Ankommend keine Reaktion, abgehend L2-Meldung (0X3302)

Asterisk:/home/tobi # asterisk -vvvvvvvvvvvgc
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a, 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
 [res_musiconhold.so] => (Music On Hold Resource)
  == Parsing '/etc/asterisk/musiconhold.conf': Found
Nov  9 17:34:48 WARNING[1076841088]: res_musiconhold.c:543 moh_register: Unableto open pseudo channel for timing...  Sound may be choppy.
  == Registered application 'MusicOnHold'
  == Registered application 'WaitMusicOnHold'
  == Registered application 'SetMusicOnHold'
 [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(4989099,*,default,0,2) (1,2,64)
  == ast_capi_pvt(4989099,*,default,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_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
  == Registered application 'HoldedCall'
 [res_agi.so] => (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 SLINR, cost 1
  == Registered translator 'lintoadpcm' from format SLINR 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 SLINR, cost 3
  == Registered translator 'lintolpc10' from format SLINR to LPC10, cost 11
 [app_setcidname.so] => (Set CallerID Name)
  == Registered application 'SetCIDName'
 [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 SLINR, cost 1
  == Registered translator 'lintogsm' from format SLINR to GSM, cost 3
 [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 'System'
 [app_record.so] => (Trivial Record Application)
  == Registered application 'Record'
 [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2))
Nov  9 17:34:48 WARNING[1076841088]: chan_iax2.c:7220 load_module: Unable to open IAX timing interface: No such file or directory
  == 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 SLINR, cost 1
  == Registered translator 'lintoulaw' from format SLINR 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 SLINR, cost 1
  == Registered translator 'lintoalaw' from format SLINR 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_pickup.so] => (PickUp/PickDown/Steal)
  == Registered application 'PickDown'
  == Registered application 'Steal'
  == Registered application 'PickUp'
 [codec_speex.so] => (Speex/PCM16 (signed linear) Codec Translator)
  == Registered translator 'speextolin' from format SPEEX to SLINR, cost 2
  == Registered translator 'lintospeex' from format SLINR to SPEEX, cost 47
 [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 SLINR, cost 4
  == Registered translator 'lintog726' from format SLINR 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 SLINR, cost 4
  == Registered translator 'lintoilbc' from format SLINR to ILBC, cost 27
 [pbx_config.so] => (Text Extension Configuration)
  == Parsing '/etc/asterisk/extensions.conf': Found
    -- Registered extension context 'default'
    -- Including context '20' in context 'default'
    -- Including context '21' in context 'default'
    -- Added extension '_xxx.' priority 1 to default
    -- Added extension '4989099' priority 1 to default
    -- Registered extension context '20'
    -- Added extension '20' priority 1 to 20
    -- Added extension '20' priority 2 to 20
    -- Registered extension context '21'
    -- Added extension '21' priority 1 to 21
    -- Added extension '21' priority 2 to 21
 [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_meetme.so] => (MeetMe conference bridge)
  == Registered application 'MeetMeAdmin'
  == Registered application 'MeetMeCount'
  == Registered application 'MeetMe'
 [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'
 [app_flash.so] => (Flash zap trunk application)
  == Registered application 'Flash'
 [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
Nov  9 17:34:48 WARNING[1076841088]: 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 '20' at [email protected]:5060 for 3600
    -- SIP Seeding '21' at [email protected]:5060 for 3600
  == SIP Listening on 0.0.0.0:5060
  == Using TOS bits 0
  == Registered channel type 'SIP' (Session Initiation Protocol (SIP))
  == Registered application 'SIPDtmfMode'
  == Parsing '/etc/asterisk/enum.conf': Found
  == Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger restarted
Asterisk Ready.
*CLI> capi debug
CAPI Debugging Enabled
*CLI>     -- Executing Dial("SIP/21-8dc9", "CAPI/4989099:01636033846") in new stack
    -- data = 4989099:01636033846
    -- capi request omsn = 4989099
  == found capi with omsn = 4989099
  == CAPI Call CAPI[contr1/4989099]/0     -- creating pipe for PLCI=-1
       > sent CONNECT_REQ MN =0x4
    -- Called 4989099:01636033846
    -- CONNECT_CONF ID=002 #0x0004 LEN=0014
  Controller/PLCI/NCCI            = 0x101
  Info                            = 0x0

    -- CONNECT_CONF ID=002 #0x0004 LEN=0014
  Controller/PLCI/NCCI            = 0x101
  Info                            = 0x0

  == received CONNECT_CONF PLCI = 0x101 INFO = 0
    -- DISCONNECT_IND ID=002 #0x0001 LEN=0014
  Controller/PLCI/NCCI            = 0x101
  Reason                          = 0x3302

  == DISCONNECT_IND PLCI=0x101 REASON=0x3302
       > sent DISCONNECT_RESP PLCI=0x101
    -- CAPI Hangingup
    -- removed pipe for PLCI = 0x101
  == No one is available to answer at this time

;
; Asterisk configuration file
;
; Module Loader configuration file
;

[modules]
autoload=yes
;
; If you want, load the GTK console right away.  
; Don't load the KDE console since
; it's not as sophisticated right now.
;
noload => pbx_gtkconsole.so
;load => pbx_gtkconsole.so
noload => pbx_kdeconsole.so
;
; Intercom application is obsoleted by
; chan_oss.  Don't load it.
;
noload => app_intercom.so
;
; Explicitly load the chan_modem.so early on to be sure
; it loads before any of the chan_modem_* 's afte rit
;
noload => chan_modem.so
load => res_musiconhold.so
;
; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not load ALSA
;
noload => chan_elsa.so
noload => chan_oss.so
;
; Module names listed in "global" section will have symbols globally
; exported to modules loaded after them.
;
;
noload => chan_modem_i4l.so
noload => chan_modem_bestdata.so
noload => chan_modem_aopen.so
load => chan_capi.so
;
[global]
;chan_modem.so=yes
chan_capi.so=yes

;
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8

[interfaces]

msn=4989099
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=default
devices=2

;
; 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
;

[general]
context=default			; 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
				; 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
				
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)

[20]
type=friend
ustername=20
secret=
host=dynamic
qualify=1200

[21]
type=friend
username=21
secret=
host=dynamic
qualify=1200

;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your 
; inbound and outbound calls in Asterisk. 
; 

;
; 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]

;
; 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.
;

;[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

[default]
;
; By default we include the demo.  In a production system, you 
; probably don't want to have the demo there.
;
include => 20
include => 21

;exten => _xxx.,1,StripMSD,1
exten => _xxx.,1,Dial,CAPI/4989099:${EXTEN}

exten => 4989099,1,Dial(SIP/20&SIP/21,20,tr)

[20]
exten => 20,1,Dial(SIP/20,20,tr)
;exten => 10,2,Dial(CAPI/4989099:b01788721016,20,tr)
exten => 20,2,Hangup

[21]
exten => 21,1,Dial(SIP/21,45)
;exten => 11,2,Dial(CAPI/4989099:b01788721016,20,tr)
exten => 21,2,Hangup