H323 nach SIP

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RoSi

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Hallo,
ich habe asterisk 1.9.2.1 mit den asterisk-addons-1.2.3 installiert.
habe auch mit den H.323 installiert.
mit show channeltypes
Code: 
> show channeltypes
Type        Description                    Devicestate  Indications  Transfer
----------  -----------                    -----------  -----------  --------
Zap         Zapata Telephony Driver w/PRI  no           yes          no
Phone       Standard Linux Telephony API D no           no           no
Console     OSS Console Channel Driver     no           yes          no
Feature     Feature Proxy Channel Driver   no           yes          no
Skinny      Skinny Client Control Protocol no           yes          no
Local       Local Proxy Channel Driver     no           yes          no
IAX2        Inter Asterisk eXchange Driver yes          yes          yes
MGCP        Media Gateway Control Protocol no           yes          no
Agent       Call Agent Proxy Channel       yes          yes          no
SIP         Session Initiation Protocol (S yes          yes          yes
OOH323      Objective Systems H323 Channel no           yes          no

kann ich sehen, dass der OOh323 Channel installiert ist.
Aber ich kann leider keinen anrufen.
Habe Netmeeting so konfiguriert, das er meinen Asterisk als Gateway nutzt.
Die Config von ooh323 sieht folgendermaßen aus:

Code: 
[general]
port= 1720
bindaddr=84.19.178.25
h323id=TEL2WEB
e164=100
callerid=TEL2WEB
gatekeeper=DISCOVER
AllowGKRouted = yes
logfile=/tel2web/var/log/h323_log
context=dialplan
disallow=all     ;Note order of disallow/allow is important.
allow=gsm
allow=ulaw
allow=alaw
allow=g729

[testuser]
type=friend
context=h323
; ip=127.0.0.1   ; UPDATE with appropriate ip address
; port=1820    ; UPDATE with appropriate port
host = dynamic
allow=all
; allow=ulaw
; e164=12345
; rtptimeout=60
dtmfmode=rfc2833

ich kann auch einen anruf inizieren, mein SIP-Telefon kliengelt auch, aber dann kommt folgende Fehlermeldung:
Code: 
-- SIP Seeding peer from astdb: '400020' at 400020@<IP-Server>:5060 for
120
    -- SIP/400020-9bc8 answered OOH323/testuser-2f35
Jul  3 10:00:17 WARNING[11979]: src/chan_h323.c:951 ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_5
    -- Attempting native bridge of OOH323/testuser-2f35 and SIP/400020-9bc8

Was kann das sein?
Danke für die Hilfe.

Rolf
 
ja und was passiert dann weiter?
Was ist nun Dein eigentliches Problem?
 
Netmeeting bricht die verbind nach 10 sec ab.
das SIP-Telefon bekommt ein Besetzt.

Gruß

Rolf
 
Also wenn die Verbindung zustandekommt und dann nach 10 Sekunden abbricht, dürfte das nicht unbedingt ein Problem des Asterisk sein, sondern der Netzwerkumgebung.

Oder hab ich was falsch verstanden?
 
Hallo,
ich würde halt gerne wissen, was diese Fehlermeldung eigentlich bedeutet:

"ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_5"

Gruß
 
Das ist doch gar keine Fehlermeldung :rolleyes:

Wenn es eine Fehlermeldung wäre, würde da ERROR stehen und nicht WARNING
 
Bin Neuling in Sachen H.323 und versuche einen H.323 trunk (später auch clients) aufzusetzen.

Was ist zu tun, ich hab eine trixbox installation, die scheinbar on ooh323 enthält, eine ooh323.conf, die beim reload geladen wird:

Code: 
asterisk1*CLI> reload
  == MySQL RealTime reloaded.
 Reloading H.323
    --   == Setting default context to default
    --   == Setting default context to default
 Reloading MGCP
 Reloading SIP
asterisk1*CLI>

aber o.g. Befehl bringt nur:

Code: 
asterisk1*CLI> show channeltypes
Type        Description                    Devicestate  Indications  Transfer
----------  -----------                    -----------  -----------  --------
Feature     Feature Proxy Channel Driver   no           yes          no
SIP         Session Initiation Protocol (S yes          yes          yes
Local       Local Proxy Channel Driver     no           yes          no
MGCP        Media Gateway Control Protocol no           yes          no
Skinny      Skinny Client Control Protocol no           yes          no
Zap         Zapata Telephony Driver w/PRI  no           yes          no
Agent       Call Agent Proxy Channel       yes          yes          no
Phone       Standard Linux Telephony API D no           no           no
IAX2        Inter Asterisk eXchange Driver yes          yes          yes
asterisk1*CLI>

Was ist zu tun ?

/T.
 
Ich habe ooh323 soweit aktiv und eine outbound route definiert. der ooh323 channel wird aus angesprochen, aber danach kommt nur noch "Couldn't make Call". Auf dem Interface passiert nichts. Irgendwelche Ideen ?
asterisk ist 1.2.12.1, addons 1.2.5
Ursprünglich stürzte "asterisk-safe" beim Call ab, schien ein bekannter Fehler gewesen zu sein, im Forum stand, neu kompilieren hilft, und es half... zumindest gegen den Crash.

Aus /var/log/asterisk/debug:
Code: 
Jan  9 10:15:14 DEBUG[9123]: app_dial.c:1635 dial_exec_full: Exiting with DIALSTATUS=CANCEL.
  == Spawn extension (macro-dialout-trunk, s, 21) exited non-zero on 'SIP/710001-0971cca0' in macro 'dialout-trunk'
  == Spawn extension (macro-dialout-trunk, s, 21) exited non-zero on 'SIP/710001-0971cca0'
Jan  9 10:15:14 DEBUG[9123]: cdr_addon_mysql.c:206 mysql_log: cdr_mysql: inserting a CDR record.
Jan  9 10:15:14 DEBUG[9123]: cdr_addon_mysql.c:222 mysql_log: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES ('2007-01-09 10:13:38','sipsrv2_h323','','116770001','from-internal', 'SIP/710001-0971cca0','OOH323/10.100.1.40:1720-ad02','Dial','OOH323/[email protected]:1720|120|r',96,0,'NO ANSWER',3,'')
Jan  9 10:15:14 DEBUG[9123]: chan_sip.c:2433 sip_hangup: update_call_counter(710001) - decrement call limit counter
Jan  9 10:15:14 DEBUG[9123] app_dial.c: Exiting with DIALSTATUS=CANCEL.
Jan  9 10:15:14 DEBUG[9123] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
Jan  9 10:15:14 DEBUG[9123] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES ('2007-01-09 10:13:38','sipsrv2_h323','','116770001','from-internal', 'SIP/710001-0971cca0','OOH323/10.100.1.40:1720-ad02','Dial','OOH323/[email protected]:1720|120|r',96,0,'NO ANSWER',3,'')
Jan  9 10:15:14 DEBUG[9123] chan_sip.c: update_call_counter(710001) - decrement call limit counter
Jan  9 10:15:30 DEBUG[9074]: chan_sip.c:7174 check_user_full: Setting NAT on RTP to 0
Jan  9 10:15:30 DEBUG[9074]: chan_sip.c:1410 __sip_ack: Stopping retransmission on '3c53430dcd14-gcpg7igj0v3j@10-101-255-161' of Response 1: Match Found
Jan  9 10:15:30 DEBUG[9074]: chan_sip.c:7174 check_user_full: Setting NAT on RTP to 0
Jan  9 10:15:30 DEBUG[9074]: chan_sip.c:10530 handle_request_invite: Checking SIP call limits for device 710001
Jan  9 10:15:30 DEBUG[9074]: chan_sip.c:6152 build_route: build_route: Contact hop: <sip:[email protected]:5060;line=r79g9ktm>
    -- Executing Set("SIP/710001-09726628", "EMERGENCYROUTE=YES") in new stack
    -- Executing Macro("SIP/710001-09726628", "dialout-trunk|4|770001||") in new stack
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1589 pbx_substitute_variables_helper_full: Expression result is '1'
    -- Executing GotoIf("SIP/710001-09726628", "1?3:2") in new stack
    -- Goto (macro-dialout-trunk,s,3)
    -- Executing Macro("SIP/710001-09726628", "user-callerid") in new stack
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1589 pbx_substitute_variables_helper_full: Expression result is '0'
    -- Executing GotoIf("SIP/710001-09726628", "0?report") in new stack
Jan  9 10:15:30 DEBUG[9372]: pbx.c:6176 pbx_builtin_gotoif: Not taking any branch
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1589 pbx_substitute_variables_helper_full: Expression result is '0'
    -- Executing GotoIf("SIP/710001-09726628", "0?start") in new stack
Jan  9 10:15:30 DEBUG[9372]: pbx.c:6176 pbx_builtin_gotoif: Not taking any branch
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '710001'
    -- Executing Set("SIP/710001-09726628", "REALCALLERIDNUM=710001") in new stack
    -- Executing NoOp("SIP/710001-09726628", "REALCALLERIDNUM is 710001") in new stack
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '710001'
    -- Executing Set("SIP/710001-09726628", "AMPUSER=710001") in new stack
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '710001'
    -- Executing Set("SIP/710001-09726628", "AMPUSERCIDNAME=710001") in new stack
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1589 pbx_substitute_variables_helper_full: Expression result is '0'
    -- Executing GotoIf("SIP/710001-09726628", "0?report") in new stack
Jan  9 10:15:30 DEBUG[9372]: pbx.c:6176 pbx_builtin_gotoif: Not taking any branch
    -- Executing Set("SIP/710001-09726628", "CALLERID(all)=710001 <710001>") in new stack
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '"710001" <710001>'
    -- Executing NoOp("SIP/710001-09726628", "Using CallerID "710001" <710001>") in new stack
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '710001'
    -- Executing Macro("SIP/710001-09726628", "record-enable|710001|OUT") in new stack
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '0'
    -- Executing GotoIf("SIP/710001-09726628", "0 > 0?2:4") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing AGI("SIP/710001-09726628", "recordingcheck|20070109-101530|1168334130.2") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
PHP Warning:  Unknown(): Unable to load dynamic library '/usr/lib/php4/imap.so' - libc-client.so.0: cannot open shared object file: No such file or directory in Unknown on line 0
Jan  9 10:15:30 DEBUG[9074] chan_sip.c: Setting NAT on RTP to 0
Jan  9 10:15:30 DEBUG[9074] chan_sip.c: Stopping retransmission on '3c53430dcd14-gcpg7igj0v3j@10-101-255-161' of Response 1: Match Found
Jan  9 10:15:30 DEBUG[9074] chan_sip.c: Setting NAT on RTP to 0
Jan  9 10:15:30 DEBUG[9074] chan_sip.c: Checking SIP call limits for device 710001
Jan  9 10:15:30 DEBUG[9074] chan_sip.c: build_route: Contact hop: <sip:[email protected]:5060;line=r79g9ktm>
Jan  9 10:15:30 DEBUG[9372] pbx.c: Expression result is '1'
Jan  9 10:15:30 DEBUG[9372] pbx.c: Expression result is '0'
Jan  9 10:15:30 DEBUG[9372] pbx.c: Not taking any branch
Jan  9 10:15:30 DEBUG[9372] pbx.c: Expression result is '0'
Jan  9 10:15:30 DEBUG[9372] pbx.c: Not taking any branch
Jan  9 10:15:30 DEBUG[9372] pbx.c: Function result is '710001'
Jan  9 10:15:30 DEBUG[9372] pbx.c: Function result is '710001'
Jan  9 10:15:30 DEBUG[9372] pbx.c: Function result is '710001'
Jan  9 10:15:30 DEBUG[9372] pbx.c: Expression result is '0'
Jan  9 10:15:30 DEBUG[9372] pbx.c: Not taking any branch
Jan  9 10:15:30 DEBUG[9372] pbx.c: Function result is '"710001" <710001>'
Jan  9 10:15:30 DEBUG[9372] pbx.c: Function result is '710001'
Jan  9 10:15:30 DEBUG[9372] pbx.c: Function result is '0'
  recordingcheck|20070109-101530|1168334130.2: Outbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing NoOp("SIP/710001-09726628", "No recording needed") in new stack
    -- Executing Macro("SIP/710001-09726628", "outbound-callerid|4") in new stack
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1589 pbx_substitute_variables_helper_full: Expression result is '1'
    -- Executing GotoIf("SIP/710001-09726628", "1?start") in new stack
    -- Goto (macro-outbound-callerid,s,3)
    -- Executing NoOp("SIP/710001-09726628", "REALCALLERIDNUM is 710001") in new stack
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is ''
    -- Executing Set("SIP/710001-09726628", "USEROUTCID=") in new stack
Jan  9 10:15:30 DEBUG[9372]: db.c:200 ast_db_get: Unable to find key '710001/emergency_cid' in family 'DEVICE'
Jan  9 10:15:30 DEBUG[9372]: func_db.c:69 function_db_read: DB: DEVICE/710001/emergency_cid not found in database.
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is ''
    -- Executing Set("SIP/710001-09726628", "EMERGENCYCID=") in new stack
    -- Executing Set("SIP/710001-09726628", "TRUNKOUTCID=sipsrv2_h323") in new stack
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1589 pbx_substitute_variables_helper_full: Expression result is '0'
    -- Executing GotoIf("SIP/710001-09726628", "0?trunkcid") in new stack
Jan  9 10:15:30 DEBUG[9372]: pbx.c:6176 pbx_builtin_gotoif: Not taking any branch
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1589 pbx_substitute_variables_helper_full: Expression result is '1'
    -- Executing GotoIf("SIP/710001-09726628", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,11)
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1589 pbx_substitute_variables_helper_full: Expression result is '0'
    -- Executing GotoIf("SIP/710001-09726628", "0?usercid") in new stack
Jan  9 10:15:30 DEBUG[9372]: pbx.c:6176 pbx_builtin_gotoif: Not taking any branch
    -- Executing Set("SIP/710001-09726628", "CALLERID(all)=sipsrv2_h323") in new stack
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1589 pbx_substitute_variables_helper_full: Expression result is '1'
    -- Executing GotoIf("SIP/710001-09726628", "1?report") in new stack
    -- Goto (macro-outbound-callerid,s,15)
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '"sipsrv2_h323" <>'
    -- Executing NoOp("SIP/710001-09726628", "CallerID set to "sipsrv2_h323" <>") in new stack
    -- Executing Set("SIP/710001-09726628", "GROUP()=OUT_4") in new stack
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '1'
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1589 pbx_substitute_variables_helper_full: Expression result is '0'
    -- Executing GotoIf("SIP/710001-09726628", "0?108") in new stack
Jan  9 10:15:30 DEBUG[9372]: pbx.c:6176 pbx_builtin_gotoif: Not taking any branch
    -- Executing Set("SIP/710001-09726628", "DIAL_NUMBER=770001") in new stack
    -- Executing Set("SIP/710001-09726628", "DIAL_TRUNK=4") in new stack
    -- Executing AGI("SIP/710001-09726628", "fixlocalprefix") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
PHP Warning:  Unknown(): Unable to load dynamic library '/usr/lib/php4/imap.so' - libc-client.so.0: cannot open shared object file: No such file or directory in Unknown on line 0
Jan  9 10:15:30 DEBUG[9117]: manager.c:1249 process_message: Manager received command 'Command'
Jan  9 10:15:30 DEBUG[9117]: manager.c:1249 process_message: Manager received command 'Command'
    -- AGI Script fixlocalprefix completed, returning 0
    -- Executing Set("SIP/710001-09726628", "OUTNUM=+11770001") in new stack
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'AMP'
    -- Executing Set("SIP/710001-09726628", "custom=AMP") in new stack
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1589 pbx_substitute_variables_helper_full: Expression result is '1'
    -- Executing GotoIf("SIP/710001-09726628", "1?16") in new stack
    -- Goto (macro-dialout-trunk,s,16)
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'AMP:OOH323/'
    -- Executing Set("SIP/710001-09726628", "pre_num=AMP:OOH323/") in new stack
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'OUTNUM'
    -- Executing Set("SIP/710001-09726628", "the_num=OUTNUM") in new stack
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '@10.100.1.40:1720'
    -- Executing Set("SIP/710001-09726628", "[email protected]:1720") in new stack
Jan  9 10:15:30 DEBUG[9372]: pbx.c:1589 pbx_substitute_variables_helper_full: Expression result is '1'
    -- Executing GotoIf("SIP/710001-09726628", "1?20:21") in new stack
    -- Goto (macro-dialout-trunk,s,20)
    -- Executing Set("SIP/710001-09726628", "the_num=+11770001") in new stack
 [b]   -- Executing Dial("SIP/710001-09726628", "OOH323/[email protected]:1720|120|r") in new stack
Jan  9 10:15:30 ERROR[9372]: src/chan_h323.c:772 ooh323_call: Failed to make call
    -- Couldn't call [email protected]:1720[/b]
  == Everyone is busy/congested at this time (0:0/0/0)
Jan  9 10:15:30 DEBUG[9372]: app_dial.c:1635 dial_exec_full: Exiting with DIALSTATUS=CHANUNAVAIL.
    -- Executing Goto("SIP/710001-09726628", "s-CHANUNAVAIL|1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing NoOp("SIP/710001-09726628", "Dial failed due to CHANUNAVAIL") in new stack
    -- Executing Macro("SIP/710001-09726628", "outisbusy|") in new stack
    -- Executing Playback("SIP/710001-09726628", "all-circuits-busy-now") in new stack
    -- Playing 'all-circuits-busy-now' (language 'en')
Jan  9 10:15:30 DEBUG[9074]: chan_sip.c:1410 __sip_ack: Stopping retransmission on '3c53430dcd14-gcpg7igj0v3j@10-101-255-161' of Response 2: Match Found
Jan  9 10:15:30 DEBUG[9372] pbx.c: Expression result is '1'
Jan  9 10:15:30 DEBUG[9372] pbx.c: Function result is ''
Jan  9 10:15:30 DEBUG[9372] db.c: Unable to find key '710001/emergency_cid' in family 'DEVICE'
Jan  9 10:15:30 DEBUG[9372] func_db.c: DB: DEVICE/710001/emergency_cid not found in database.
Jan  9 10:15:30 DEBUG[9372] pbx.c: Function result is ''
Jan  9 10:15:30 DEBUG[9372] pbx.c: Expression result is '0'
Jan  9 10:15:30 DEBUG[9372] pbx.c: Not taking any branch
Jan  9 10:15:30 DEBUG[9372] pbx.c: Expression result is '1'
Jan  9 10:15:30 DEBUG[9372] pbx.c: Expression result is '0'
Jan  9 10:15:30 DEBUG[9372] pbx.c: Not taking any branch
Jan  9 10:15:30 DEBUG[9372] pbx.c: Expression result is '1'
Jan  9 10:15:30 DEBUG[9372] pbx.c: Function result is '"sipsrv2_h323" <>'
Jan  9 10:15:30 DEBUG[9372] pbx.c: Function result is '1'
Jan  9 10:15:30 DEBUG[9372] pbx.c: Expression result is '0'
Jan  9 10:15:30 DEBUG[9372] pbx.c: Not taking any branch
Jan  9 10:15:30 DEBUG[9117] manager.c: Manager received command 'Command'
Jan  9 10:15:30 DEBUG[9117] manager.c: Manager received command 'Command'
Jan  9 10:15:30 DEBUG[9372] pbx.c: Function result is 'AMP'
Jan  9 10:15:30 DEBUG[9372] pbx.c: Expression result is '1'
Jan  9 10:15:30 DEBUG[9372] pbx.c: Function result is 'AMP:OOH323/'
Jan  9 10:15:30 DEBUG[9372] pbx.c: Function result is 'OUTNUM'
Jan  9 10:15:30 DEBUG[9372] pbx.c: Function result is '@10.100.1.40:1720'
Jan  9 10:15:30 DEBUG[9372] pbx.c: Expression result is '1'
Jan  9 10:15:30 DEBUG[9372] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL.
Jan  9 10:15:30 DEBUG[9074] chan_sip.c: Stopping retransmission on '3c53430dcd14-gcpg7igj0v3j@10-101-255-161' of Response 2: Match Found
    -- Executing Playback("SIP/710001-09726628", "pls-try-call-later") in new stack
    -- Playing 'pls-try-call-later' (language 'en')
Jan  9 10:15:34 WARNING[9074]: chan_sip.c:1226 retrans_pkt: Maximum retries exceeded on transmission 3c53429db239-k6t77xj1g0uc@10-101-255-161 for seqno 2 (Critical Response)
    -- Executing Macro("SIP/710001-09726628", "hangupcall") in new stack
    -- Executing ResetCDR("SIP/710001-09726628", "w") in new stack
Jan  9 10:15:34 DEBUG[9372]: cdr_addon_mysql.c:206 mysql_log: cdr_mysql: inserting a CDR record.
Jan  9 10:15:34 DEBUG[9372]: cdr_addon_mysql.c:222 mysql_log: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES ('2007-01-09 10:15:30','sipsrv2_h323','','116770001','from-internal', 'SIP/710001-09726628','OOH323/10.100.1.40:1720-a1e4','ResetCDR','w',4,4,'ANSWERED',3,'')
    -- Executing NoCDR("SIP/710001-09726628", "") in new stack
Jan  9 10:15:34 WARNING[9372]: cdr.c:443 ast_cdr_free: CDR on channel 'SIP/710001-09726628' not posted
Jan  9 10:15:34 WARNING[9372]: cdr.c:445 ast_cdr_free: CDR on channel 'SIP/710001-09726628' lacks end
    -- Executing Wait("SIP/710001-09726628", "5") in new stack
Jan  9 10:15:34 DEBUG[9372] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
Jan  9 10:15:34 DEBUG[9372] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES ('2007-01-09 10:15:30','sipsrv2_h323','','116770001','from-internal', 'SIP/710001-09726628','OOH323/10.100.1.40:1720-a1e4','ResetCDR','w',4,4,'ANSWERED',3,'')
    -- Executing Hangup("SIP/710001-09726628", "") in new stack
 
niemand ?

Ich habe mittlerweile versucht den OH323 statt dem OOH323 zu benutzen, aber den bekomme ich garnicht aktiviert ?!
 
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