[Asterisk-Users] An example config for using a Wildcard
X100P and a SIP phone?
Regovich, Timothy
timothy_regovich at merck.com
Mon Feb 23 11:41:25 MST 2004
Try moving your sip phone into its own context, instead of default (I use
"sip") and create a [sip] section in your extensions.conf
Add a sepcific extension to test your outgoing, like :
exten => _5XXXX,1,Dial,Zap/1/800551212
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 1:02 PM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] An example config for using a Wildcard X100P
and a SIP phone?
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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