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:[EMAIL PROTECTED]
[iaxtel700]
exten => _91700NXXXXXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])[trunkint]
;
; International long distance through trunk
;
exten => _9011.,1,Dial(${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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Sent: Monday, February 23, 2004 10:25 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] 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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