[Asterisk-Users] An example config for using a Wildcard X100P and a SIP phone?

Jason newsgroup at med-web.com
Mon Feb 23 11:01:31 MST 2004


Timothy,

I have minimally modified the demo files that came with Asterisk, so 
what is posted below is most of the comments and the demo section 
removed from the config files.

Thanks!

; SIP Configuration for Asterisk
;
[general]
port = 5060            ; Port to bind to
bindaddr = 0.0.0.0        ; Address to bind to

context = default        ; Default for incoming calls

[sipphone]
type=friend
username=sipphone
fromuser=Sipster                ; Specify user to put in "from" instead 
of callerid
secret=password
host=dynamic
defaultip=192.168.1.201
amaflags=default                ; Choices are default, omit, billing, 
documentation
accountcode=Sipster             ; Users may be associated with an 
accountcode tp ease billing
mailbox=431

--------------------------
extensions.conf
--------------------------
[general]

static=yes

writeprotect=no

[globals]
;CONSOLE=Console/dsp                ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest                    ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/1                    ; Trunk interface
TRUNKMSD=1                    ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:pass at provider

[iaxtel700]
exten => _91700NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)

[trunkint]
;
; International long distance through trunk
;
exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9011.,2,Congestion

[trunkld]
;
; Long distance context accessed through trunk
;
exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91NXXNXXXXXX,2,Congestion

[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9NXXXXXX,2,Congestion

[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91800NXXXXXX,2,Congestion
exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91888NXXXXXX,2,Congestion
exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91877NXXXXXX,2,Congestion
exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91866NXXXXXX,2,Congestion

[international]
;
; Master context for international long distance
;
ignorepat => 9
include => longdistance
include => trunkint

[longdistance]
;
; Master context for long distance
;
ignorepat => 9
include => local
include => trunkld

[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
;include => default
;include => parkedcalls
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider

[macro-stdexten];
;
; Standard extension macro:
;   ${ARG1} - Extension  (we could have used ${MACRO_EXTEN} here as well
;   ${ARG2} - Device(s) to ring
;
exten => s,1,Dial(${ARG2})                    ; Ring the interface, 20 
seconds maximum
exten => s,2,Voicemail(u${ARG1})                ; If unavailable, send 
to voicemail w/ unavail announce
exten => s,3,Goto(default,s,1)                    ; If they press #, 
return to start
exten => s,102,Voicemail(b${ARG1})                ; If busy, send to 
voicemail w/ busy announce
exten => s,103,Goto(default,s,1)                ; If they press #, 
return to start

[default]
;
; By default we include the demo.  In a production system, you
; probably don't want to have the demo there.
;
include => local

exten => 431,1,Dial,SIP/sipphone


Regovich, Timothy wrote:

>Jason,
>
>Include your sip and extensions files so people can take a look.
>
>T
>
>-----Original Message-----
>From: asterisk-users-admin at lists.digium.com
>[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Jason
>Sent: Monday, February 23, 2004 10:25 AM
>To: asterisk-users at lists.digium.com
>Cc: horner at med-web.com
>Subject: [Asterisk-Users] An example config for using a Wildcard X100P and a
>SIP phone?
>
>
>Hello.
>
>I've just recently purchased the Asterisk Developers Kit so we can 
>figure out how to get away from our Nortel system and go to IP based 
>phones. I have a RH 9 box loaded with Asterisk (a very recent cvs download).
>
>Either way, I can call the asterisk box and get their demo playing fine. 
>I can even call the SIP phone I've hooked up when I call in from my cell 
>phone to the asterisk box, and that works.
>
>I cannot call out with my SIP phone though. It'll dial, ring my cell 
>phone twice and then give up and complain that its busy. Even if I try 
>to answer the cell phone during the first ring.
>
>Does anyone have a config they could share with me on how to make this 
>setup work? This sounds like it should be fairly trivial, but I've 
>beaten my head against the wall on this for a few days. =)
>
>Thanks alot,
>Jason
>
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