No loop now, but instead I get this:

Mar 26 15:42:18 NOTICE[1854] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Mar 26 15:42:18 VERBOSE[1854] logger.c: == Everyone is busy/congested at this time (1:0/0/1)
Mar 26 15:42:18 DEBUG[1854] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL.


dave cantera wrote:

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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