Sprachübertragung nur in eine Richtung möglich

mazi

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Hallo,

derzeit habe ich das folgende Problem, dass bei einer Verbindung von einem Softwaretelefon zu einem Telefon an einer TK-Anlage die Sprache lediglich vom Telefon an der TK-Anlage an das Softwaretelefon übertragen wird, aber keine Sprachübertragung vom Softwaretelefon an die TK-Anlage erfolgt. Mit einem Protokolltester konnte ich feststellen, dass kein Sprachsignal aus der Fritz-Card (ISDN-Karte) an die TK-Anlage gesendet wird.
Beim Starten von Asterisk wurden die folgenden Logs generiert:

event_log:
Apr 13 10:41:41 asterisk[1176]: Started Asterisk Event Logger
Apr 13 10:41:42 asterisk[1176]: Restarted Asterisk Event Logger

messages:
Apr 13 10:41:42 WARNING[1176]: Unable to open pseudo channel for timing... Sound may be choppy.
Apr 13 10:41:42 WARNING[1176]: Unable to open IAX timing interface: No such device
Apr 13 10:41:42 WARNING[1176]: Unable to open /dev/dsp: No such device


Beim Aufbau der Verbindung vom Softphone zum Endgerät an der TK-Anlage habe ich die folgenden Logs erhalten:

Master.csv:
"","333","8002443","fullaccess","""Martin""<333>","IAX2/martin@martin/1","Modem[i4l]/ttyI0","Dial","Modem/ttyI0:8002443","2005-04-13 10:43:06","2005-04-13 10:43:12","2005-04-13 10:43:25",19,13,"ANSWERED","DOCUMENTATION"


Asterisk Command Line Interface Ausgabe:
smsgate:/etc/asterisk # asterisk -vvvvc
== Parsing '/etc/asterisk/asterisk.conf': Found
== 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_modem.so] => (Generic Voice Modem Driver)
== Parsing '/etc/asterisk/modem.conf': Found
== Loading modem driver chan_modem_i4l.so => (ISDN4Linux Emulated Modem Driver)
-- Configured modem /dev/ttyI0 with driver i4l (Linux ISDN)
-- Configured modem /dev/ttyI1 with driver i4l (Linux ISDN)
== Registered channel type 'Modem' (Generic Voice Modem Channel Driver)
[res_musiconhold.so] => (Music On Hold Resource)
== Parsing '/etc/asterisk/musiconhold.conf': Found
Apr 13 10:41:42 WARNING[1176]: res_musiconhold.c:565 moh_register: Unable to open pseudo channel for timing... Sound may be choppy.
== 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] => (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'
[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'
[chan_zap.so] => (Zapata Telephony w/PRI)
== Parsing '/etc/asterisk/zapata.conf': Found
-- Automatically generated pseudo channel
== Registered channel type 'Zap' (Zapata Telephony Driver w/PRI)
== Registered channel type 'Tor' (Zapata Telephony Driver w/PRI)
== Registered application 'CallingPres'
== Manager registered action ZapTransfer
== Manager registered action ZapHangup
== Manager registered action ZapDialOffhook
== Manager registered action ZapDNDon
== Manager registered action ZapDNDoff
== Manager registered action ZapShowChannels
[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_directory.so] => (Extension Directory)
== Registered application 'Directory'
[app_playback.so] => (Trivial Playback Application)
== Registered application 'Playback'
[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 2
== 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 2
[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'
[skipping chan_alsa.so]
[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))
Apr 13 10:41:42 WARNING[1176]: chan_iax2.c:7468 load_module: Unable to open IAX timing interface: No such device
== Manager registered action IAXpeers
== Parsing '/etc/asterisk/iax.conf': Found
-- Seeding 'michael' at 192.168.168.69:4569 for 60
-- Seeding 'martin' at 192.168.168.191:4569 for 60
== Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2))
== Using TOS bits 8
== 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)
[cdr_pgsql.so] => (PostgreSQL CDR Backend)
== Parsing '/etc/asterisk/cdr_pgsql.conf': Found
[app_zapscan.so] => (Scan Zap channels application)
== Registered application 'ZapScan'
[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_zapras.so] => (Zap RAS Application)
== Registered application 'ZapRAS'
[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 1
== Registered translator 'lintog726' from format slin to g726, cost 2
[app_lookupblacklist.so] => (Look up Caller*ID name/number from blacklist database)
== Registered application 'LookupBlacklist'
[app_zapbarge.so] => (Barge in on Zap channel application)
== Registered application 'ZapBarge'
[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 3
== Registered translator 'lintoilbc' from format slin to ilbc, cost 17
[pbx_config.so] => (Text Extension Configuration)
== Parsing '/etc/asterisk/extensions.conf': Found
-- Registered extension context 'macro-stdiax'
-- Added extension 's' priority 1 to macro-stdiax
-- Added extension 's' priority 2 to macro-stdiax
-- Added extension 's' priority 3 to macro-stdiax
-- Added extension 's' priority 102 to macro-stdiax
-- Added extension 's' priority 103 to macro-stdiax
-- Registered extension context 'fullaccess'
-- Including context 'local' in context 'fullaccess'
-- Registered extension context 'local'
-- Added extension '999' priority 1 to local
-- Added extension '999' priority 2 to local
-- Added extension '999' priority 3 to local
-- Added extension '999' priority 4 to local
-- Added extension '222' priority 1 to local
-- Added extension '333' priority 1 to local
-- Added extension '_XXX.' priority 1 to local
-- Registered extension context 'noaccess'
-- Added extension 's' priority 1 to noaccess
-- Added extension 's' priority 2 to noaccess
-- Added extension 's' priority 3 to noaccess
-- Added extension 's' priority 4 to noaccess
-- Added extension 's' priority 5 to noaccess
[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_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'
[chan_modem_bestdata.so] => (BestData (Conexant V.90 Chipset) VoiceModem Driver)
[app_cut.so] => (Cuts up variables)
== Registered application 'Cut'
[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_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
== Skinny listening on 192.168.168.70:2000
== 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
[chan_modem_aopen.so] => (A/Open (Rockwell Chipset) ITU-2 VoiceModem Driver)
[app_waitforring.so] => (Waits until first ring after time)
== Registered application 'WaitForRing'
[app_setcidnum.so] => (Set CallerID Number)
== Registered application 'SetCIDNum'
[chan_oss.so] => (OSS Console Channel Driver)
Apr 13 10:41:42 WARNING[1176]: chan_oss.c:434 soundcard_init: Unable to open /dev/dsp: No such device
== No sound card detected -- console channel will be unavailable
== Turn off OSS support by adding 'noload=chan_oss.so' in /etc/asterisk/modules.conf
[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 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/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
Asterisk Ready.
-- Accepting AUTHENTICATED call from 192.168.168.191, requested format = 4, actual format = 4
-- Executing Dial("IAX2/martin@martin/1", "Modem/ttyI0:8002443") in new stack
-- Called ttyI0:8002443
-- Modem[i4l]/ttyI0 answered IAX2/martin@martin/1
-- Registered 'martin' (AUTHENTICATED) at 192.168.168.191:4569
-- Hungup 'Modem[i4l]/ttyI0'
== Spawn extension (fullaccess, 8002443, 1) exited non-zero on 'IAX2/martin@martin/1'
-- Hungup 'IAX2/martin@martin/1'
-- Registered 'martin' (AUTHENTICATED) at 192.168.168.191:4569
-- Registered 'martin' (AUTHENTICATED) at 192.168.168.191:4569



Anbei ein Auszug meiner verwendeten Konfigurationsdateien:

extensions.conf:
[general]
static=yes
writeprotect=yes

[globals]

[macro-stdiax]

exten => s,1,Dial(IAX2/${ARG1}|20|Ttr)
exten => s,2,Voicemail2(u${ARG2})
exten => s,3,Hangup
exten => s,102,Voicemail2(b${ARG2})
exten => s,103,Hangup

[fullaccess]
include => local

[local]
exten => 999,1,Answer;
exten => 999,2,Background(demo-congrats)
exten => 999,3,Queue(holdloop)
exten => 999,4,Hangup

exten => 222,1,Macro(stdiax,michael,${EXTEN})
exten => 333,1,Macro(stdiax,martin,${EXTEN})
exten => _XXX.,1,Dial(Modem/ttyI0:${EXTEN})

[noaccess]
exten => s,1,Ring
exten => s,2,Wait(5)
exten => s,3,Answer
exten => s,4,Playback(ss-noservice) ; invalid extension
exten => s,5,Hangup


iax.conf:
[general]
bindaddr=0.0.0.0
bindport=4569
context=noaccess
group=1
callgroup=1
pickupgroup=1
amaflags=default
bandwidth=high ; changed form low to high
allow=all ; changed from disallow to allow
allow=ulaw ; changed from disallow to allow
allow=alaw ; changed from disallow to allow
allow=gsm
allow=iLBC
allow=Speex
jitterbuffer=yes
dropcount=2
maxjitterbuffer=500
maxexccessbuffer=400
tos=throughput
mailboxdetail=yes

[guest]
type=user
context=iaxguest
callerid="Guest IAX User"

[michael]
type=friend
username=michael
secret=password
auth=md5
host=dynamic
context=fullaccess
mailbox=222
callerid="Michael"<222>

[martin]
type=friend
username=martin
secret=password
auth=md5
host=dynamic
context=fullaccess
mailbox=333
callerid="Martin"<333>


modem.conf:
[interfaces]

context=remote
driver=i4l
language=de
type=autodetect
dialtype=tone
mode=ring
group=1
msn=01835991825021
incomingmsn=*
outgoingmsn=333
device => /dev/ttyI0
device => /dev/ttyI1



Eine Übertragung der Sprache zwischen 2 Softwaretelefone erfolgt ohne Probleme in beide Richtungen.
Würde mich sehr freuen, wenn mir jemand weitere Info's zur Behebung meines Problems gibt oder ein Fallbeispiel einstellt.
Bereits im Voraus vielen Dank für Eure Unterstützung.

Gruß

Mazi
 
Hi,

das gleiche Probleme hatte ich kürzlich auch, eingehend klappte alles von ISDN -> CAPI perfekt, aber ausgehend SIP->CAPI hörten entweder beide nix oder der Anrufer nix.
Lösung für Fehler a) war modprobe zaptel + modprobe ztdummy
Lösung für Fehler b) war ich hatte die CVS Version compiliert, eigentlich gab es keine Fehler - naja bis auf das, Lösung: bristuff, schätze es hing mit dem capi Patch zusammen.


Gruss
 
Es kann auch sein, dass eure chan_capi Version nicht mit eurer Kernelversion kompatibel ist. Am besten verwendet ihr das neuste chan_capi_cm.. -> sourceforge
 

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