Asterisk nimmt keine Gespräche an

Casper

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Hallo Leute,

erstmal DANKE für die 1a Seite und FAQ's!
Hatte bis vor einer Woche noch nie was mit Linux und Asterisk zu tun. Dank der guten Anleitungen hier, läuft mein Server jetzt und auch Asterisk funktioniert. - zumindest fast.

Habe folgendes Problem:
Ich kann über X-lite der auf einem Client läuft problemlos raustelefonieren. Nur eingehende Gespräche kommen anscheinend nicht an. Habe die Config Vorlagen von der sipsnip Seite benutzt. http://www.sipsnip.com/de/help.php?folder=14&topic=0#opentopic Darauf hat dann eigentlich auch das raustelefonieren gleich funktioniert, nur kommt eben nichts rein.

Mein Server läuft unter Suse 9.2 mit der aktuellsten Asterisk Version. Der Server ist im Router als DMZ eingetragen.

Kann mir bei dem Problem jemand weiterhelfen und mir mal ein Tip geben?

cu
Casper[web:1feaa38435]http://www.sipsnip.com/de/help.php?folder=14&topic=0#opentopic[/web:1feaa38435]
 
Hmm... :) auch Dir Hallo & willkommen *g*

Casper schrieb:
erstmal DANKE für die 1a Seite und FAQ's!
*freu* :D

Nur eingehende Gespräche kommen anscheinend nicht an. Habe die Config Vorlagen von der sipsnip Seite benutzt.
Hmm... was meint Asterisk denn, wenn Du Deinen sipsnip-account anrufst? Auf der Asterisk-Konsole sollte dann ja irgendwas zu sehen sein. Falls nicht ist schon vorher was mit Firewall/router durcheinander.

Ansonsten sollte Asterisk dort einen Fehler rauswerfen. Poste den doch mal hier, dann sieht man besser was los ist ;)

Gruss,
rajo
 
das ging aber flott :) *freu*

in der konsole wird nichts angezeigt. oder gibts da ein sip befehl mit dem ich mir das protokollieren lassen kann?
 
Casper schrieb:
das ging aber flott :) *freu*

in der konsole wird nichts angezeigt. oder gibts da ein sip befehl mit dem ich mir das protokollieren lassen kann?

wie startest Du asterisk? pack mal ein par v's bei den startaufruf -- also entweder asterisk -gcvvvvvvvvvv oder sowas wie asterisk -rvvvvvv
dann sollte der ein wenig gesprächiger werden.
Alternativ "set verbose <zahl>" (z.b. 4) auf der Asterisk-Konsole.

Dann solltest Du sehen, dass ein Anruf von extern per sipsnip reinkommt.

Weiterhin gibt es noch "sip debug" bzw. "sip no debug". Aber das braucht es eigentlich nur wenn die Konsole gar nix zeigt.
 
hmm also folgendes hat er mir rausgeworfen:

ip read:
0 headers, 0 lines
Feb 4 20:45:02 NOTICE[1590]: chan_sip.c:3999 sip_reregister: -- Re-registration for [email protected]
12 headers, 0 lines
Reliably Transmitting:
REGISTER sip:sipsnip.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.114:5060;branch=z9hG4bK38375483
From: <sip:[email protected]>;tag=as7237c296
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 106 REGISTER
User-Agent: Asterisk PBX
Authorization: Digest username="Account", realm="sipsnip.com", algorithm=MD5,uri="sip:sipsnip.com", nonce="4203d0fede0c39d97b3e8e1af6e6fda7db2a82b3", response="66f0bb97e99c321851b567e9ec1aa683", opaque=""
Expires: 120
Contact: <sip:[email protected]>
Event: registration
Content-Length: 0
(no NAT) to xxx.xxx.xxx.xxx:5060
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.114:5060;branch=z9hG4bK38375483;rport=5060;received=213.54.151.15
From: <sip:[email protected]>;tag=as7237c296
To: <sip:[email protected]>;tag=3eab139ab56eb1eef058d4c1d4845bab.608e
Call-ID: [email protected]
CSeq: 106 REGISTER
Contact: <sip:[email protected]:5060>;q=0.00;expires=120
Server: Sip EXpress router (0.8.14-4 (i386/linux))
Content-Length: 0
Warning: 392 server5.sipsnip.com:5060 "Noisy feedback tells: pid=10894 req_src_ip=xxx.xxx.xxx.xxx req_src_port=5060 in_uri=sip:sipsnip.com out_uri=sip:sipsnip.com via_cnt==1"
10 headers, 0 lines
Feb 4 20:45:02 NOTICE[1590]: chan_sip.c:6779 handle_response: Outbound Registration: Expiry for sipsnip.com is 120 sec (Scheduling reregistration in 105000 ms)
Destroying call '[email protected]'
Sip read:
0 headers, 0 lines
Sip read:
0 headers, 0 lines
Sip read:
0 headers, 0 lines
 
und hier das log vom asterisk start:
Code:
  == Parsing '/etc/asterisk/asterisk.conf': Not found (No such file or directory)
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.0.5, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer <[email protected]>
=========================================================================
  == Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger Started /var/log/asterisk/event_log
  == Manager registered action Ping
  == Manager registered action Events
  == Manager registered action Logoff
  == Manager registered action Hangup
  == Manager registered action Status
  == Manager registered action Setvar
  == Manager registered action Getvar
  == Manager registered action Redirect
  == Manager registered action Originate
  == Manager registered action MailboxStatus
  == Manager registered action Command
  == Manager registered action ExtensionState
  == Manager registered action AbsoluteTimeout
  == Manager registered action MailboxCount
  == Manager registered action ListCommands
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/rtp.conf': Found
  == RTP Allocating from port range 10000 -> 20000
Asterisk PBX Core Initializing
Registering builtin applications:
 [AbsoluteTimeout]
  == Registered application 'AbsoluteTimeout'
 [Answer]
  == Registered application 'Answer'
 [BackGround]
  == Registered application 'BackGround'
 [Busy]
  == Registered application 'Busy'
 [Congestion]
  == Registered application 'Congestion'
 [DigitTimeout]
  == Registered application 'DigitTimeout'
 [Goto]
  == Registered application 'Goto'
 [GotoIf]
  == Registered application 'GotoIf'
 [GotoIfTime]
  == Registered application 'GotoIfTime'
 [Hangup]
  == Registered application 'Hangup'
 [NoOp]
  == Registered application 'NoOp'
 [Prefix]
  == Registered application 'Prefix'
 [Progress]
  == Registered application 'Progress'
 [ResetCDR]
  == Registered application 'ResetCDR'
 [ResponseTimeout]
  == Registered application 'ResponseTimeout'
 [Ringing]
  == Registered application 'Ringing'
 [SayNumber]
  == Registered application 'SayNumber'
 [SayDigits]
  == Registered application 'SayDigits'
 [SayAlpha]
  == Registered application 'SayAlpha'
 [SayPhonetic]
  == Registered application 'SayPhonetic'
 [SetAccount]
  == Registered application 'SetAccount'
 [SetAMAFlags]
  == Registered application 'SetAMAFlags'
 [SetGlobalVar]
  == Registered application 'SetGlobalVar'
 [SetLanguage]
  == Registered application 'SetLanguage'
 [SetVar]
  == Registered application 'SetVar'
 [StripMSD]
  == Registered application 'StripMSD'
 [Suffix]
  == Registered application 'Suffix'
 [Wait]
  == Registered application 'Wait'
 [WaitExten]
  == Registered application 'WaitExten'
Asterisk Dynamic Loader Starting:
  == Parsing '/etc/asterisk/modules.conf': Found
 [chan_capi.so] => (Common ISDN API for Asterisk)
  == Parsing '/etc/asterisk/capi.conf': Found
    -- This box has 1 capi controller(s).
    -- CAPI[contr1] supports DTMF
    -- CAPI[contr1] supports supplementary services
       > sent FACILITY_REQ (CONTROLLER=0x1)
       > FACILITY_CONF INFO = 0
       > HOLD/RETRIEVE
       > TERMINAL PORTABILITY
       > ECT
       > 3PTY
       > CF
       > CD
       > MCID
       > CCBS
       > MWI
       > CCNR
  == ast_capi_pvt(990432,*,capicall,0,2) (1,2,64)
  == ast_capi_pvt(990432,*,capicall,0,2) (1,2,64)
    -- listening on contr1 CIPmask = 0x1fff03ff
  == Registered channel type 'CAPI' (Common ISDN API Driver (0.3.5) aLaw CVS HEAD)
 [res_musiconhold.so] => (Music On Hold Resource)
  == Parsing '/etc/asterisk/musiconhold.conf': Found
  == Registered application 'MusicOnHold'
  == Registered application 'WaitMusicOnHold'
  == Registered application 'SetMusicOnHold'
 [res_indications.so] => (Indications Configuration)
  == Parsing '/etc/asterisk/indications.conf': Found
    -- Registered indication country 'cl'
    -- Registered indication country 'tw'
    -- Registered indication country 'us'
    -- Registered indication country 'au'
    -- Registered indication country 'fr'
    -- Registered indication country 'de'
    -- Registered indication country 'nl'
    -- Registered indication country 'uk'
    -- Registered indication country 'fi'
    -- Registered indication country 'no'
    -- Registered indication country 'br'
    -- Registered indication country 'za'
    -- Registered indication country 'it'
    -- Registered indication country 'us-o'
    -- Registered indication country 'gr'
    -- Registered indication country 'ru'
    -- Registered indication country 'nz'
    -- Setting default indication country to 'us'
  == Registered application 'Playtones'
  == Registered application 'StopPlaytones'
 [res_features.so] => (Call Parking Resource)
  == Parsing '/etc/asterisk/features.conf': Found
    -- Registered extension context 'parkedcalls'
    -- Added extension '700' priority 1 to parkedcalls
  == Registered application 'ParkedCall'
  == Registered application 'Park'
  == Manager registered action ParkedCalls
 [res_agi.so]Warning, flexible rate not heavily tested!
 => (Asterisk Gateway Interface (AGI))
  == Registered application 'DeadAGI'
  == Registered application 'EAGI'
  == Registered application 'AGI'
 [res_crypto.so] => (Cryptographic Digital Signatures)
    -- Loaded PUBLIC key 'iaxtel'
    -- Loaded PUBLIC key 'freeworlddialup'
 [res_adsi.so] => (ADSI Resource)
  == Parsing '/etc/asterisk/adsi.conf': Found
 [res_monitor.so] => (Call Monitoring Resource)
  == Registered application 'Monitor'
  == Registered application 'StopMonitor'
  == Registered application 'ChangeMonitor'
  == Manager registered action Monitor
  == Manager registered action StopMonitor
  == Manager registered action ChangeMonitor
 [app_sms.so] => (SMS/PSTN handler)
  == Registered application 'SMS'
 [app_hasnewvoicemail.so] => (Indicator for whether a voice mailbox has messages in a given folder.
  == Registered application 'HasVoicemail'
  == Registered application 'HasNewVoicemail'
 [format_wav_gsm.so] => (Microsoft WAV format (Proprietary GSM))
  == Registered file format wav49, extension(s) WAV|wav49
 [app_url.so] => (Send URL Applications)
  == Registered application 'SendURL'
 [skipping chan_modem_i4l.so]
 [app_test.so] => (Interface Test Application)
  == Registered application 'TestClient'
  == Registered application 'TestServer'
 [chan_mgcp.so] => (Media Gateway Control Protocol (MGCP))
  == Parsing '/etc/asterisk/mgcp.conf': Found
  == MGCP Listening on 0.0.0.0:2727
  == Using TOS bits 0
  == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP))
 [app_eval.so] => (Reevaluates strings)
  == Registered application 'Eval'
 [app_sendtext.so] => (Send Text Applications)
  == Registered application 'SendText'
 [app_exec.so] => (Executes applications)
  == Registered application 'Exec'
 [app_txtcidname.so] => (TXTCIDName)
  == Registered application 'TXTCIDName'
  == Parsing '/etc/asterisk/enum.conf': Found
 [cdr_manager.so] => (Asterisk Call Manager CDR Backend)
  == Parsing '/etc/asterisk/cdr_manager.conf': Found
 [app_capiCD.so] => ((CAPI*) Call Deflection, the magic thing.)
  == Registered application 'capiCD'
 [app_directory.so] => (Extension Directory)
  == Registered application 'Directory'
 [app_playback.so] => (Trivial Playback Application)
  == Registered application 'Playback'
 [app_capiNoES.so] => ((CAPI*) No Echo Suppression.)
  == Registered application 'capiNoES'
 [codec_adpcm.so] => (Adaptive Differential PCM Coder/Decoder)
  == Registered translator 'adpcmtolin' from format adpcm to slin, cost 1
  == Registered translator 'lintoadpcm' from format slin to adpcm, cost 1
 [chan_local.so] => (Local Proxy Channel)
  == Registered channel type 'Local' (Local Proxy Channel Driver)
 [app_groupcount.so] => (Group Management Routines)
  == Registered application 'GetGroupCount'
  == Registered application 'SetGroup'
  == Registered application 'CheckGroup'
 [app_adsiprog.so] => (Asterisk ADSI Programming Application)
  == Registered application 'ADSIProg'
 [app_chanisavail.so] => (Check if channel is available)
  == Registered application 'ChanIsAvail'
 [app_qcall.so] => (Call from Queue)
 [app_softhangup.so] => (Hangs up the requested channel)
  == Registered application 'SoftHangup'
 [codec_lpc10.so] => (LPC10 2.4kbps (signed linear) Voice Coder)
  == Registered translator 'lpc10tolin' from format lpc10 to slin, cost 4
  == Registered translator 'lintolpc10' from format slin to lpc10, cost 6
 [app_setcidname.so] => (Set CallerID Name)
  == Registered application 'SetCIDName'
 [skipping pbx_gtkconsole.so]
 [format_g726.so] => (Raw G.726 (16/24/32/40kbps) data)
  == Registered file format g726-40, extension(s) g726-40
  == Registered file format g726-32, extension(s) g726-32
  == Registered file format g726-24, extension(s) g726-24
  == Registered file format g726-16, extension(s) g726-16
 [format_g729.so] => (Raw G729 data)
  == Registered file format g729, extension(s) g729
 [app_userevent.so] => (Custom User Event Application)
  == Registered application 'UserEvent'
 [codec_gsm.so] => (GSM/PCM16 (signed linear) Codec Translator)
  == Registered translator 'gsmtolin' from format gsm to slin, cost 1
  == Registered translator 'lintogsm' from format slin to gsm, cost 4
 [app_authenticate.so] => (Authentication Application)
  == Registered application 'Authenticate'
 [format_pcm_alaw.so] => (Raw aLaw 8khz PCM Audio support)
  == Registered file format alaw, extension(s) alaw|al
 [format_ilbc.so] => (Raw iLBC data)
  == Registered file format iLBC, extension(s) ilbc
 [format_h263.so] => (Raw h263 data)
  == Registered file format h263, extension(s) h263
 [app_forkcdr.so] => (Fork The CDR into 2 seperate entities.)
  == Registered application 'ForkCDR'
 [app_ices.so] => (Encode and Stream via icecast and ices)
  == Registered application 'ICES'
 [app_nbscat.so] => (Silly NBS Stream Application)
  == Registered application 'NBScat'
 [codec_a_mu.so] => (A-law and Mulaw direct Coder/Decoder)
  == Registered translator 'alawtoulaw' from format alaw to ulaw, cost 1
  == Registered translator 'ulawtoalaw' from format ulaw to alaw, cost 1
 [app_system.so] => (Generic System() application)
  == Registered application 'TrySystem'
  == Registered application 'System'
 [app_record.so] => (Trivial Record Application)
  == Registered application 'Record'
 [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2))
  == Manager registered action IAXpeers
  == Parsing '/etc/asterisk/iax.conf': Found
  == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2))
  == Using TOS bits 16
  == IAX Ready and Listening on 0.0.0.0 port 4569
  == Loaded firmware 'iaxy.bin'
  == Parsing '/etc/asterisk/iaxprov.conf': Found
    -- Loaded provisioning template 'default'
 [app_milliwatt.so] => (Digital Milliwatt (mu-law) Test Application)
  == Registered application 'Milliwatt'
 [app_parkandannounce.so] => (Call Parking and Announce Application)
  == Registered application 'ParkAndAnnounce'
 [app_sayunixtime.so] => (Say time)
  == Registered application 'SayUnixTime'
  == Registered application 'DateTime'
 [pbx_spool.so] => (Outgoing Spool Support)
 [app_capiMCID.so] => ((CAPI*) Malicious Caller ID, the evil thing.)
  == Registered application 'capiMCID'
 [app_macro.so] => (Extension Macros)
  == Registered application 'Macro'
 [app_random.so] => (Random goto)
  == Registered application 'Random'
 [codec_ulaw.so] => (Mu-law Coder/Decoder)
  == Registered translator 'ulawtolin' from format ulaw to slin, cost 1
  == Registered translator 'lintoulaw' from format slin to ulaw, cost 1
 [app_capiRETRIEVE.so] => ((CAPI*) RETRIEVE)
  == Registered application 'capiRETRIEVE'
 [chan_agent.so] => (Agent Proxy Channel)
  == Registered channel type 'Agent' (Call Agent Proxy Channel)
  == Registered application 'AgentLogin'
  == Registered application 'AgentCallbackLogin'
  == Registered application 'AgentMonitorOutgoing'
  == Parsing '/etc/asterisk/agents.conf': Found
 [app_controlplayback.so] => (Control Playback Application)
  == Registered application 'ControlPlayback'
 [format_jpeg.so] => (JPEG (Joint Picture Experts Group) Image Format)
  == Registered format 'jpg' (JPEG (Joint Picture Experts Group))
 [codec_alaw.so] => (A-law Coder/Decoder)
  == Registered translator 'alawtolin' from format alaw to slin, cost 1
  == Registered translator 'lintoalaw' from format slin to alaw, cost 1
 [app_transfer.so] => (Transfer)
  == Registered application 'Transfer'
 [cdr_csv.so] => (Comma Separated Values CDR Backend)
 [app_voicemail.so] => (Comedian Mail (Voicemail System))
  == Registered application 'VoiceMail'
  == Registered application 'VoiceMail2'
  == Registered application 'VoiceMailMain'
  == Registered application 'VoiceMailMain2'
  == Registered application 'MailboxExists'
  == Parsing '/etc/asterisk/voicemail.conf': Found
 [app_verbose.so] => (Send verbose output)
  == Registered application 'Verbose'
 [app_setcdruserfield.so] => (CDR user field apps)
  == Registered application 'SetCDRUserField'
  == Registered application 'AppendCDRUserField'
  == Manager registered action SetCDRUserField
 [codec_g726.so] => (ITU G.726-32kbps G726 Transcoder)
  == Registered translator 'g726tolin' from format g726 to slin, cost 3
  == Registered translator 'lintog726' from format slin to g726, cost 4
 [app_lookupblacklist.so] => (Look up Caller*ID name/number from blacklist database)
  == Registered application 'LookupBlacklist'
 [app_getcpeid.so] => (Get ADSI CPE ID)
  == Registered application 'GetCPEID'
 [app_enumlookup.so] => (ENUM Lookup)
  == Registered application 'EnumLookup'
  == Parsing '/etc/asterisk/enum.conf': Found
 [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator)
  == Registered translator 'ilbctolin' from format ilbc to slin, cost 4
  == Registered translator 'lintoilbc' from format slin to ilbc, cost 34
 [pbx_config.so] => (Text Extension Configuration)
  == Parsing '/etc/asterisk/extensions.conf': Found
Feb  4 20:49:14 WARNING[1674]: config.c:621 cfg_process: No '=' (equal sign) inline 24 of extensions.conf
    -- Registered extension context 'fromsipsnip'
    -- Added extension 'sipsnip1' priority 1 to fromsipsnip
    -- Added extension 'sipsnip1' priority 2 to fromsipsnip
    -- Added extension 'sipsnip1' priority 3 to fromsipsnip
    -- Registered extension context 'default'
    -- Registered extension context 'app88'
    -- Added extension '_.' priority 1 to app88
    -- Added extension '_.' priority 2 to app88
 [app_read.so] => (Read Variable Application)
  == Registered application 'Read'
 [app_alarmreceiver.so] => (Alarm Receiver for Asterisk)
  == Parsing '/etc/asterisk/alarmreceiver.conf': Found
  == Registered application 'AlarmReceiver'
 [format_gsm.so] => (Raw GSM data)
  == Registered file format gsm, extension(s) gsm
 [app_dial.so] => (Dialing Application)
  == Registered application 'Dial'
 [app_striplsd.so] => (Strip trailing digits)
  == Registered application 'StripLSD'
 [app_capiECT.so] => ((CAPI*) ECT)
  == Registered application 'capiECT'
 [app_disa.so] => (DISA (Direct Inward System Access) Application)
  == Registered application 'DISA'
 [app_cdr.so] => (Make sure asterisk doesn't save CDR for a certain call)
  == Registered application 'NoCDR'
 [app_image.so] => (Image Transmission Application)
  == Registered application 'SendImage'
 [skipping chan_modem_bestdata.so]
 [app_cut.so] => (Cuts up variables)
  == Registered application 'Cut'
 [skipping chan_modem.so]
 [app_festival.so] => (Simple Festival Interface)
  == Registered application 'Festival'
 [app_echo.so] => (Simple Echo Application)
  == Registered application 'Echo'
 [chan_phone.so] => (Linux Telephony API Support)
  == Parsing '/etc/asterisk/phone.conf': Found
  == Registered channel type 'Phone' (Standard Linux Telephony API Driver)
 [format_pcm.so] => (Raw uLaw 8khz Audio support (PCM))
  == Registered file format pcm, extension(s) pcm|ulaw|ul|mu
 [app_privacy.so] => (Require phone number to be entered, if no CallerID sent)
  == Registered application 'PrivacyManager'
 [skipping app_intercom.so]
 [app_setcallerid.so] => (Set CallerID Application)
  == Registered application 'SetCallerPres'
  == Registered application 'SetCallerID'
 [pbx_wilcalu.so] => (Wil Cal U (Auto Dialer))
 [app_capiHOLD.so] => ((CAPI*) HOLD)
  == Registered application 'capiHOLD'
 [app_substring.so] => ((Deprecated) Save substring digits in a given variable)
  == Registered application 'SubString'
 [chan_skinny.so] => (Skinny Client Control Protocol (Skinny))
  == Parsing '/etc/asterisk/skinny.conf': Found
Feb  4 20:49:14 WARNING[1674]: chan_skinny.c:2584 reload_config: Unable to get our IP address, Skinny disabled
  == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny))
 [format_sln.so] => (Raw Signed Linear Audio support (SLN))
  == Registered file format sln, extension(s) sln|raw
 [app_zapateller.so] => (Block Telemarketers with Special Information Tone)
  == Registered application 'Zapateller'
 [app_queue.so] => (True Call Queueing)
  == Registered application 'Queue'
  == Manager registered action Queues
  == Manager registered action QueueStatus
  == Manager registered action QueueAdd
  == Manager registered action QueueRemove
  == Registered application 'AddQueueMember'
  == Registered application 'RemoveQueueMember'
  == Parsing '/etc/asterisk/queues.conf': Found
 [app_mp3.so] => (Silly MP3 Application)
  == Registered application 'MP3Player'
 [app_lookupcidname.so] => (Look up CallerID Name from local database)
  == Registered application 'LookupCIDName'
 [format_wav.so] => (Microsoft WAV format (8000hz Signed Linear))
  == Registered file format wav, extension(s) wav
 [app_senddtmf.so] => (Send DTMF digits Application)
  == Registered application 'SendDTMF'
 [format_vox.so] => (Dialogic VOX (ADPCM) File Format)
  == Registered file format vox, extension(s) vox
 [skipping chan_modem_aopen.so]
 [app_waitforring.so] => (Waits until first ring after time)
  == Registered application 'WaitForRing'
 [app_setcidnum.so] => (Set CallerID Number)
  == Registered application 'SetCIDNum'
 [skipping chan_oss.so]
 [app_talkdetect.so] => (Playback with Talk Detection)
  == Registered application 'BackgroundDetect'
 [app_db.so] => (Database access functions for Asterisk extension logic)
  == Registered application 'DBget'
  == Registered application 'DBput'
  == Registered application 'DBdel'
  == Registered application 'DBdeltree'
 [chan_sip.so] => (Session Initiation Protocol (SIP))
  == Parsing '/etc/asterisk/sip.conf': Found
    -- SIP Seeding '88' at [email protected]:5060 for 1800
  == SIP Listening on 192.168.1.114:5060
  == Using TOS bits 0
  == Registered channel type 'SIP' (Session Initiation Protocol (SIP))
  == Registered application 'SIPDtmfMode'
  == Parsing '/etc/asterisk/enum.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
  == Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger restarted
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/enum.conf': Found
  == Parsing '/etc/asterisk/rtp.conf': Found
  == RTP Allocating from port range 10000 -> 20000
[color=red] [/color]
 
hmm. nja das log vom asterisk start nutzt wenig.

Starte asterisk mit vielen v's und ruf dann mal den asterisk an / telefoniere raus -- asterisk sollte dann eigentlich ausgaben bringen was es so alles machen will. Die sind der interessante Teil.
 
hmm so wie es aussieht, kommt anscheinend gar nichts rein. habe gerade nochmal geschaut. wenn ich raus telefoniere, dann loggt der mir ewig viel mit. wenn ich aber auf der 01805 nr anrufe, tut sich gar nichts im log file. :-/
 
Blockt da evtl. eine Firewall etwas zuviel so dass der * nix mitbekommt?
 
Hab mal hier meine extensions.conf

[general]
static=yes
writeprotect=no

[globals]

[fromsipsnip]
;Anrufe fuer snip-Account 1 an App. 88 leiten

exten => sipsnip1,1,SetCallerID(${CALLERIDNUM})
exten => sipsnip1,2,Dial(SIP/88,30,r) ;Anruf an Nebenstelle 88 geben
exten => sipsnip1,3,Hangup

;Diese Mimik beliebig fortsetzen
;Anrufe füsnip-Account 2 an App. 89 leiten
;exten => sipsnip2,1,SetCallerID(${CALLERIDNUM})
;exten => sipsnip2,2,Dial(SIP/89,30,r) ;Anruf an Nebenstelle 89 geben
;exten => sipsnip2,3,Hangup

[default]
;Hier wäen die einzelnen Nebenstellen üen Ihnen zugeordneten
Account raus

[app88]
exten => _.,1,Dial(SIP/${EXTEN}@sipsnip1,30,r)
exten => _.,2,Hangup

;diese Mimik beliebig fortsetzen
;[app89]
;exten => _.,1,Dial(SIP/${EXTEN}@sipsnip2,30,r)
;exten => _.,2,Hangup

und hier die sip.conf

general]
port = 5060
bindaddr=192.168.1.114
language=de
nat=yes
allow=all

register => XXX:[email protected]/sipsnip

[sipsnip1] ; das ist der OUTGOING Kontext füsnip-Benutzer 1
type=peer
username=xxx
fromuser=xxx
secret=xxx
context=default
host=sipsnip.com
fromdomain=sipsnip.com
insecure=very
reinvite=no
canreinvite=no
nat=yes
allow=all

; [sipsnip2] ; hier kaemen analog die OUTGOING-Kontexte fütere
; sipsnip-Benutzer hin
;

[sipsnip_in] ; das ist der Incoming-Kontext fuer alle ueber einen Asterisk registrierten sipsnip-Accounts !
type=peer
fromdomain=sipsnip.com
host=sipsnip.com
context=fromsipsnip

[88] ; eine Nebenstelle habe ich einfach mal 88 genannt
type=friend
host=dynamic
username=88
secret=geheim
context=app88
insecure=very
caninvite=no
canreinvite=no
allow=ulaw
nat=no ; wenn sie hinter einem Router mit NAT sitzt ! Ansonsten "no"


;[89] ; analog weitere SIP-Nebenstellen definieren
 
hmm eigentlich nicht. hab die asterisk kiste als dmz auf dem router eingetragen. der ganze traffic wird ja dann an die kiste weitergeleitet.

die firewall von suse ist auch abgeschaltet weil ich auch erst dachte dass es daran liegt....
 
hmm, gib mal in deiner Asterisk Konsole "sip show registry" ein, wenn unter State "Registered" steht, dann hast du wahrscheinlich ein Portproblem. In dem Fall würde ich die Ports im Router mal direkt durchrouten (ich traue den DMZ's nicht). Ausserdem würde ich sowieso nie einen Rechner in eine DMZ stellen wo ich nicht 100% sicher bin das der dicht ist (hatte da schon meine Erfahrungen).

Grüssle
 
ja registriert ist er. kann ja auch raustelefonieren...

welche ports braucht denn der asterisk genau?
ich richte dann mal ein portforwarding ein und teste es dann nochmal...
 
Casper schrieb:
welche ports braucht denn der asterisk genau?
ich richte dann mal ein portforwarding ein und teste es dann nochmal...

5060 udp für SIP
irgendeine beliebige udp-Portrange für die RTP-Audioströme. Diese dann auch in der rtp.conf asterisk mitteilen.
 
hab die ports mal eingetragen. funzt aber trotzdem nicht...
:evil:
 
Mein Vorschlag:

sip.conf:

[general]
context=fromsipsnip

[sipsnip1]
context=fromsipsnip

[sipsnip2]
context=fromsipsnip
...

kann imho entfallen:
[sipsnip_in]
 
kapier ich jetzt net... :-/

ich hab in der sip.conf gar kein [global]

hab das mal so geändert - sofern ich das jetzt richtig interpretiert habe:

[general]
port = 5060
bindaddr=192.168.1.114
language=de
nat=yes
allow=all

register => xxx:[email protected]/sipsnip

[sipsnip1] ; das ist der OUTGOING Kontext füsnip-Benutzer 1
type=peer
username=xxx
fromuser=xxx
secret=xxx
context=fromsipsnip
host=sipsnip.com
fromdomain=sipsnip.com
insecure=very
reinvite=no
canreinvite=no
nat=yes
allow=all

; [sipsnip2] ; hier kaemen analog die OUTGOING-Kontexte fütere
; sipsnip-Benutzer hin
;

;[sipsnip_in] ; das ist der Incoming-Kontext fuer alle ueber einen Asterisk registrierten sipsnip-Accounts !

;type=peer
;fromdomain=sipsnip.com
;host=sipsnip.com
;context=fromsipsnip

[88]
type=friend
host=dynamic
username=88
secret=geheim
context=app88
insecure=very
caninvite=no
canreinvite=no
allow=ulaw
nat=yes
 
Sorry - general (kommt bei mir manchmal durcheinander)!
dort auch:
context=fromsipsnip
 
hat auch nichts gebracht...

raus geht, rein nicht.

hab mir das mal durch den kopf gehen lassen.

raustelefonieren geht. das sehe ich auch im debug log.
wenn ich jedoch die 01805 nr. von sipgate anrufe, dann erscheint nichts im log. ich müsste doch zumindest sehen dass da was reinkommt, oder?
lasse ich das telefon dann mal ne weile klingeln, geht irgendwann ne mailbox ran wir beim handy. quasi, gesprächspartner nicht erreichbar oder sowas. kommt das von sipsnip oder vom asterisk?
 
Ich denke ein Problem könnte auch sein, dass du sipsnip mit peer statt mit friend definiert hast. Hierzu gibt es unter voip-info.org die Erklärung, dass bei type=peer nur einmal registriert wird und nicht mehr bei Ablauf der expiration-time. Ändert sich etwas wenn du type=friend auch anstelle von type=peer verwendest?!
Ausserdem was siehst du denn im debug-mode 'asterisk -dddvvvr'?
Kommen denn Anrufe über sipsnip rein?
Bedenke auch, dass du dich nicht selbst bei sipsnip über voip anrufen kannst!
Sipsnip stellt ein 'loop detected' fest und es klingelt nicht (siehst du als Antwort bzw. Fehlermeldung im obigen debug-mode)!
 

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