[Asterisk-Users] Repost: Cisco 7960 and Asterisk...not working....

[Amer Nasir]

The small "x" means that the phone is not registered. Make sure you
have the correct secret. Look at asterisk cli debug output. sip debug.
Look for the response from asterisk. is it a 401 or 404?
You can also try setting host=dynamic.
and before all this check the basic networking stuff.



On Tue, 14 Dec 2004 00:15:09 -0000, Paul A Brown <paul at fowlmere.com> wrote:
>  
>   
> Anyone help me here? I am a newbie so be gentle ;-)...... 
>   
> It worked once and then I played with the configs. 
>   
> I have a static IP address which is on my private network.. Phone is
> 192.192.192.220 and asterisk server is 192.192.192.22 
>   
> I have the 7690 with a SIP iamge (Whatever latest is ) 
>   
> I have 3 lines setup with Free World Dial up and have the 4th setup to
> connect to my asterisk server. Here are my config files......It worked once
> but now the phone sits there with a 'x' next to it :-( 
>   
> ;
> ; SIP Configuration for Asterisk
> ;
> ; Syntax for specifying a SIP device in extensions.conf is
> ; SIP/devicename where devicename is defined in a section below.
> ;
> ; You may also use 
> ; SIP/username at domain to call any SIP user on the Internet
> ; (Don't forget to enable DNS SRV records if you want to use this)
> ; 
> ; If you define a SIP proxy as a peer below, you may call
> ; SIP/proxyhostname/user or SIP/user at proxyhostname 
> ; where the proxyhostname is defined in a section below 
> ; 
> ; Useful CLI commands to check peers/users:
> ;   sip show peers  Show all SIP peers (including friends)
> ;   sip show users  Show all SIP users (including friends)
> ;   sip show registry  Show status of hosts we register with
> ;
> ;   sip debug   Show all SIP messages
> ; 
>   
> [general]
> context=home   ; Default context for incoming calls 
>   
> port=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 
>   
> ;[sip_proxy]
> ; For incoming calls only. Example: FWD (Free World Dialup)
> ;type=user
> ;context=from-fwd 
>   
> ;[sip_proxy-out]
> ;type=peer            ; we only want to call out, not be called
> ;secret=guessit
> ;username=yourusername  ; Authentication user for outbound proxies
> ;fromuser=yourusername  ; Many SIP providers require this!
> ;host=box.provider.com
> ;------------------------------------------------
> ; Test Ext 2201 
> ; <extension use> - <users name> - <extension number>
> ;------------------------------------------------ 
>   
>  [2201]
>  type=friend
>  host=192.192.192.220
>  context=home
>  secret=xxxxxx
>  callerid="Paul" <2201>
>  mailbox=2201
>  dtmfmode=rfc2833
>  nat=no 
>   
> EXTENSIONS.CONF 
>   
> writeprotect=no 
>   
> [globals]
>  PHONES1=SIP/2201
>  PHONES1VM=2201
>  PHONES2=SIP/2202
>  PHONES2VM=2202
> CONSOLE=Console/dsp    ; Console interface for demo
> ;CONSOLE=Zap/1
> ;CONSOLE=Phone/phone0
> IAXINFO=guest     ; IAXtel username/password
> ;IAXINFO=myuser:mypass
> TRUNK=Zap/g2     ; Trunk interface
> TRUNKMSD=1     ; MSD digits to strip (usually 1 or 0)
>  
> [iaxtel700]
> exten => _91700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel) 
>   
> [iaxprovider]
> ;switch => IAX2/user:[key]@myserver/mycontext 
>   
> [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
>  
> ;This will create a macro we will use in the dialling plan
>  [macro-vmessage]
>  exten => s,1,VoiceMail2(u${ARG1})
>  exten => s,2,Playback(groovy)
>  exten => s,3,Playback(goodbye)
>  exten => s,4,Hangup 
>   
> [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 
>   
>   
>   
> ; ----------------------------------------------
> ; DEFINE EXTENSIONS
> ; ---------------------------------------------- 
>   
>  [home]
>  ; Next, add an extension for voicemail .
>  ; now if we dial 8, we can check voicemail.
>  ;
>  exten => 8,1,VoiceMailMain2
>  exten => 8,2,Hangup
>  ; Add some more extensions for the two lines . now we'll be able to call
> one line from the other.
>  ; And if no one answers, it will go to the mailbox for that line.
>  ;
>  ; Line 1
>  ;
>  exten => 2201,1,Dial(${PHONES1},20,Ttm)
>  exten => 2201,2,Macro(vmessage,${PHONES1VM})
>  exten => 2201,3,Hangup
>  ;
>  ; Line 2
>  ;
>  exten => 2202,1,Dial(${PHONES2},20,Ttm)
>  exten => 2202,2,Macro(vmessage,${PHONES2VM})
>  exten => 2202,3,Hangup
>  ;
>  ; Line 3
>  ;
>  exten => 2203,1,Dial(${PHONES3},20,Ttm)
>  exten => 2203,2,Macro(vmessage,${PHONES3VM})
>  exten => 2203,3,Hangup 
>   
> ; ----------------------------------------------
> ; END DEFINE EXTENSIONS
> ; ---------------------------------------------- 
>   
> 
> [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,2,Answer   ; Answer the line
> exten => s,3,DigitTimeout,5  ; Set Digit Timeout to 5 seconds
> exten => s,4,ResponseTimeout,10  ; Set Response Timeout to 10 seconds
> exten => s,5,BackGround(demo-congrats) ; Play a congratulatory message
> exten => s,6,BackGround(demo-instruct) ; Play some instructions 
>   
> exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
> exten => 2,2,Goto(s,6) 
>   
> exten => 3,1,SetLanguage(fr)  ; Set language to french
> exten => 3,2,Goto(s,5)   ; 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 => 1234,2,Macro(stdexten,1234,${CONSOLE}) 
>   
> exten => 1235,1,Voicemail(u1234)  ; Right to voicemail 
>   
> exten => 1236,1,Dial(Console/dsp)  ; Ring forever
> exten => 1236,2,Voicemail(u1234)  ; Unless busy 
>   
> ;
> ; # for when they're done with the demo
> ;
> exten => #,1,Playback(demo-thanks)  ; "Thanks for trying the demo"
> exten => #,2,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 => 500,2,Dial(IAX2/guest at misery.digium.com/s at default) ; Call the
> Asterisk demo
> exten => 500,3,Playback(demo-nogo) ; Couldn't connect to the demo site
> exten => 500,4,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 => 600,2,Echo   ; Do the echo test
> exten => 600,3,Playback(demo-echodone) ; Let them know it's over
> exten => 600,4,Goto(s,6)  ; Start over 
>   
> ;
> ; Give voicemail at extension 8500
> ;
> exten => 8500,1,VoicemailMain
> exten => 8500,2,Goto(s,6)
> ;
> ; Here's what a phone entry would look like (IXJ for example)
> ;
> ;exten => 1265,1,Dial(Phone/phone0,15)
> ;exten => 1265,2,Goto(s,5) 
>   
> ;[mainmenu]
> ;
> ; Example "main menu" context with submenu
> ;
> ;exten => s,1,Answer
> ;exten => s,2,Background(thanks)  ; "Thanks for calling press 1 for sales, 2
> for support, ..."
> ;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,2,Wait,2
> ;exten => s,3,Background(submenuopts) ; "Thanks for calling the sales
> department.  Press 1 for steve, 2 for..."
> ;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@${EXTEN:2}@otherprovider.net,30,rT) 
>   
> ; Real extensions would go here. Generally you want real extensions to be 4
> or 5
> ; digits long (although there is no such requirement) and start with a
> single
> ; digit that is fairly large (like 6 or 7) so that you have plenty of room
> to
> ; overlap extensions and menu options without conflict.  You can alias them
> with
> ; names, too and use global variables 
>   
> ;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1 ; Channel hints for presence
> ;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer
> ;exten => 6245,1,Dial(${HINT},20,rtT)  ; Use hint as listed
> ;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 => 8500,2,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. 
> ; 
> 
>  
>  
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