8612312345)
exten => 8612312345,103,Hangup
include => from-sip
;exten => 8612312349,1,VoicemailMain(${CALLERIDNUM})
;route calls from h323 protocaol to sip to gatgekeeper
[h323]
exten => _.,1,dial(SIP/BYEXTENSION at 216.162.116.46:5060)
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