No Incoming in Asterisk... Μπορεί να βοηθήσει κανείς??

[chris25873]

Παλεύω από χτές μια εγκατάσταση Asterisk και σε μεγάλο βαθμό τα έχω καταφέρει (ας είναι καλά το Asterisk-GUI !) .
Έχω κολλήσει ομως στις εισερχόμενες κλήσεις.
Ενώ έχω κάνει κανονικά register 2 providers (evoice & voipdiscount) και κάνω κλήσεις προς τα έξω, δεν μπορώ να δεχτω κλήσεις στο evoice. Καταλαβαίνω ότι κάτι δεν έχω ρυθμίσει στο extensions.conf αλλά επειδή έχει ρυθμιστεί με macros δεν καταλαβαίνω τι πάει στραβά.

Trunk 1 -> evoice
Trunk 2 -> voipdiscount
Παραθέτω και το μήνυμα λάθους που βγάζει το asterisk

[Apr 30 12:31:01] WARNING[22820]: chan_sip.c:8385 check_auth: username mismatch, have <trunk_1>, digest has <s>
[Apr 30 12:31:01] NOTICE[22820]: chan_sip.c:13827 handle_request_invite: Failed to authenticate user "30210961xxxx(δικη μου διορθωση)" <sip:210961χχχχ@194.30.193.119>;tag=as01767162

Παραθέτω το αρχείο extensions.conf μήπως κανείς μπορεί να βοηθήσει....

;!
;! Automatically generated configuration file
;! Filename: extensions.conf (/etc/asterisk/extensions.conf)
;! Generator: Manager
;! Creation Date: Wed Apr 30 10:39:16 2008
;!
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified.  Remember that all comments
; made in the file will be lost when that happens. 
;
; XXX Not yet implemented XXX
;
static = yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command "dialplan save" too
;
writeprotect = no
;
; If autofallthrough is set, then if an extension runs out of
; things to do, it will terminate the call with BUSY, CONGESTION
; or HANGUP depending on Asterisk's best guess. This is the default.
;
; If autofallthrough is not set, then if an extension runs out of 
; things to do, Asterisk will wait for a new extension to be dialed 
; (this is the original behavior of Asterisk 1.0 and earlier).
;
;autofallthrough=no
;
; If clearglobalvars is set, global variables will be cleared 
; and reparsed on an extensions reload, or Asterisk reload.
;
; If clearglobalvars is not set, then global variables will persist
; through reloads, and even if deleted from the extensions.conf or
; one of its included files, will remain set to the previous value.
;
; NOTE: A complication sets in, if you put your global variables into
; the AEL file, instead of the extensions.conf file. With clearglobalvars
; set, a "reload" will often leave the globals vars cleared, because it
; is not unusual to have extensions.conf (which will have no globals)
; load after the extensions.ael file (where the global vars are stored).
; So, with "reload" in this particular situation, first the AEL file will
; clear and then set all the global vars, then, later, when the extensions.conf
; file is loaded, the global vars are all cleared, and then not set, because
; they are not stored in the extensions.conf file.
;
clearglobalvars = no
;
; If priorityjumping is set to 'yes', then applications that support
; 'jumping' to a different priority based on the result of their operations
; will do so (this is backwards compatible behavior with pre-1.2 releases
; of Asterisk). Individual applications can also be requested to do this
; by passing a 'j' option in their arguments.
;
;priorityjumping=yes
;
; User context is where entries from users.conf are registered.  The
; default value is 'default'
;
;userscontext=default
;
; You can include other config files, use the #include command
; (without the ';'). Note that this is different from the "include" command
; that includes contexts within other contexts. The #include command works
; in all asterisk configuration files.
;#include "filename.conf"
; The "Globals" category contains global variables that can be referenced
; in the dialplan with the GLOBAL dialplan function:
; ${GLOBAL(VARIABLE)}
; ${${GLOBAL(VARIABLE)}} or ${text${GLOBAL(VARIABLE)}} or any hybrid
; Unix/Linux environmental variables can be reached with the ENV dialplan
; function: ${ENV(VARIABLE)}
;
[globals]
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
;
; Note the 'G2' in the TRUNK variable above. It specifies which group (defined
; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to use in
; the specified group. The four possible options are:
;
; g: select the lowest-numbered non-busy Zap channel
;    (aka. ascending sequential hunt group).
; G: select the highest-numbered non-busy Zap channel
;    (aka. descending sequential hunt group).
; r: use a round-robin search, starting at the next highest channel than last
;    time (aka. ascending rotary hunt group).
; R: use a round-robin search, starting at the next lowest channel than last
;    time (aka. descending rotary hunt group).
;
TRUNKMSD = 1  ; MSD digits to strip (usually 1 or 0)
trunk_1 = SIP/trunk_1
trunk_2 = SIP/trunk_2

[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-${DIALSTATUS},1)  ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(${ARG1},u)  ; 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(${ARG1},b)  ; If busy, send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1)  ; If they press #, return to start
exten => _s-.,1,Goto(s-NOANSWER,1)  ; Treat anything else as no answer
exten => a,1,VoicemailMain(${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-${DIALSTATUS},1)  ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(${ARG1},u)  ; 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(${ARG1},b)  ; If busy, send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1)  ; If they press #, return to start
exten => s-DONTCALL,1,Goto(${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-page];
;
; Paging macro:
;
;       Check to see if SIP device is in use and DO NOT PAGE if they are
;
;   ${ARG1} - Device to page
exten => s,1,ChanIsAvail(${ARG1}|js)  ; j is for Jump and s is for ANY call
exten => s,n,GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail)
exten => s,n(autoanswer),Set(_ALERT_INFO="RA")  ; This is for the PolyComs
exten => s,n,SIPAddHeader(Call-Info: Answer-After=0)  ; This is for the Grandstream, Snoms, and Others
exten => s,n,NoOp()  ; Add others here and Post on the Wiki!!!!
exten => s,n,Dial(${ARG1}||)
exten => s,n(fail),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,${GLOBAL(CONSOLE)})
exten => 1235,1,Voicemail(1234,u)  ; Right to voicemail
exten => 1236,1,Dial(Console/dsp)  ; Ring forever
exten => 1236,n,Voicemail(1234,b)  ; Unless busy
;
; # for when they're done with the demo
;
exten => #,1,Playback(demo-thanks)  ; "Thanks for trying the demo"
exten => #,n,Hangup  ; Hang them up.
;
; A timeout and "invalid extension rule"
;
exten => t,1,Goto(#,1)  ; If they take too long, give up
exten => i,1,Playback(invalid)  ; "That's not valid, try again"
;
; Create an extension, 500, for dialing the
; Asterisk demo.
;
exten => 500,1,Playback(demo-abouttotry)  ; Let them know what's going on
exten => 500,n,Dial(IAX2/guest@pbx.digium.com/s@default)  ; Call the Asterisk demo
exten => 500,n,Playback(demo-nogo)  ; Couldn't connect to the demo site
exten => 500,n,Goto(s,6)  ; Return to the start over message.
;
; Create an extension, 600, for evaluating echo latency.
;
exten => 600,1,Playback(demo-echotest)  ; Let them know what's going on
exten => 600,n,Echo  ; Do the echo test
exten => 600,n,Playback(demo-echodone)  ; Let them know it's over
exten => 600,n,Goto(s,6)  ; Start over
;
;   You can use the Macro Page to intercom a individual user
exten => 76245,1,Macro(page,SIP/Grandstream1)
; or if your peernames are the same as extensions
exten => _7XXX,1,Macro(page,SIP/${EXTEN})
;
;
; System Wide Page at extension 7999
;
exten => 7999,1,Set(TIMEOUT(absolute)=60)
exten => 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n|d)
; 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)
;
;   The page context calls up the page macro that sets variables needed for auto-answer
;   It is in is own context to make calling it from the Page() application as simple as 
;   Local/{peername}@page
;
[page]
exten => _X.,1,Macro(page,SIP/${EXTEN})
;[mainmenu]
;
; Example "main menu" context with submenu
;
;exten => s,1,Answer
;exten => s,n,Background(thanks)      ; "Thanks for calling press 1 for sales, 2 for support, ..."
;exten => s,n,WaitExten
;exten => 1,1,Goto(submenu,s,1)
;exten => 2,1,Hangup
;include => default
;
;[submenu]
;exten => s,1,Ringing               ; Make them comfortable with 2 seconds of ringback
;exten => s,n,Wait,2
;exten => s,n,Background(submenuopts)   ; "Thanks for calling the sales department.  Press 1 for steve, 2 for..."
;exten => s,n,WaitExten
;exten => 1,1,Goto(default,steve,1)
;exten => 2,1,Goto(default,mark,2)
[default]
;
; By default we include the demo.  In a production system, you 
; probably don't want to have the demo there.
;
include => demo
exten => 9999,1,VoiceMailMain
exten = 2000,1,Queue(${EXTEN})
exten = o,1,Goto(default,1000,1)

[macro-trunkdial]
exten = s,1,set(CALLERID(all)=${IF($["${LEN(${CALLERID(num)})}" > "6"]?${CALLERID(all)}:${ARG2})})
exten = s,n,Dial(${ARG1})
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Hangup
exten = s-BUSY,1,Hangup
exten = _s-.,1,NoOp

[asterisk_guitools]
exten = executecommand,1,System(${command})
exten = executecommand,n,Hangup()
exten = record_vmenu,1,Answer
exten = record_vmenu,n,Playback(vm-intro)
exten = record_vmenu,n,Record(${var1})
exten = record_vmenu,n,Playback(vm-saved)
exten = record_vmenu,n,Playback(vm-goodbye)
exten = record_vmenu,n,Hangup
exten = play_file,1,Answer
exten = play_file,n,Playback(${var1})
exten = play_file,n,Hangup

[DID_trunk_1]
include = default
exten = _X.,1,Goto(default|o|1)
exten = s,1,ExecIf($[ "${CALLERID(num)}"="" ],SetCallerPres,unavailable)
exten = s,2,ExecIf($[ "${CALLERID(num)}"="" ],Set,CALLERID(all)=unknown <0000000>)
exten = s,3,Goto(default|o|1)

[numberplan-custom-1]
plancomment = Default DialPlan
include = default
include = parkedcalls
exten = _213XXXXX!,1,Macro(trunkdial,${trunk_1}/${EXTEN:0},${trunk_1_cid})
comment = _213XXXXX!,1,hol,standard
exten = _00XXXXX!,1,Macro(trunkdial,${trunk_2}/${EXTEN:0},${trunk_2_cid})
comment = _00XXXXX!,1,disc,standard
exten = _210XXXXX!,1,Macro(trunkdial,${trunk_2}/${EXTEN:0},${trunk_2_cid})
comment = _210XXXXX!,1,astika,standard
exten = _6XXXXX!,1,Macro(trunkdial,${trunk_1}/${EXTEN:0},${trunk_1_cid})
comment = _6XXXXX!,1,kinita,standard

[DID_trunk_2]
include = default

[sdikr]

Θα πρέπει στο αρχείο sip.conf, να επιτρέψεις το anonymous sip calls, καθώς και να ανοίξεις την πόρτα 5060 *ΥΓ τα παραπάνω είναι δοκιμασμένα σε trixbox

[chris25873]

Thnx για την πληροφορία

Το μηχάνημα έχει off το firewall και μπορεί να κάνει κλήσεις κανονικά στο voipdiscount και στο evoice.
Το anonymous το κοιταξα και ειναι yes by default. Γι καλο και για κακο ομως το έκανα yes χειροκίνητα, αλλά πάλι τίιποτα....

[sdikr]

Thnx για την πληροφορία

Το μηχάνημα έχει off το firewall και μπορεί να κάνει κλήσεις κανονικά στο voipdiscount και στο evoice.
Το anonymous το κοιταξα και ειναι yes by default. Γι καλο και για κακο ομως το έκανα yes χειροκίνητα, αλλά πάλι τίιποτα....

Στο trixbox, ζητάει να πας μεσά στο sip-conf, και να αλλάξεις το context το ποιο απλό είναι να το βάλεις from-pstn

[ngoundras]

Εχεις δοκιμάσει να αυξήσεις το vebrosity level στο CLI και να κάνεις μια κλήση προς το asterisk να δείς σε τι context πεφτει η εισερχόμενη κλήση?

[chris25873]

Ευχαριστώ για την βοήθεια, αλλα τελικά βρέθηκε η άκρη.
Αυτό που έλειπε ήταν το insecure=port,invite στο sip.conf. Να 'ναι καλά και τα παιδιά στο awmn που με βοήθησαν, αλλά και εσείς για τον χρόνο σας...!

Thnx!