[Asterisk-Users] Problems with registration

Joshua Abbott jabbott at successfulhosting.com
Sat Aug 27 15:19:28 MST 2005


My phone still says Not-Registered. I have a Polycom SoundPoint 600 SIP 
phone.
Here is my sip.conf file:

;
; SIP Configuration
;

[general]
context=default            ; Default context for incoming calls
port=5060 ;added
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

; START FWD1 ACCT
register=691657:ImaCowsertjoshu at fwd.pulver.com/691657
 [fwd]
 type=friend
 secret=********
 username=691657
 host=fwd.pulver.com
 dtmfmode=inband
 context=home
 nat=yes
 canreinvite=no
 disallow=all
 allow=all
; END FWD1 ACCT

[7890]
type=friend
host=192.168.2.29
context=home
secret=********
callerid="OFFICE PHONE #2" <7890>
mailbox=7890
dtmfmode=rfc2833
nat=0

AND HERE IS MY EXTENSIONS.CONF FILE

[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[globals]
CONSOLE=Console/dsp
PHONES1=SIP/7890 ; Phone 1 Def
PHONES1VM=7890   ; Phone 1 VM Def
FWDUSERID1=691657
MYNAME1=My name
MYPHONE1=691657
TRUNKMSD=1                    ; MSD digits to strip (usually 1 or 0)

[fwd-forced-fwd1]
; Check to see if the called number starts with a "7" and
 ; if so, set the call parameters and bounce the call to the
 ; Free World Dialup SIP server.
 ;
 ; NOTE: Calls to unknown users will result in "invalid extension"
 ; message being played.
 ;
 exten => _7.,1,SetCallerID(${FWDUSERID1})
 exten => _7.,2,SetCIDName(${MYNAME1})
 exten => _7.,3,Dial(SIP/${EXTEN:1}@fwd)
 exten => _7.,4,Playback(invalid)
 exten => _7.,5,Hangup

[from-sip-fwd1]
exten => ${FWDUSERID1},1,Dial(${PHONES1},30,Ttm)
 exten => ${FWDUSERID1},2,Voicemail2(u${PHONES1VM})
 exten => ${FWDUSERID1},3,Hangup


[dundi-e164-canonical]
;
; List canonical entries here
;
;exten => 12564286000,1,Macro(std-exten,6000,IAX2/foo)
;exten => _125642860XX,1,Dial(IAX2/otherbox/${EXTEN:7})

[dundi-e164-customers]
;
; If you are an ITSP or Reseller, list your customers here.
;
;exten => _12564286000,1,Dial(SIP/customer1)
;exten => _12564286001,1,Dial(IAX2/customer2)

[dundi-e164-via-pstn]
;
; If you are freely delivering calls to the PSTN, list them here
;
;exten => _1256428XXXX,1,Dial(Zap/g2/${EXTEN:7}) ; Expose all of 256-428
;exten => _1256325XXXX,1,Dial(Zap/g2/${EXTEN:7}) ; Ditto for 256-325

[dundi-e164-local]
;
; Context to put your dundi IAX2 or SIP user in for
; full access
;
include => dundi-e164-canonical
include => dundi-e164-customers
include => dundi-e164-via-pstn

[dundi-e164-switch]
;
; Just a wrapper for the switch
;
switch => DUNDi/e164

[dundi-e164-lookup]
;
; Locally to lookup, try looking for a local E.164 solution
; then try DUNDi if we don't have one.
;
include => dundi-e164-local
include => dundi-e164-switch
;
; DUNDi can also be implemented as a Macro instead of using
; the Local channel driver.
;
[macro-dundi-e164]
;
; ARG1 is the extension to Dial
;
exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup

;
; Here are the entries you need to participate in the IAXTEL
; call routing system.  Most IAXTEL numbers begin with 1-700, but
; there are exceptions.  For more information, and to sign
; up, please go to www.gnophone.com or www.iaxtel.com
;
[iaxtel700]
exten => _91700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)

;
; The SWITCH statement permits a server to share the dialplain with
; another server. Use with care: Reciprocal switch statements are not
; allowed (e.g. both A -> B and B -> A), and the switched server needs
; to be on-line or else dialing can be severly delayed.
;
[iaxprovider]
;switch => IAX2/user:[key]@myserver/mycontext

[trunkint]
;
; International long distance through trunk
;
exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten => _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

[trunkld]
;
; Long distance context accessed through trunk
;
exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
exten => _91NXXNXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

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

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

[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
include => home

;
; You can use an alternative switch type as well, to resolve
; extensions that are not known here, for example with remote
; IAX switching you transparently get access to the remote
; Asterisk PBX
;
; switch => IAX2/user:password at bigserver/local
;
; An "lswitch" is like a switch but is literal, in that
; variable substitution is not performed at load time
; but is passed to the switch directly (presumably to
; be substituted in the switch routine itself)
;
; lswitch => Loopback/12${EXTEN}@othercontext
;
; An "eswitch" is like a switch but the evaluation of
; variable substitution is performed at runtime before
; being passed to the switch routine.
;
; eswitch => IAX2/context@${CURSERVER}

[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},20)                    ; Ring the interface, 
20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1)                ; Jump based on 
status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s-NOANSWER,1,Voicemail(u${ARG1})        ; If unavailable, send 
to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1)            ; If they press #, 
return to start

exten => s-BUSY,1,Voicemail(b${ARG1})            ; If busy, send to 
voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1)                ; If they press #, 
return to start

exten => _s-.,1,Goto(s-NOANSWER,1)                ; Treat anything else 
as no answer

exten => a,1,VoicemailMain(${ARG1})                ; If they press *, 
send the user into VoicemailMain

[macro-stdPrivacyexten];
;
; Standard extension macro:
;   ${ARG1} - Extension  (we could have used ${MACRO_EXTEN} here as well
;   ${ARG2} - Device(s) to ring
;   ${ARG3} - Optional DONTCALL context name to jump to (assumes the s,1 
extension-priority)
;   ${ARG4} - Optional TORTURE context name to jump to (assumes the s,1 
extension-priority)`
;
exten => s,1,Dial(${ARG2},20|p)                    ; Ring the interface, 
20 seconds maximum, call screening option (or use P for databased call 
screening)
exten => s,2,Goto(s-${DIALSTATUS},1)                ; Jump based on 
status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s-NOANSWER,1,Voicemail(u${ARG1})        ; If unavailable, send 
to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1)            ; If they press #, 
return to start

exten => s-BUSY,1,Voicemail(b${ARG1})            ; If busy, send to 
voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1)                ; If they press #, 
return to start

exten => s-DONTCALL,1,Goto(${ARG3},s,1)               ; Callee chose to 
send this call to a polite "Don't call again" script.

exten => s-TORTURE,1,Goto(${ARG4},s,1)                ; Callee chose to 
send this call to a telemarketer torture script.

exten => _s-.,1,Goto(s-NOANSWER,1)                ; Treat anything else 
as no answer

exten => a,1,VoicemailMain(${ARG1})                ; If they press *, 
send the user into VoicemailMain


[macro-vmessage] ;This will create a macro we will use in the dialling plan
exten => s,1,VoiceMail2(u${ARG1})
exten => s,2,Playback(groovy)
exten => s,3,Playback(goodbye)
exten => s,4,Hangup

[demo]
;
; We start with what to do when a call first comes in.
;
exten => s,1,Wait,1            ; Wait a second, just for fun
exten => s,n,Answer            ; Answer the line
exten => s,n,Set(TIMEOUT(digit)=5)    ; Set Digit Timeout to 5 seconds
exten => s,n,Set(TIMEOUT(response)=10)    ; Set Response Timeout to 10 
seconds
exten => s,n(restart),BackGround(demo-congrats)    ; Play a 
congratulatory message
exten => s,n(instruct),BackGround(demo-instruct)    ; Play some instructions
exten => s,n,WaitExten        ; Wait for an extension to be dialed.

exten => 2,1,BackGround(demo-moreinfo)    ; Give some more information.
exten => 2,n,Goto(s,instruct)

exten => 3,1,Set(LANGUAGE()=fr)        ; Set language to french
exten => 3,n,Goto(s,restart)            ; Start with the congratulations

exten => 1000,1,Goto(default,s,1)
;
; We also create an example user, 1234, who is on the console and has
; voicemail, etc.
;
exten => 1234,1,Playback(transfer,skip)        ; "Please hold while..."
                    ; (but skip if channel is not up)
exten => 1234,n,Macro(stdexten,1234,${CONSOLE})

exten => 1235,1,Voicemail(u1234)        ; Right to voicemail

exten => 1236,1,Dial(Console/dsp)        ; Ring forever
exten => 1236,n,Voicemail(u1234)        ; Unless busy

;
; # for when they're done with the demo
;
exten => #,1,Playback(demo-thanks)        ; "Thanks for trying the demo"
exten => #,n,Hangup            ; Hang them up.

;
; A timeout and "invalid extension rule"
;
exten => t,1,Goto(#,1)            ; If they take too long, give up
exten => i,1,Playback(invalid)        ; "That's not valid, try again"

;
; Create an extension, 500, for dialing the
; Asterisk demo.
;
exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,n,Dial(IAX2/guest at misery.digium.com/s at default)    ; Call 
the Asterisk demo
exten => 500,n,Playback(demo-nogo)    ; Couldn't connect to the demo site
exten => 500,n,Goto(s,6)        ; Return to the start over message.

;
; Create an extension, 600, for evaulating echo latency.
;
exten => 600,1,Playback(demo-echotest)    ; Let them know what's going on
exten => 600,n,Echo            ; Do the echo test
exten => 600,n,Playback(demo-echodone)    ; Let them know it's over
exten => 600,n,Goto(s,6)        ; Start over

;
; Give voicemail at extension 8500
;
exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)
;
; Here's what a phone entry would look like (IXJ for example)
;
;exten => 1265,1,Dial(Phone/phone0,15)
;exten => 1265,n,Goto(s,5)

;[mainmenu]
;
; Example "main menu" context with submenu
;
;exten => s,1,Answer
;exten => s,n,Background(thanks)        ; "Thanks for calling press 1 
for sales, 2 for support, ..."
;exten => s,n,WaitExten
;exten => 1,1,Goto(submenu,s,1)
;exten => 2,1,Hangup
;include => default
;
;[submenu]
;exten => s,1,Ringing                    ; Make them comfortable with 2 
seconds of ringback
;exten => s,n,Wait,2
;exten => s,n,Background(submenuopts)    ; "Thanks for calling the sales 
department.  Press 1 for steve, 2 for..."
;exten => s,n,WaitExten
;exten => 1,1,Goto(default,steve,1)
;exten => 2,1,Goto(default,mark,2)

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

;
; Extensions like the two below can be used for FWD, Nikotel, sipgate etc.
; Note that you must have a [sipprovider] section in sip.conf whereas
; the otherprovider.net example does not require such a peer definition
;
;exten => _41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,,r)
;exten => _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT)

; Real extensions would go here. Generally you want real extensions to 
be 4 or 5
; digits long (although there is no such requirement) and start with a 
single
; digit that is fairly large (like 6 or 7) so that you have plenty of 
room to
; overlap extensions and menu options without conflict.  You can alias 
them with
; names, too and use global variables

;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1,Joe Schmoe ; Channel 
hints for presence
;exten => 6245,1,Dial(SIP/Grandstream1,20,rt)    ; permit transfer
;exten => 6245,n(dial),Dial(${HINT},20,rtT)        ; Use hint as listed
;exten => 6245,n,Voicemail(u6245)        ; Voicemail (unavailable)
;exten => 6245,s+1,Hangup            ; s+1, same as n
;exten => 6245,dial+101,Voicemail(b6245)    ; Voicemail (busy)
;exten => 6361,1,Dial(IAX2/JaneDoe,,rm)        ; ring without time limit
;exten => 6389,1,Dial(MGCP/aaln/1 at 192.168.0.14)
;exten => 6394,1,Dial(Local/6275/n)        ; this will dial ${MARK}

;exten => 6275,1,Macro(stdexten,6275,${MARK})    ; assuming ${MARK} is 
something like Zap/2
;exten => mark,1,Goto(6275|1)            ; alias mark to 6275
;exten => 6536,1,Macro(stdexten,6236,${WIL})    ; Ditto for wil
;exten => wil,1,Goto(6236|1)
;
; Some other handy things are an extension for checking voicemail via
; voicemailmain
;
;exten => 8500,1,VoicemailMain
;exten => 8500,n,Hangup
;
; Or a conference room (you'll need to edit meetme.conf to enable this room)
;
;exten => 8600,1,Meetme(1234)
;
; Or playing an announcement to the called party, as soon it answers
;
;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
;
; For more information on applications, just type "show applications" at 
your
; friendly Asterisk CLI prompt.
;
; 'show application <command>' will show details of how you
; use that particular application in this file, the dial plan.
;

[dialout-fwd1]
include => fwd-forced-fwd1
include => from-sip-fwd1


; --------------------------
;     DEFINE EXTENSIONS
; --------------------------

[home]
include => dialout-fwd1
; Next, add an extension for voicemail.
; now if we dial 8, we can check voicemail.
;
exten => 8,1,VoiceMailMain2
exten => 8,2,Hangup
;
; Line 1
;
exten => 7890,1,Dial(${PHONES1},20,Ttm)
exten => 7890,2,Macro(vmessage,${PHONES1VM})
exten => 7890,3,Hangup


; --------------------------
;    END DEFINE EXTENSIONS
; --------------------------




More information about the asterisk-users mailing list