HIlfs - nichts geht

slm

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Hallo,

Ich bin zurseit daran bei mir im Geschäft ein Asterisk server aufzubauen.
Dabei sollen intern alle telephone auf voip (SIP) umgestellt werden und über Asterisk mit isdn ans telephonetz gehen.
Ich habe dafür einen älteren Rechner mit Suse 8.0 installiert und asterisk-1.0.9 darauf installiert und aein SIPtelephon (ALL7950 SIP) von Conrad gekauft.
Asterisk läuft!

Bevor ich mich an die isdnweiterleitung ranmache, will ich erst das SIP telephon zum laufen bringen.
Ich schaffe es aber nicht das SIP telephon asterisk anzumelden.
Am Netzwerk liegt es nicht, da die dynamische Adressvergabe mit dhcp funktioniert.
Beim wählen auch bei den testnummern #100 etc.. kommt immer nur ein besetztzeichen.
Kann daran liegen das die SIP rfc2833 von asterisk unkompatibel mit der rfc3261 vom sip telephon ist ?

Mein telephon habe ich den Namen 1234 gegeben.

Als sip.conf habe ich die sample datei genommen und mit :
[1234]
type=friend
username=1234
secret=meinpasswort
host=dynamic
disallow=all
allow=ulaw
allow=alaw
dtmfmode=rfc2833
context=from-sip
[1235]
type=friend
username=1235
secret=meinpasswort
host=dynamic
disallow=all
allow=ulaw
allow=alaw
dtmfmode=rfc2833
context=from-sip

erweiter.

Bei extensions.conf habe ich auch die sample datei von der installation genommen und mit:
exten => 1235,1,Answer
exten => 1235,2,Echo

erweitert.


Übrigens beim starten von asterisk kommen paar warnings:

linux:/etc/asterisk # asterisk -c
Asterisk 1.0.9, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer <[email protected]>
=========================================================================
[ Booting......Nov 22 12:25:01 WARNING[15755]: res_musiconhold.c:565 moh_register: Unable to open pseudo channel for timing... Sound may be choppy.
............................................Warning, flexibel rate not heavily tested!
.Nov 22 12:25:01 WARNING[15755]: chan_iax2.c:7477 load_module: Unable to open IAX timing interface: No such device
..............................................Nov 22 12:25:02 WARNING[15755]: chan_skinny.c:2587 reload_config: Unable to get our IP address, Skinny disabled
............. ]
Asterisk Ready.
*CLI>

und

*CLI> sip show peers
Name/username Host Dyn Nat ACL Mask Port Status
1235/1235 (Unspecified) D 255.255.255.255 0 Unmonitored
1234/1234 (Unspecified) D 255.255.255.255 0 Unmonitored
*CLI> sip show ch
channel channels
*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Format
0 active SIP channel(s)



Hilfe............was muss ich den machen damit es läuft??
Und danke im vorraus.
 
Willkommen im Forum

Dabei sollen intern alle telephone auf voip (SIP) umgestellt werden und über Asterisk mit isdn ans telephonetz gehen.

Vernünftiger Ansatz.
Somit spricht auch nichts gegen * 1.2.0, da wird auch per default ztdummy kompiliert, dessen Fehlen die Ursache der Warnings ist.

Bevor ich mich an die isdnweiterleitung ranmache, will ich erst das SIP telephon zum laufen bringen.
Ebenfalls sehr vernünftig.
Übrigens kannst du Conf Dateien komplett und platzsparend posten mit Hilfe des # Symbols über dem Eingabefeld.
Empfehlenswert ist es, die originalen Dateien als Referenz zu speichern
und dann frisch zu beginnen, bzw. alles nicht benötigte zu entfernen.

Wie hast du denn dein Allnet konfiguriert?

Ich würde erst mal zum Testen ein X-Lite konfigurieren. Dann bist du sicher dass *seitig alles stimmt.

Du kannst auch gerne meine Live CD probieren, dort gibt es einen Demo Modus bei dem solche Dinge schon vorkonfiguriert sind. ISDN Unterstützung für AVM Karten ist auch dabei.

Gruß,
Robert
 
vor allem fehlt in der extensions.conf ein Eintrag [from-sip] ...
 
Anfangs wollte ich asterisk 1.2.0 nehmen, doch der kompiliert nicht mit gcc unter version 3.0.
Leider schlug die Installation mit gcc 3.0 fehl, deshalb habe ich auf asterisk 1.0.9 zurück gegriffen.
Übrigens: Mein sip tel hat den Namen 1234, damit will ich auf 1235 telephonieren und auf das echo stoßen.

Mein asterisk server hat die ip 192.168.10.2

Mein telephon bekommt via dhcp die ip 192.168.10.101.
Seine subnetzmaske ist 255.255.255.0
Die router ip (die egentlich nicht benötigt wird): 192.168.10.2
DNS (die auch nicht benötigt wird): 192.168.10.2
SNTP Server IP: nichts


extension.conf:

Code:
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your 
; inbound and outbound calls in Asterisk. 
; 
; This configuration file is reloaded 
; - With the "extensions reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI

;
; The "General" category is for certain variables.  
;
[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 'save dialplan' too
;
writeprotect=no

; 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 ${VARIABLE} or ${ENV(VARIABLE)} for Environmental variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[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=IAX2/user:pass@provider

;
; Any category other than "General" and "Globals" represent 
; extension contexts, which are collections of extensions.  
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal.  In patterns, some characters have special meanings:
;
;   X - any digit from 0-9
;   Z - any digit from 1-9
;   N - any digit from 2-9
;   [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
;   . - wildcard, matches anything remaining (e.g. _9011. matches 
;	anything starting with 9011 excluding 9011 itself)
;
; For example the extension _NXXXXXX would match normal 7 digit dialings, 
; while _1NXXNXXXXXX would represent an area code plus phone number
; preceeded by a one.
;
; Each step of an extension is ordered by priority, which must
; always start with 1 to be considered a valid extension.
;
; Contexts contain several lines, one for each step of each
; extension, which can take one of two forms as listed below,
; with the first form being preferred.  One may include another
; context in the current one as well, optionally with a
; date and time.  Included contexts are included in the order
; they are listed.
;
;[context]
;exten => someexten,priority,application(arg1,arg2,...)
;exten => someexten,priority,application,arg1|arg2...
;
; Timing list for includes is 
;
;   <time range>|<days of week>|<days of month>|<months>
;
;include => daytime|9:00-17:00|mon-fri|*|*
;
; ignorepat can be used to instruct drivers to not cancel dialtone upon
; receipt of a particular pattern.  The most commonly used example is
; of course '9' like this:
;
;ignorepat => 9
;
; so that dialtone remains even after dialing a 9.
;

;
; 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 [url]www.gnophone.com[/url] or [url]www.iaxtel.com[/url]
;
[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,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9011.,2,Congestion

[trunkld]
;
; Long distance context accessed through trunk
;
exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91NXXNXXXXXX,2,Congestion

[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9NXXXXXX,2,Congestion

[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91800NXXXXXX,2,Congestion
exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91888NXXXXXX,2,Congestion
exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91877NXXXXXX,2,Congestion
exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91866NXXXXXX,2,Congestion

[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
;
; 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@bigserver/local

[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

[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/[email protected]/s@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/[email protected])
;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. 
;
exten => 1235,1,Answer
exten => 1235,2,Echo

sip.conf:
Code:
;
; 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@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@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
;
;   reload chan_sip.so		Reload configuration file
;				Active SIP peers will not be reconfigured
;

[general]
context=default			; Default context for incoming calls
;recordhistory=yes		; Record SIP history by default 
				; (see sip history / sip no history)
;realm=mydomain.tld		; 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
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
				; 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
				
;pedantic=yes			; Enable slow, pedantic checking for Pingtel
				; and multiline formatted headers for strict
				; SIP compatibility (defaults to "no")
;tos=184                        ; Set IP QoS to either a keyword or numeric val
;tos=lowdelay                   ; lowdelay,throughput,reliability,mincost,none
;maxexpirey=3600		; Max length of incoming registration we allow
;defaultexpirey=120		; Default length of incoming/outoing registration
;notifymimetype=text/plain	; Allow overriding of mime type in MWI NOTIFY
;videosupport=yes		; Turn on support for SIP video


;disallow=all			; First disallow all codecs
;allow=ulaw			; Allow codecs in order of preference
;allow=ilbc			; Note: codec order is respected only in [general]
;musicclass=default		; Sets the default music on hold class for all SIP calls
				; This may also be set for individual users/peers
;language=en			; Default language setting for all users/peers
				; This may also be set for individual users/peers
;relaxdtmf=yes			; Relax dtmf handling
;rtptimeout=60			; Terminate call if 60 seconds of no RTP activity
				; when we're not on hold
;rtpholdtimeout=300		; Terminate call if 300 seconds of no RTP activity
				; when we're on hold (must be > rtptimeout)
;trustrpid = no			; If Remote-Party-ID should be trusted
;progressinband=no		; If we should generate in-band ringing always
;useragent=Asterisk PBX		; Allows you to change the user agent string
;nat=no				; NAT settings 
                                ; yes = Always ignore info and assume NAT
                                ; no = Use NAT mode only according to RFC3581 
                                ; never = Never attempt NAT mode or RFC3581 support
				; route = Assume NAT, don't send rport (work around more UNIDEN bugs)
;promiscredir = no      ; If yes, allows 302 or REDIR to non-local SIP address
;                       ; Note that promiscredir when redirects are made to the
;                       ; local system will cause loops since SIP is incapable
;                       ; of performing a "hairpin" call.
;
; If regcontext is specified, Asterisk will dynamically 
; create and destroy a NoOp priority 1 extension for a given
; peer who registers or unregisters with us.  The actual extension
; is the 'regexten' parameter of the registering peer or its
; name if 'regexten' is not provided.  More than one regexten may be supplied
; if they are separated by '&'.  Patterns may be used in regexten.
;
;regcontext=iaxregistrations
;
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
;       register => user[:secret[:authuser]]@host[:port][/extension]
;
; If no extension is given, the 's' extension is used. The extension
; needs to be defined in extensions.conf to be able to accept calls
; from this SIP proxy (provider)
;
; host is either a host name defined in DNS or the name of a 
; section defined below.
;
; Examples:
;
;register => 1234:[email protected]	
;
;     This will pass incoming calls to the 's' extension
;
;
;register => 2345:password@sip_proxy/1234
;
;    Register 2345 at sip provider 'sip_proxy'.  Calls from this provider connect to local 
;    extension 1234 in extensions.conf default context, unless you define 
;    unless you configure a [sip_proxy] section below, and configure a context.
;	 Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
;        Tip 2: Use separate type=peer and type=user sections for SIP providers
;                      (instead of type=friend) if you have calls in both directions
  

;externip = 200.201.202.203	; Address that we're going to put in outbound SIP messages
				; if we're behind a NAT

				; The externip and localnet is used
				; when registering and communicating with other proxies
				; that we're registered with
				; You may add multiple local networks.  A reasonable set of defaults
				; are:
;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
;localnet=10.0.0.0/255.0.0.0	; Also RFC1918
;localnet=172.16.0.0/12		; Another RFC1918 with CIDR notation
;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network

;-----------------------------------------------------------------------------------
; Users and peers have different settings available. Friends have all settings,
; since a friend is both a peer and a user
;
; User config options:        Peer configuration:
; --------------------        -------------------
; context                     context
; permit                      permit
; deny                        deny
; secret                      secret
; md5secret                   md5secret
; dtmfmode                    dtmfmode
; canreinvite                 canreinvite
; nat                         nat
; callgroup                   callgroup
; pickupgroup                 pickupgroup
; language                    language
; allow                       allow
; disallow                    disallow
; insecure                    insecure
; trustrpid                   trustrpid
; progressinband              progressinband
; promiscredir                promiscredir
; callerid
; accountcode
; amaflags
; incominglimit
; restrictcid
;                             mailbox
;                             username
;                             template
;                             fromdomain
;                             regexten
;                             fromuser
;                             host
;                             mask
;                             port
;                             qualify
;                             defaultip
;                             rtptimeout
;                             rtpholdtimeout

;[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

;[grandstream1]
;type=friend 			; either "friend" (peer+user), "peer" or "user"
;context=from-sip
;fromuser=grandstream1		; overrides the callerid, e.g. required by FWD
;callerid=John Doe <1234>
;host=192.168.0.23		; we have a static but private IP address
;nat=no				; there is not NAT between phone and Asterisk
;canreinvite=yes		; allow RTP voice traffic to bypass Asterisk
;dtmfmode=info			; either RFC2833 or INFO for the BudgeTone
;incominglimit=1		; permit only 1 outgoing call at a time
				; from the phone to asterisk
mailbox=1234@default  ; mailbox 1234 in voicemail context "default"
disallow=all			; need to disallow=all before we can use allow=
allow=ulaw			; Note: In user sections the order of codecs
				; listed with allow= does NOT matter!
allow=alaw
allow=g723.1			; Asterisk only supports g723.1 pass-thru!
allow=g729			; Pass-thru only unless g729 license obtained


;[xlite1]
;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
;type=friend
;regexten=1234                 ; When they register, create extension 1234
;username=xlite1
;callerid="Jane Smith" <5678>
;host=dynamic
;nat=yes                       ; X-Lite is behind a NAT router
;canreinvite=no                ; Typically set to NO if behind NAT
;disallow=all
;allow=gsm                     ; GSM consumes far less bandwidth than ulaw
;allow=ulaw
;allow=alaw


;[snom]
;type=friend			; Friends place calls and receive calls
;context=from-sip		; Context for incoming calls from this user
;secret=blah
;language=de			; Use German prompts for this user 
;host=dynamic			; This peer register with us
;dtmfmode=inband		; Choices are inband, rfc2833, or info
;defaultip=192.168.10.101		; IP used until peer registers
;username=snom			; Username to use in INVITE until peer registers
;mailbox=1234,2345		; Mailboxes for message waiting indicator
;restrictcid=yes		; To have the callerid restriced -> sent as ANI
;disallow=all
;allow=ulaw                     ; dtmfmode=inband only works with ulaw or alaw!
;mailbox=1234@context,2345      ; Mailbox(-es) for message waiting indicator


;[polycom]
;type=friend			; Friends place calls and receive calls
;context=from-sip		; Context for incoming calls from this user
;secret=blahpoly
;host=dynamic			; This peer register with us
;dtmfmode=rfc2833		; Choices are inband, rfc2833, or info
;username=polly			; Username to use in INVITE until peer registers
;disallow=all
;allow=ulaw                     ; dtmfmode=inband only works with ulaw or alaw!
;progressinband=no		; Polycom phones don't work properly with "never"


;[pingtel]
;type=friend
;username=pingtel
;secret=blah
;host=dynamic
;insecure=yes			; To match a peer based by IP address only and not peer
;insecure=very			; To allow registered hosts to call without re-authenticating
;qualify=1000			; Consider it down if it's 1 second to reply
				; Helps with NAT session
				; qualify=yes uses default value
;callgroup=1,3-4		; We are in caller groups 1,3,4
;pickupgroup=1,3-5		; We can do call pick-p for call group 1,3,4,5
;defaultip=192.168.0.60		; IP address to use if peer has not registred

;[cisco1]
;type=friend
;username=cisco1
;secret=blah
;qualify=200			; Qualify peer is no more than 200ms away
;nat=yes			; This phone may be natted
				; Send SIP and RTP to  IP address that packet is 
				; received from instead of trusting SIP headers 
;host=dynamic			; This device registers with us
;canreinvite=no			; Asterisk by default tries to redirect the
				; RTP media stream (audio) to go directly from
				; the caller to the callee.  Some devices do not
				; support this (especially if one of them is 
				; behind a NAT).
;defaultip=192.168.0.4

;[cisco2]
;type=friend
;username=cisco2
;fromuser=markster		; Specify user to put in "from" instead of callerid
;fromdomain=yourdomain.com	; Specify domain to put in "from" instead of callerid
				; fromuser and fromdomain are used when Asterisk
				; places calls to this account.  It is not used for
				; calls from this account.
;secret=blah
;host=dynamic
;defaultip=192.168.0.4
;amaflags=default		; Choices are default, omit, billing, documentation
;accountcode=markster		; Users may be associated with an accountcode to ease billing


[1234] 
type=friend 
username=1234
secret=meinpasswort 
host=dynamic
disallow=all 
allow=ulaw 
allow=alaw 
dtmfmode=rfc2833
context=from-sip

[1235]
type=friend
username=1235
secret=meinpasswort
host=dynamic
disallow=all
allow=ulaw
allow=alaw
dtmfmode=rfc2833
context=from-sip
 
Wie schon gesagt, fehlt der context [from-sip] in deiner extensions.conf. Das ist aber erst mal unbedeutend für die Anmeldung des Allnet
Gib mal im CLI sip debug ein, und starte dann dein Telefon neu, tut sich da überhaupt was?
Ansonsten kann ich nur wiederholen was ich bereits bezüglich Entrümpeln der Configs und testen mit X-Lite geschrieben habe.
 
Gemacht:
0 active SIP channel(s)
*CLI> sip debug
SIP Debugging Enabled
*CLI>
*CLI>

telephon neu gestartet....nichts
 
Ich habe zudem alles andere auskommentiert.
Woran erkennt man genau oder wo kann man den nachsehen ob das telephon nun angemeldet ist.
 
- Ich habe alles aus sip.conf wie auch extentions.conf rausgeschmissen was nicht notwenig ist.
- habe [from-sip] in extension.conf eingefügt.
- X-lite geht nicht weil mir libgthread-2.0.so.0 fehlt.

Code:
[general]
[globals]
[from-sip]
exten => 1235,1,Answer
exten => 1235,2,Echo

Code:
[general]
context=default			; Default context for incoming calls
		; Set this to your host name or domain name
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

mailbox=1234@default  ; mailbox 1234 in voicemail context "default"
disallow=all			; need to disallow=all before we can use allow=
allow=ulaw			; Note: In user sections the order of codecs
				; listed with allow= does NOT matter!
allow=alaw
allow=g723.1			; Asterisk only supports g723.1 pass-thru!
allow=g729			; Pass-thru only unless g729 license obtained




[1234] 
type=friend 
username=1234
;secret=einpasswort 
host=dynamic
disallow=all 
allow=ulaw 
allow=alaw 
;dtmfmode=rfc2833
context=from-sip

[1235]
type=friend
username=1235
secret=meinpasswort
host=dynamic
disallow=all
allow=ulaw
allow=alaw
dtmfmode=rfc2833
context=from-sip

Das sip telephon kann ich zwar anpingen aber anmelden tut es sich trotzdem nicht.
Was soll ich nun tun ????
Hilfe .... ist denn jemand hier ???
 
Woran erkennt man genau oder wo kann man den nachsehen ob das telephon nun angemeldet ist.

wenn bei "sip show peers" dein Telefon mit einer IP-Adresse auftaucht, ist es registriert.

Hilfe .... ist denn jemand hier ???

Nicht rumheulen :!:
Deine Asterisk-Konfiguration könnte so erstmal funktionieren. Der Fehler liegt mit ziemlicher Sicherheit - wie schon geschrieben - an einer falschen Konfiguration Deines Telefons.
 
Sorry, ich wusste nicht, dass du Platin-Support gebucht hattest.
Vielleicht solltest du uns, wie ich bereits in meiner ersten Anwort angeregt hatte, die Einstellungen deines Telefons zukommen lassen.
 
Ihr habt recht, es liegt wohl an das scheiss telephon.
Dieses Drecksding sollte von eimen Browser aus fein konfiguriert werden wie z.b. den port etc..
Dummerweise lässt es mich nicht rein da irgendwie ein Passwort schon eingestellt ist.
...ich lass es umtauschen...und auf ein neues....
Es lebe Linux und all die Geduldspielchen für die zarten Nerven.

Danke für eure Hilfe
Stephan
 
PEBKAC

Dieses Drecksding sollte von eimen Browser aus fein konfiguriert werden wie z.b. den port etc..
Dummerweise lässt es mich nicht rein da irgendwie ein Passwort schon eingestellt ist.
Es handelt sich um originalverpackte Neuware?
Du verwendest Port 9999 um mit dem Browser darauf zuzugreifen?
Du hast die Anleitung gelesen und bei der Erstanmeldung die Felder für Benutzer und Kennwort leer gelassen?
Du hast die Webseite von Allnet besucht und dort gelesen wie man bei einem vergessenen Passwort vorgeht?
Du bist überzeugt, dass deine Aktion wirtschaftlich sinnvoll ist und sich für die Kommunikation in deiner Firma dadurch Vorteile ergeben?
Es lebe Linux und all die Geduldspielchen für die zarten Nerven.
Dazu sag ich mal jetzt besser nichts.
 
Ich habe jetzt ein neues sip-telephon geholt und jetzt klappt es auch.
Da haben die mir (conrad) eins gegeben wo schon einer rumgemacht hat, da die Verpackung
auch anders ist.

Alleine kommt man schlecht auf solche Sachen, bei den vielen Einstellungen geht man eher von einem Programmier oder Tipfehler aus.
Vorallem wenn man noch keine Erfahrungen hat und nicht weiss wie das auszusehen hat.

Nochmals vielen dank.
 
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