Hallo
habe folgendes Problem:
Asterisk Bristuffxxx8n, Fedora 2 und OH323, habe OH323 Eingerichtet ist auch registriert: Wenn ein Call reinkommt klingelt der SIP Teilnehmer, wenn ich jetzt jedoch abnehme kommt ein Hangup
*CLI> [4]WrapH323EndPoint::CreateConnection: Creating a H323Connection [1962]
[4]WrapH323Connection::WrapH323Connection: WrapH323Connection created.
[2]WrapH323Connection::OnReceivedSignalSetup: Received SETUP message...
[2]WrapH323Connection::OnAnswerCall: User ----- [Gatekeeeper IP] is calling us...
[3]WrapH323Connection::OnAnswerCall: Call reference: 1962
[3]WrapH323Connection::OnAnswerCall: Call token: ip$Gatekeeeper IP:41660/1962
[3]WrapH323Connection::OnAnswerCall: Call source alias: ----- [Gatekeeeper IP](20)
[3]WrapH323Connection::OnAnswerCall: Call dest alias: xxxxxRufnummer xxxxxRufnummer E164:xxxxxRufnummer ip$Asterisk IP:1720(72)
[3]WrapH323Connection::OnAnswerCall: Call source e164: (0)
[3]WrapH323Connection::OnAnswerCall: Call dest e164: xxxxxRufnummer(14)
[3]WrapH323Connection::OnAnswerCall: Call RDNIS: (0)
[3]WrapH323Connection::OnAnswerCall: Remote Party number:
[3]WrapH323Connection::OnAnswerCall: Remote Party name: ----- [213.30.225.5]
[3]WrapH323Connection::OnAnswerCall: Remote Party address: -----@ip$Gatekeeeper IP:41660
[3]WrapH323Connection::OnAnswerCall: Remote Application: Surpass Siemens 4/130(21)
-- Executing VoiceMail("OH323/R1962", "u14") in new stack
[2]WrapperAPI::h323_answer_call: Answering call.
[2]WrapH323EndPoint::AnswerCall: Request to answer call ip$Gatekeeeper IP:41660/1962
[2]WrapH323EndPoint::AnswerCall: Call answered [ip$Gatekeeeper IP:41660/1962]
-- Playing 'vm-theperson' (language 'en')
[2]WrapH323Connection::OnReceivedReleaseComplete: Received RELEASE COMPLETE message [ip$Gatekeeeper IP:41660/1962]
[2]WrapH323EndPoint::ClearCall: Request to clear call [ip$Gatekeeeper IP:41660/1962]
[2]WrapH323EndPoint::ClearCall: Request to clear call [ip$Gatekeeeper IP:41660/1962]
[2]WrapH323EndPoint::OnConnectionCleared: Connection [ip$Gatekeeeper IP:41660/1962] closed.
-- H.323 call 'ip$Gatekeeeper IP:41660/1962' cleared, reason 24 (Call ended with Q.931 cause)
[2]WrapH323EndPoint::OnConnectionCleared: Call with "----- [213.30.225.5]" completed
[4]WrapH323Connection::WrapH323Connection: WrapH323Connection deleted.
Sep 21 20:49:22 WARNING[6306]: file.c:980 ast_waitstream: Unexpected control subclass '5'
== Spawn extension (from_oh323, (xxxxxx ist die Rufnummer), 1) exited non-zero on 'OH323/R1962'
-- Hungup 'OH323/R1962'
[highlight=red]
das war die Ausgabe in der CLI.
oh323.conf
;
; Configuration file of OpenH323 channel driver
; in diesem File ist bereits alles dokumentiert
;-----------------------------------------
; General configuration options
; (ports, jitter, GK, ...)
;-----------------------------------------
[general]
;
; Address to bind to for incoming connections.
; Default is ALL.
;
listenAddress=195.226.186.70
;
; Port to listen to.
; Default value is 1720.
;
listenPort=1720
;
; Port to connect to.
; (Used only when we don't have a gatekeeper)
; Default value is 1720.
;
connectPort=1720
;
; Configure TCP port range to be used by H.323
;
tcpStart=10000
tcpEnd=20000
;geaendert
;tcpStart=1718
;tcpEnd=1719
;
; Configure UDP port range to be used by H.323
; Note: The port range used by RTP are configured from
; "rtp.conf"
;
udpStart=10000
udpEnd=20000
;
;
;
; Enable fast start (yes,no).
;
fastStart=no
;
; Enable H.245 tunnelling (yes,no).
;
h245Tunnelling=yes
;
; Enable early H.245 messages in call SETUP message.
;
h245inSetup=no
;
; Enable in-band-DTMF detection.
; (Note: Netmeeting uses in-band DTMFs)
;
inBandDTMF=no
;
; Enable silence suppression.
;
silenceSuppression=no
;
; Set jitter buffer (in milliseconds, 20...10000).
;
jitterMin=20
jitterMax=100
;
; Set IP Type-of-Service byte for RTP channels.
; Valid values for this option are:
; lowdelay, throughput, reliability, mincost, none
;
ipTos=none
;
; Set the maximum number of inbound/outbound/simultaneous
; H.323 connections.
;
outboundMax=10
inboundMax=10
simultaneousMax=10
;
; Set the bandwidth limit for H.323 connections.
; The value is in Kbps.
;
;bandwidthLimit=1024
;
; Set tracing options for the wrapper library and for the
; OpenH323 library.
; libTraceFile can be 'stdout' or a full path name to the tracefile.
; Only trace info for OpenH323 is logged in libTraceFile.
;
wrapLibTraceLevel=9
libTraceLevel=9
libTraceFile=/var/log/asterisk/oh323.log
;
; Disable gatekeeper or specify a gatekeeper.
; Valid values for this option are:
; DISABLE,
; DISCOVER,
; <gatekeeper's DNS name>,
; <gatekeeper's ip>,
; GKID:<gatekeeper's id>
;
gatekeeper=IP Adresse
;gatekeeper=DISABLE
;gatekeeper=DISCOVER
;gatekeeper=GKID:RRS
;
;Set the gatekeeper password
;
;gatekeeperPassword=xxx
;
;Set the gatekeeper registration timeout
;
;gatekeeperTTL=600
;
; Set the mode for sending user-input
; Valid values for this option are:
; Q931 - Q.931 Keypad Information Element
; STRING - H.245 string
; TONE - H.245 tone
; RFC2833 - RFC2833
;
userInputMode=TONE
;
; AMA flags (default, omit, billing, documentation)
;
amaFlags=billing
;
; Account code
; dieser Punkt ist nur für das Billing relevant
;accountCode=xxxx
;
; Set the default context of H.323 calls.
;
context=from_oh323
;-----------------------------------------
; Configure H.323 aliases, prefixes and
; related ASTERISK's contexts
;-----------------------------------------
[register]
;
; Aliases/prefixes associated with the default context
; defined in section [general].
;
alias=Rufnummer
;unten funktioniert dieses digit
;alias=asterisk
;gwprefix=heinz
;
; Aliases/prefixes routed in "all-aliases" context.
;
;context=all-aliases
;alias=ASTERISK
;alias=666
;
; Aliases/prefixes routed in "more-aliases" context.
;
;context=more-aliases
;alias=665
;
; Aliases/prefixes routed in "all-prefixes" context.
;
;context=all-prefixes
;gwprefix=00
;gwprefix=01
;
;Aliases/prefixes routed in "more-stuff" context.
;
;context=more-stuff
;alias=664
;gwprefix=02
;-----------------------------------------
; Specify and configure CODEC related
; options
;-----------------------------------------
[codecs]
;
; Define the codec list of the channel driver.
; Every "codec" option may have a "frames" option
; associated with it.
; Valid values for the "codec" option are:
; G711U - G.711 u-Law
; G711A - G.711 A-Law
; G7231 - G.723.1(6.3k)
; G72316K3 - G.723.1(6.3k)
; G72315K3 - G.723.1(5.3k)
; G7231A6K3 - G.723.1A(6.3k)
; G7231A6K3 - G.723.1A(6.3k)
; G726 - G.726(32k)
; G72616K - G.726(16k)
; G72624K - G.726(24k)
; G72632K - G.726(32k)
; G72640K - G.726(40k)
; G728 - G.728
; G729 - G.729
; G729A - G.729A
; G729B - G.729B
; G729AB - G.729AB
; GSM0610 - GSM 0610
; MSGSM - Microsoft GSM Audio Capability
; LPC10 - LPC-10
; Number of frames in RTP packet (if not specified) is 1.
;
codec=G711A
frames=20
;codec=G711U
;frames=20
;codec=GSM0610
;frames=4
;codec=G7231
;frames=2
;codec=G729
;frames=2
Wer hat einen Tipp!!!
DANKE
habe folgendes Problem:
Asterisk Bristuffxxx8n, Fedora 2 und OH323, habe OH323 Eingerichtet ist auch registriert: Wenn ein Call reinkommt klingelt der SIP Teilnehmer, wenn ich jetzt jedoch abnehme kommt ein Hangup
*CLI> [4]WrapH323EndPoint::CreateConnection: Creating a H323Connection [1962]
[4]WrapH323Connection::WrapH323Connection: WrapH323Connection created.
[2]WrapH323Connection::OnReceivedSignalSetup: Received SETUP message...
[2]WrapH323Connection::OnAnswerCall: User ----- [Gatekeeeper IP] is calling us...
[3]WrapH323Connection::OnAnswerCall: Call reference: 1962
[3]WrapH323Connection::OnAnswerCall: Call token: ip$Gatekeeeper IP:41660/1962
[3]WrapH323Connection::OnAnswerCall: Call source alias: ----- [Gatekeeeper IP](20)
[3]WrapH323Connection::OnAnswerCall: Call dest alias: xxxxxRufnummer xxxxxRufnummer E164:xxxxxRufnummer ip$Asterisk IP:1720(72)
[3]WrapH323Connection::OnAnswerCall: Call source e164: (0)
[3]WrapH323Connection::OnAnswerCall: Call dest e164: xxxxxRufnummer(14)
[3]WrapH323Connection::OnAnswerCall: Call RDNIS: (0)
[3]WrapH323Connection::OnAnswerCall: Remote Party number:
[3]WrapH323Connection::OnAnswerCall: Remote Party name: ----- [213.30.225.5]
[3]WrapH323Connection::OnAnswerCall: Remote Party address: -----@ip$Gatekeeeper IP:41660
[3]WrapH323Connection::OnAnswerCall: Remote Application: Surpass Siemens 4/130(21)
-- Executing VoiceMail("OH323/R1962", "u14") in new stack
[2]WrapperAPI::h323_answer_call: Answering call.
[2]WrapH323EndPoint::AnswerCall: Request to answer call ip$Gatekeeeper IP:41660/1962
[2]WrapH323EndPoint::AnswerCall: Call answered [ip$Gatekeeeper IP:41660/1962]
-- Playing 'vm-theperson' (language 'en')
[2]WrapH323Connection::OnReceivedReleaseComplete: Received RELEASE COMPLETE message [ip$Gatekeeeper IP:41660/1962]
[2]WrapH323EndPoint::ClearCall: Request to clear call [ip$Gatekeeeper IP:41660/1962]
[2]WrapH323EndPoint::ClearCall: Request to clear call [ip$Gatekeeeper IP:41660/1962]
[2]WrapH323EndPoint::OnConnectionCleared: Connection [ip$Gatekeeeper IP:41660/1962] closed.
-- H.323 call 'ip$Gatekeeeper IP:41660/1962' cleared, reason 24 (Call ended with Q.931 cause)
[2]WrapH323EndPoint::OnConnectionCleared: Call with "----- [213.30.225.5]" completed
[4]WrapH323Connection::WrapH323Connection: WrapH323Connection deleted.
Sep 21 20:49:22 WARNING[6306]: file.c:980 ast_waitstream: Unexpected control subclass '5'
== Spawn extension (from_oh323, (xxxxxx ist die Rufnummer), 1) exited non-zero on 'OH323/R1962'
-- Hungup 'OH323/R1962'
[highlight=red]
das war die Ausgabe in der CLI.
oh323.conf
;
; Configuration file of OpenH323 channel driver
; in diesem File ist bereits alles dokumentiert
;-----------------------------------------
; General configuration options
; (ports, jitter, GK, ...)
;-----------------------------------------
[general]
;
; Address to bind to for incoming connections.
; Default is ALL.
;
listenAddress=195.226.186.70
;
; Port to listen to.
; Default value is 1720.
;
listenPort=1720
;
; Port to connect to.
; (Used only when we don't have a gatekeeper)
; Default value is 1720.
;
connectPort=1720
;
; Configure TCP port range to be used by H.323
;
tcpStart=10000
tcpEnd=20000
;geaendert
;tcpStart=1718
;tcpEnd=1719
;
; Configure UDP port range to be used by H.323
; Note: The port range used by RTP are configured from
; "rtp.conf"
;
udpStart=10000
udpEnd=20000
;
;
;
; Enable fast start (yes,no).
;
fastStart=no
;
; Enable H.245 tunnelling (yes,no).
;
h245Tunnelling=yes
;
; Enable early H.245 messages in call SETUP message.
;
h245inSetup=no
;
; Enable in-band-DTMF detection.
; (Note: Netmeeting uses in-band DTMFs)
;
inBandDTMF=no
;
; Enable silence suppression.
;
silenceSuppression=no
;
; Set jitter buffer (in milliseconds, 20...10000).
;
jitterMin=20
jitterMax=100
;
; Set IP Type-of-Service byte for RTP channels.
; Valid values for this option are:
; lowdelay, throughput, reliability, mincost, none
;
ipTos=none
;
; Set the maximum number of inbound/outbound/simultaneous
; H.323 connections.
;
outboundMax=10
inboundMax=10
simultaneousMax=10
;
; Set the bandwidth limit for H.323 connections.
; The value is in Kbps.
;
;bandwidthLimit=1024
;
; Set tracing options for the wrapper library and for the
; OpenH323 library.
; libTraceFile can be 'stdout' or a full path name to the tracefile.
; Only trace info for OpenH323 is logged in libTraceFile.
;
wrapLibTraceLevel=9
libTraceLevel=9
libTraceFile=/var/log/asterisk/oh323.log
;
; Disable gatekeeper or specify a gatekeeper.
; Valid values for this option are:
; DISABLE,
; DISCOVER,
; <gatekeeper's DNS name>,
; <gatekeeper's ip>,
; GKID:<gatekeeper's id>
;
gatekeeper=IP Adresse
;gatekeeper=DISABLE
;gatekeeper=DISCOVER
;gatekeeper=GKID:RRS
;
;Set the gatekeeper password
;
;gatekeeperPassword=xxx
;
;Set the gatekeeper registration timeout
;
;gatekeeperTTL=600
;
; Set the mode for sending user-input
; Valid values for this option are:
; Q931 - Q.931 Keypad Information Element
; STRING - H.245 string
; TONE - H.245 tone
; RFC2833 - RFC2833
;
userInputMode=TONE
;
; AMA flags (default, omit, billing, documentation)
;
amaFlags=billing
;
; Account code
; dieser Punkt ist nur für das Billing relevant
;accountCode=xxxx
;
; Set the default context of H.323 calls.
;
context=from_oh323
;-----------------------------------------
; Configure H.323 aliases, prefixes and
; related ASTERISK's contexts
;-----------------------------------------
[register]
;
; Aliases/prefixes associated with the default context
; defined in section [general].
;
alias=Rufnummer
;unten funktioniert dieses digit
;alias=asterisk
;gwprefix=heinz
;
; Aliases/prefixes routed in "all-aliases" context.
;
;context=all-aliases
;alias=ASTERISK
;alias=666
;
; Aliases/prefixes routed in "more-aliases" context.
;
;context=more-aliases
;alias=665
;
; Aliases/prefixes routed in "all-prefixes" context.
;
;context=all-prefixes
;gwprefix=00
;gwprefix=01
;
;Aliases/prefixes routed in "more-stuff" context.
;
;context=more-stuff
;alias=664
;gwprefix=02
;-----------------------------------------
; Specify and configure CODEC related
; options
;-----------------------------------------
[codecs]
;
; Define the codec list of the channel driver.
; Every "codec" option may have a "frames" option
; associated with it.
; Valid values for the "codec" option are:
; G711U - G.711 u-Law
; G711A - G.711 A-Law
; G7231 - G.723.1(6.3k)
; G72316K3 - G.723.1(6.3k)
; G72315K3 - G.723.1(5.3k)
; G7231A6K3 - G.723.1A(6.3k)
; G7231A6K3 - G.723.1A(6.3k)
; G726 - G.726(32k)
; G72616K - G.726(16k)
; G72624K - G.726(24k)
; G72632K - G.726(32k)
; G72640K - G.726(40k)
; G728 - G.728
; G729 - G.729
; G729A - G.729A
; G729B - G.729B
; G729AB - G.729AB
; GSM0610 - GSM 0610
; MSGSM - Microsoft GSM Audio Capability
; LPC10 - LPC-10
; Number of frames in RTP packet (if not specified) is 1.
;
codec=G711A
frames=20
;codec=G711U
;frames=20
;codec=GSM0610
;frames=4
;codec=G7231
;frames=2
;codec=G729
;frames=2
Wer hat einen Tipp!!!
DANKE