nathan,
try dial() directly to the extension
[to-sip]
exten => _X.,1,Dial(SIP/[EMAIL PROTECTED],120)
try
exten => _X.,1,Dial(SIP/${EXTEN},20)
where ${EXTEN} = 201
and
[201] in /etc/sip.conf is
[201]
type=friend ; Friends place calls and receive calls
context=from-sip ; Context for incoming calls from this user
in extensions.conf
[from-sip]
exten => 201,1,Wait(1)
exten => 201,n,Answer()
exten => 201,n,Dial(SIP/201,15)
exten => 201,n,VoiceMailMain
exten => 201,n,Hangup()
Nathan Bell wrote:
Sorry, forgot to attach the sip.conf and extensions.conf files.
Attached now.
------------------------------------------------------------------------
[general]
context=from-sip ; Default context for incoming calls
; if asterisk was compiled with OSP support.
realm=actarg.com ; Realm for digest authentication
; defaults to "asterisk"
; Realms MUST be globally unique according to
RFC 3261
; Set this to your host name or domain name
bindport=5060 ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
autodomain=yes ; Turn this on to have Asterisk add local host
; name and local IP to domain list.
; and multiline formatted headers for strict
qualify=yes
disallow=all
allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
progressinband=no ; Polycom phones don't work properly with "never"
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
nat=no ; there is not NAT between phone and Asterisk
canreinvite=no ; disallow RTP voice traffic to bypass Asterisk
[201]
type=friend ; Friends place calls and receive calls
context=from-sip ; Context for incoming calls from this user
secret=asteriskpassword
host=dynamic ; This peer register with us
callerid=John Doe <201>
[202]
type=friend ; Friends place calls and receive calls
context=from-sip ; Context for incoming calls from this user
secret=asteriskpassword
host=dynamic ; This peer register with us
callerid=Jane Doe <202>
------------------------------------------------------------------------
; from outside T1
[from-ptsn]
exten => s,1,Answer()
include => cac-ext
include => sip-ext
include => intertel-ext
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup()
; from sip lines
[from-sip]
include => internal
; generic interal route
[internal]
exten => s,1,Answer()
include => cac-ext
include => sip-ext
include => intertel-ext
include => to-ptsn
; check if extension is to sip
[sip-ext]
exten => _20X,1,Goto(to-sip,${EXTEN},1)
; send call to sip
[to-sip]
exten => _X.,1,Dial(SIP/[EMAIL PROTECTED],120)
exten => _X.,2,Playback(vm-nobodyavail)
exten => _X.,3,Hangup()
exten => _X.,102,Playback(tt-allbusy)
exten => _X.,103,Hangup()
------------------------------------------------------------------------
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