Hallo Forum,
Ich habe Asterisk auf Kanotix laufen.
Asterisk meldet sich bei Sipgate an.
Ich kann Anrufe bekommen über Sipgate.
Aber abgehen ist etwas faul.
Hier habe ich die Meldung vom CLI Mode:
Hier ist meine extension.cfg :
und hier meine sip.conf:
Dann habe ich noch ein zweites Problem,
Wenn zwei Sipteilnehmer mit einander verbunden sind dann kommt aus dem Lautsprecher => chr chr chr chr chr was kann das sein ???
mfg
Enterprise
Ich habe Asterisk auf Kanotix laufen.
Asterisk meldet sich bei Sipgate an.
Ich kann Anrufe bekommen über Sipgate.
Aber abgehen ist etwas faul.
Hier habe ich die Meldung vom CLI Mode:
Code:
-- Executing Dial("SIP/77-8158", "SIP/06********@sipgate|60") in new stack
-- Called 06*********@sipgate
Jun 5 13:58:17 NOTICE[2653]: chan_sip.c:6875 handle_response: Failed to authenticate on INVITE to '"77" <sip:4318***@si
pgate.net>;tag=as2eac11bf'
== Spawn extension (default, *5206*********, 1) exited non-zero on 'SIP/77-8158'
-- Got SIP response 481 "Call Leg Does Not Exist" back from 217.10.79.9
Asterisk*CLI>Hier ist meine extension.cfg :
Code:
[general]
static=yes
writeprotect=yes
[globals]
[incoming-sipgate]
exten => 4318***,1,Dial(SIP/77)
[incoming-capi] ;ankommend von CAPI
exten => 9,1,SetAccount(isdn)
exten => 9,2,Disa,no-password|default
exten => 44,1,SetLAnguage(de)
exten => 44,2,VoiceMAil(11)
exten => 45,1,Wait(1)
exten => 45,2,Answer()
exten => 45,3,capiHOLD
exten => 45,4,capiECT(${REMOTE-CAPI}:b11)
;exten => 45,3,Answer
;exten => 45,4,Playtones(440)
exten => t,4,Playback(invalid)
[default]
include => intern
include => kennziffern
include => parkedcalls
include => enum
include => sipout
;---------------------------------------------------------------------
kennziffern]
exten => 3333,1,Goto(mainmenu,s,1)
exten => 95,1,SetLanguage(de)
exten => 95,2,VoicemailMain()
exten => 95,3,Hangup()
exten => 96,1,VoiceMail(11)
exten => 900,1,MeetMe(|E)
exten => 901,1,MeetMe(901|M)
exten => 902,1,MeetMe(902|M)
exten => 910,1,MeetMe(9999)
exten => 911,1,Directory(default)
exten => 912,1,SayAlpha(abcdefghijk)
exten => 913,1,Wait(1)
exten => 913,2,Read(SOUND_NO|custom/menu1|2)
exten => 913,3,Record(custom/${SOUND_NO}:gsm)
exten => 913,4,Goto(default,913,2)
exten => 914,1,Read(SOUND_NO|custom/menu1|2)
exten => 914,2,Playback(custom/${SOUND_NO})
exten => 914,3,Goto(default,914,1)
exten => 915,1,Playback(aufnahme)
exten => 920,1,SetLanguage(de)
exten => 920,2,Background(pbx-invalid)
exten => 920,3,Goto(920,1)
exten => 982,1,Goto(mainmenu,s,1)
exten => 983,1,Goto(demo-1,s,1)
exten => 990,1,SetLanguage(de)
exten => 990,2,Answer
exten => 990,3,Queue(gruppe1|Tt|||145)
exten => 990,4,VoiceMail(u77)
exten => 991,1,SetLanguage(de)
exten => 991,2,Answer
exten => 991,3,Queue(gruppe2||||145)
exten => 991,4,VoiceMail(u77)
exten => 998,1,AgentLogin()
exten => 999,1,AgentCallbackLogin()
;-------- Extern Routing --------------------------------------
exten => _0[2-9].,1,SetAccount(extern)
exten => _0[2-9].,2,SetAMAFlags(billing)
exten => _0[2-9].,3,Dial(SIP/${EXTEN}@sipsnip,30)
exten => _0[2-9].,4,Dial(SIP/${EXTEN}@sipgate,30)
exten => _0[2-9].,5,Dial(SIP/${EXTEN}@sipgate,30)
exten => _0[2-9].,7,Playback(invalid)
exten => _0[1].,1,SetAccount(mobile)
exten => _0[1].,2,SetAMAFlags(billing)
exten => _0[1].,3,Dial(SIP/${EXTEN}@freenet,30)
exten => _0[1].,4,Dial(SIP/${EXTEN}@sipsnip,30)
exten => _0[1].,5,Dial(SIP/${EXTEN}@sipgate,30)
exten => _0[1].,6,Playback(invalid)
exten => _00.,1,SetAccount(ausland)
exten => _00.,2,SetAMAFlags(billing)
exten => _00.,3,Dial(SIP/${EXTEN}@sipsnip,30)
exten => _00.,4,Dial(SIP/${EXTEN}@sipgate,30)
exten => _00.,5,Dial(SIP/${EXTEN}@sipsnip,30)
exten => _00.,6,Playback(invalid)
exten => _[1-9]XXX.,1,SetAccount(extern)
exten => _[1-9]XXX.,2,SetAMAFlags(billing)
exten => _[1-9]XXX.,3,Dial(SIP/06203${EXTEN}@sipsnip,30)
exten => _[1-9]XXX.,4,Dial(SIP/06203${EXTEN}@freenet,30)
exten => _[1-9]XXX.,5,Dial(SIP/06203${EXTEN}@sipgate,30)
exten => _[1-9]XXX.,6,Playback(invalid
exten => _*50.,1,Dial(SIP/${EXTEN:3}@sipsnip,60)
exten => _*51.,1,Dial(SIP/${EXTEN:3}@freenet,60)
exten => _*52.,1,Dial(SIP/${EXTEN:3}@sipgate,60)
exten => _*53.,1,Dial(SIP/${EXTEN:3}@webde,60)
exten => _*55.,1,Dial(Zap/g1/${EXTEN:3})
exten => _*1XX,1,Dial(${REMOTE-SKC}/${EXTEN:2})
;exten => _[1-2]X,1,Dial(Zap/g1/${EXTEN})
;------------ Kurzwahl ---------------------------------------
exten => 8000,1,Goto(kennziffern,0172********,1)
exten => t,1,Playback(invalid)
;---------------------------------------------------------------
[enum]
exten => _*1.,1,EnumLookup(${EXTEN:2})
exten => _*1.,2,Dial(${ENUM})
[intern]
exten => 77,1,Dial(SIP/77,15,rtT)
exten => 77,2,Queue(gruppe2|r||15)
;exten => 77,2,VoiceMail(u77)
exten => 78,1,Dial(SIP/78,15,rtT)
exten => 78,2,Queue(gruppe1|r||15)
exten => 78,3,VoiceMail(u78)
exten => 11,1,Dial(Zap/g1/11)
exten => 21,1,Dial(Zap/g1/21)
[submenu]
exten => 1,1,Goto(default,66,1)
exten => 2,1,Goto(default,77,2)
exten => s,1,Ringing()
exten => s,2,Wait(2)
exten => s,3,Background(men2)
[mainmenu]
exten => 1,1,Goto(submenu,s,1)
exten => 2,1,Goto(demo,s,1)
exten => 3,1,Dial(SIP/06*********}@freenet,30,tr)
exten => 4,1,Hangup()
exten => 5,1,Directory(default)
exten => s,1,Ringing()
exten => s,2,Wait(4)
exten => s,3,Answer()
exten => s,4,Background(men1)
;[sipout]
;exten => _*52.,1,SetCallerId,4318***
;exten => _*52.,2,Dial(SIP/${EXTEN}@sipgate,30,trg)
;exten => _*52.,3,Hangup
[demo-1]
exten => #,1,Playback(demo-thanks)
exten => #,2,Hangup()
exten => 1000,1,Goto(default,s,1)
exten => 1234,1,Playback(transfer,skip)
exten => 1234,2,Macro(stdexten,1234,${CONSOLE})
exten => 1235,1,Voicemail(u1234)
exten => 1236,1,Dial(Console/dsp)
exten => 1236,2,Voicemail(u1234)
exten => 2,1,BackGround(demo-moreinfo)
exten => 2,2,Goto(s,6)
exten => 3,1,SetLanguage(fr)
exten => 3,2,Goto(s,5)
exten => 500,1,Playback(demo-abouttotry)
exten => 500,2,Dial(IAX2/[email protected]/s@default)
exten => 500,3,Playback(demo-nogo)
exten => 500,4,Goto(s,6)
exten => 600,1,Playback(demo-echotest)
exten => 600,2,Echo()
exten => 600,3,Playback(demo-echodone)
exten => 600,4,Goto(s,6)
exten => i,1,Playback(invalid)
exten => s,1,Wait(1)
exten => s,2,Answer()
exten => s,3,DigitTimeout(5)
exten => s,4,ResponseTimeout(10)
exten => s,5,BackGround(demo-congrats)
exten => s,6,BackGround(demo-instruct)
exten => t,1,Goto(#,1)
[macro-stdexten]
exten => a,1,VoicemailMain(${ARG1})
exten => s,1,Dial(${ARG2},20)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-.,1,Goto(s-NOANSWER,1)
exten => s-BUSY,1,Voicemail(b${ARG1})
exten => s-BUSY,2,Goto(default,s,1)
exten => s-NOANSWER,1,Voicemail(u${ARG1})
exten => s-NOANSWER,2,Goto(default,s,1)und hier meine sip.conf:
Code:
; 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=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=192.168.1.50 ; IP address to bind to (0.0.0.0 binds to all)
;srvlookup=no ; 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=0x18 ; Set IP QoS to either a keyword or numeric val
;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none
maxexpirey=12000 ; Max length of incoming registration we allow
defaultexpirey=12000 ; 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=all
allow=ulaw ; Allow codecs in order of preference
;allow=gsm ; 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=de ; 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=yes ; 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
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
;register => 1234:[email protected]
;
externip = exyz*****.no-ip.com ; Address that we're going to put in outbound SIP messages
;externip = 80.184.174.5 ; 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=192.168.1.0/255.255.255.0; All RFC 1918 addresses are local networks
;canreinvite=no
insecure=very
dtmf=rfc2833
;-------------------------------------------------------------------------
; Sipgate Einstellungen
;-------------------------------------------------------------------------
;port = 5060
bindaddr = 0.0.0.0
;context = sip-out
qualify=no
;disable=all
allow=alaw
;allow=ulaw
;allow=g729
allow=gsm
allow=slinear
srvlookup=yes
canreinvite=yes
;language=de
;register => SIPID:[email protected]/SIPID
;[sipgate-out]
;type=friend
;insecure=very ; otherwise I get authentication errors
;nat=yes
;username=SIPID
;fromuser=SIPID
;fromdomain=sipgate.de
;secret=SIPPW
;host=sipgate.de
;qualify=yes
;-----------------------------------------------------------------------------------
register=>4318***:[email protected]/4318***
[sipgate]
type=friend
username=4318***
secret=passwort
host=sipgate.de
fromuser=4318***
fromdomain=sipgate.net
context=incoming-sipgate
canreinvite=no
qualify=yes
allow=alaw
allow=ulaw
;allow=g729
allow=gsm
insecure=very
nat=yes
dtmf=rfc2833
tos=0x18
accountcode=sipgate
;[freenet]
;type=friend
;username=yxtzkk
;secret=password
;host=freenet.de
;fromuser=yxtzkk
;fromdomain=freenet.de
;context=incoming-freenet
;canreinvite=no
;qualify=yes
;allow=ulaw
;insecure=very
;nat=yes
;dtmf=rfc2833
;tos=0x18
;accountcode=freenet
[77]
type=friend
username=77
secret=77
callerid="77" <77>
host=dynamic
nat=no ; X-Lite is behind a NAT router
canreinvite=no ; Typically set to NO if behind NAT
;disallow=all
allow=alaw
allow=ulaw
;allow=g729
allow=gsm ; GSM consumes far less bandwidth than ulaw
context=default
mailbox=77
[78]
type=friend
username=78
secret=78
callerid="78" <78>
host=dynamic
nat=no ; X-Lite is behind a NAT router
canreinvite=no ; Typically set to NO if behind NAT
;disallow=all
allow=alaw
allow=ulaw
;allow=g729
allow=gsm ; GSM consumes far less bandwidth than ulaw
context=default
mailbox=78Dann habe ich noch ein zweites Problem,
Wenn zwei Sipteilnehmer mit einander verbunden sind dann kommt aus dem Lautsprecher => chr chr chr chr chr was kann das sein ???
mfg
Enterprise
