Optipoint 420 S an Asterisk: kein RTP?

[Telephonmann]

Hi Forum!

Ich hab ein kleines Problem bei dem ich nicht mehr weiter weiß:
Ich habe jetzt insgesamt 5 opti 420 S an den Asterisk gehängt, doch schaffe ich es nicht ein Gespräch zwischen 2 von ihnen zu führen. Jedoch von einem opti raus über die ISDN-Karte im Asterisk bzw vice versa ist kein problem. ->STRANGE
Die SIP Signalisierung scheint einwandfrei zu funktionieren, nur bleibt dann die Leitung stumm. Von Seiten des Asterisk sieht alles ok aus:

 -- Executing Dial("SIP/1622-007cac00", "SIP/1619|30") in new stack
    -- Called 1619
    -- SIP/1619-007d52e0 is ringing
    -- SIP/1619-007d52e0 answered SIP/1622-007cac00
    -- Attempting native bridge of SIP/1622-007cac00 and SIP/1619-007d52e0
  == Spawn extension (default, 1619, 1) exited non-zero on 'SIP/1622-007cac00'

bzw mit SIP debug

 -- Executing Dial("SIP/2000-007d52e0", "SIP/1619|30") in new stack
We're at 192.168.11.66 port 19086
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 12 lines
Reliably Transmitting (no NAT) to 192.168.11.188:5060:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.11.66:5060;branch=z9hG4bK44e88cf6;rport
From: "christophk" <sip:[email protected]>;tag=as722df14e
To: <sip:[email protected]:5060;transport=udp>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 15 Jun 2007 15:32:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 4243 4243 IN IP4 192.168.11.66
s=session
c=IN IP4 192.168.11.66
t=0 0
m=audio 19086 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Called 1619
unstable*CLI>
<-- SIP read from 192.168.11.188:5060:
SIP/2.0 100 Trying
From: "christophk" <sip:[email protected]>;tag=as722df14e
To: <sip:[email protected]:5060>;epid=SC28ab41;tag=8e12774d3b3190e
Via: SIP/2.0/UDP 192.168.11.66:5060;branch=z9hG4bK44e88cf6;rport
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length: 0
User-Agent: optiPoint 410_420/V6 V6 R1.64.4


--- (8 headers 0 lines) ---
unstable*CLI>
<-- SIP read from 192.168.11.188:5060:
SIP/2.0 180 Ringing
From: "christophk" <sip:[email protected]>;tag=as722df14e
To: <sip:[email protected]:5060>;epid=SC28ab41;tag=8e12774d3b3190e
Via: SIP/2.0/UDP 192.168.11.66:5060;branch=z9hG4bK44e88cf6;rport
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length: 0
Allow-Events: talk, hold, conference
Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,UPDATE
Contact: Franz Haberfellner <sip:[email protected]:5060;transport=udp>
User-Agent: optiPoint 410_420/V6 V6 R1.64.4


--- (11 headers 0 lines) ---
    -- SIP/1619-007ed6f0 is ringing
unstable*CLI>
<-- SIP read from 192.168.11.188:5060:
SIP/2.0 200 OK
From: "christophk" <sip:[email protected]>;tag=as722df14e
To: <sip:[email protected]:5060>;epid=SC28ab41;tag=8e12774d3b3190e
Via: SIP/2.0/UDP 192.168.11.66:5060;branch=z9hG4bK44e88cf6;rport
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length: 270
Allow-Events: talk, hold, conference
Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,UPDATE
Content-Type: application/sdp
Supported: replaces
Contact: Franz Haberfellner <sip:[email protected]:5060;transport=udp>
User-Agent: optiPoint 410_420/V6 V6 R1.64.4

v=0
o=MxSIP 0 1224313814 IN IP4 192.168.11.188
s=SIP Call
c=IN IP4 192.168.11.188
t=0 0
m=audio 5002 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

--- (13 headers 13 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.11.188:5002
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:[email protected]:5060;transport=udp>
set_destination: Parsing <sip:[email protected]:5060;transport=udp> for address/port to send to
set_destination: set destination to 192.168.11.188, port 5060
Transmitting (no NAT) to 192.168.11.188:5060:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.11.66:5060;branch=z9hG4bK318aed83;rport
From: "christophk" <sip:[email protected]>;tag=as722df14e
To: <sip:[email protected]:5060;transport=udp>;tag=8e12774d3b3190e
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/1619-007ed6f0 answered SIP/2000-007d52e0
    -- Attempting native bridge of SIP/2000-007d52e0 and SIP/1619-007ed6f0
Scheduling destruction of call '[email protected]' in 32000 ms
set_destination: Parsing <sip:[email protected]:5060;transport=udp> for address/port to send to
set_destination: set destination to 192.168.11.188, port 5060
Reliably Transmitting (no NAT) to 192.168.11.188:5060:
BYE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.11.66:5060;branch=z9hG4bK11c4a4ce;rport
From: "christophk" <sip:[email protected]>;tag=as722df14e
To: <sip:[email protected]:5060;transport=udp>;tag=8e12774d3b3190e
Call-ID: [email protected]
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
  == Spawn extension (default, 1619, 1) exited non-zero on 'SIP/2000-007d52e0'
unstable*CLI>
<-- SIP read from 192.168.11.188:5060:
SIP/2.0 200 OK
From: "christophk" <sip:[email protected]>;tag=as722df14e
To: <sip:[email protected]:5060>;tag=8e12774d3b3190e;epid=SC28ab41
Via: SIP/2.0/UDP 192.168.11.66:5060;branch=z9hG4bK11c4a4ce;rport
Call-ID: [email protected]
CSeq: 103 BYE
Content-Length: 0
Supported: replaces
User-Agent: optiPoint 410_420/V6 V6 R1.64.4


--- (9 headers 0 lines) ---
Destroying call '[email protected]'

doch wird wie gesagt nach der Meldung Attempting native bridge... keine Sprache übertragen. Terminiert wird der Anruf dann wieder ordnungsgemäß.

Auch von anderen SIP-Phones zu den Siemens und umgekehrt ist kein Gespräch möglich, sehr wohl kann ich es aber über ein opti zu einem anderen teilnehmer weiterleiten, dort wird das gespräch dann korrekt übertragen (sofern das nicht auch ein opti ist).

Meine Frage: was mach ich falsch? ;-)
canreinvite=no hab ich schon für alle optis gesetzt, das scheinen sie net recht zu mögen

* Name       : 1619
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : default
  Subscr.Cont. : <Not set>
  Language     : de
  AMA flags    : Unknown
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 4
  Pickupgroup  : 4
  Mailbox      : 1619
  VM Extension : asterisk
  LastMsgsSent : 0/0
  Call limit   : 0
  Dynamic      : Yes
  Callerid     : "" <6902>
  Expire       : 926
  Insecure     : no
  Nat          : RFC3581
  ACL          : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Trust RPID   : No
  Send RPID    : No
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       :
  Addr->IP     : 192.168.11.188 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: 1619
  SIP Options  : (none)
  Codecs       : 0x8000e (gsm|ulaw|alaw|h263)
  Codec Order  : (none)
  Status       : Unmonitored
  Useragent    : optiPoint 410_420/V6 V6 R1.64.4
  Reg. Contact : sip:[email protected]:5060;transport=udp

ist hier irgendwo ein Fehler ersichtlich? die RTP-Ports auf den optis hab ich zwischenzeitlich geändert, hat aber auch nichts gebracht.
ich steh momentan wirklich auf dem schlauch, wär super wenn mir jemand einen Tipp geben könnte, weiter conf-files poste ich natürlich gern wenns weiterhilft.

schon mal danke im voraus

lg
Chris

 

[sip@home]

Hi Chris,

das Verhalten ist wirklich strange.
Beim Durchsehen konnte ich auf die Schnelle auch nichts finden, kenne aber einige die diese Konfiguration am Laufen haben. Die schauen sich das mal an.

Sicher kennst Du dieses Konfig-Dokument für optipoint@Asterisk, ansonsten kannst Du ja mal checken, ob die Konfig der phones bei Dir ähnlich ist:
http://wiki.siemens-enterprise.com/images/2/2b/Admin_Manual_optiPoint_410-420_Asterisk.pdf

In der Fehlerliste habe ich diesen Eintrag gefunden (allerdings nicht explizit für die SW Version, die bei Dir im Einsatz ist):
NA02795635 No RTP on call. After reboot not longer seen
http://wiki.siemens-enterprise.com/index.php/optiPoint_410/420_S_Release_Notes#V6_R2.66.1
Ich nehme an, die phones wurden von Dir schon mehrfach gebootet ;-)

Ciao,
sip@home

 

[Dirk07]

Halten wir einmal fest:

ein externer Call funktioniert ( Opti410S über Asterisk ) rein sowie rausgehend.
ein interner Call funktioniert nicht, da es keine Payload gibt.

Das sieht sehr nach Routing-Problemen aus: Das System wird gefunden, von daher klappt der Callaufbau auch, sowie der externe Call = Payload über das System.
Wie sind die Einstellungen im Phone : IP-addresse / Subnetmask und Default-Route richtig gesetzt?
Kannst du von einem Phone das Andere anpingen ?

Eine andere Idee: Was bringt es wenn du im Asterisk das Codec GSM einmal deaktivierst? Welche Codecs sind im Optie420S aktiv ? G711 und G729 ?

Eine tiefere Fehleranalyse: Wenn du mit dem Ethereal-Trace mal schaust, wird ein RTP Stream zum anderen Phone gesendet? Wenn ja: kommt er am anderen Ende an? Bzw. wenn 2 Opti im gleichen Subnet wie der Asterisk sind, wie sieht es dann aus ?

 

[Wagner_c]

hm, ich habe das selbe Problem, aber auch seit heute gestern ging noch alles einwandfrei.

es ging aber erst dann nicht, als ich mit canreinvite herumgespielt hatte (wollte mit Nistnet arbeiten und hatte vorher auf canreinvite auf no gesetzt)

dann habe ich auch mit Ethereal überprüft was Sache ist...dabei kam folgendes raus, dass die Verbindung hatl normal von statten geht, allerdings danach KEIN RTP-Stream vollzogen wird...(es wird kein rtp Paket verschickt)

allerdings geschieht dies nur, wenn ich canreinvite auf yes gestellt habe, bei no läuft alles bestens...

ich werd jetzt erstmal ein Update der Software am Telefon machen...mal schaun, ob das was hilft ansonsten werd ich hier mal die Tipps ausprobieren

ich meld mich in jedem Fall nochmal!

Grüße

 

[Telephonmann]

Hi Leute, danke schonmal für eure raschen Antworten, es gibt mir wieder Hoffnung

@sip:
Ja, nach dem siemens-asterisk pdf hab ich meine konfig aufgebaut, habs mir auch schon wieder 2 mal durchgelesen, sollte alles passen, sind im grunde ja nur basics (das mit dem canreinvite=no steht da zb nirgends)

wie du richtig vermutest habe ich die geräte schon mehrmals an- und wieder ausgesteckt. (ich muss mich des öfteren mir den HFA-versionen rumschlagen, da wirkt das oft wunder )

@Dirk:
Ich habe die ip/netzwerkeinstellungen überprüft: alle gleiches supnetz, alle richtigen gateway eingetragen, pingtest von jedem zu jedem funktioniert.

was ich jetzt noch gemacht habe:
gsm-codec mit disallow=gsm im abschnitt [general] der sip.conf deaktiviert;
QoS L2/3 auf allen optis deaktiviert;
die Einstellung Gateway IP Address unter SIP Details rausgenommen (kann mir wer sagen für was die is?)

hab mit dem tollen eingebauten trace-tool der optis mal auf die audio-signalisierung gelauscht (SIP Signaling / Call Control und alle RTP getraced)

hier die nebenstelle die anruft:

----------------------------------------------
Phone type: optiPoint 420 economy plus S
Application version: V6 R1.64.4
MAC address: 00:01:e3:28:ab:41
IP address: 192.168.11.188
Terminal no.: 1619
Time zone: 1
-----------------------------------------------
0,366,18-Jun-2007,13:42:32,1005881,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 Failed to find callref for calltype: ccPreConsultationCall : 0
0,379,18-Jun-2007,13:42:34,1006008,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 Failed to find callref for calltype: ccPreConsultationCall : 0
0,380,18-Jun-2007,13:42:34,1006008,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 CALLREC ALLOC = 2 : 0
0,381,18-Jun-2007,13:42:34,1006008,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 Failed to find callref for calltype: ccPrimaryCall : 0
0,382,18-Jun-2007,13:42:34,1006008,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 Failed to find callref for calltype: ccConsultationCall : 0
0,383,18-Jun-2007,13:42:34,1006008,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 Set calltype = ccPrimaryCall for callref 2 : 0
0,384,18-Jun-2007,13:42:34,1006008,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 E_FM_new_outgoing_call: S_FM_idle ---> S_FM_new_call_outgoing : : 0
0,385,18-Jun-2007,13:42:34,1006008,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 E_CC_call_make: S_CC_idle ---> S_CC_call_setup : 2 : 0
0,387,18-Jun-2007,13:42:34,1006008,tIPP_SiPSig,0xc0f04770,347,sg_parser.cpp,13,2,0,0 CC->SIG: sgCallMake:
0,388,18-Jun-2007,13:42:34,1006008,tIPP_SiPSig,0xc0f04770,325,sg_callrec_mgr.cpp,13,2,0,0 INT: sgCallRec_Alloc: ALLOC: cRef=2, numcRefs=1
0,389,18-Jun-2007,13:42:34,1006009,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpOpen()
0,390,18-Jun-2007,13:42:34,1006009,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpStartRxTask() - SessionPtr = C062E25C
0,391,18-Jun-2007,13:42:34,1006009,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpStartRxTask()
0,392,18-Jun-2007,13:42:34,1006009,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpOpen() - Session Ptr = c062e25c
0,393,18-Jun-2007,13:42:34,1006009,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpGetLocalAddress() RTP Port = 138A, RTCP Port = 138B
0,394,18-Jun-2007,13:42:34,1006010,tIPP_SiPSig,0xc0f04770,2125,sg_call_service_int.cpp,13,2,0,0 CODEC: g722_Try, Fallback: g723
0,395,18-Jun-2007,13:42:34,1006010,tIPP_SiPSig,0xc0f04770,544,sg_call_service_int.cpp,13,2,0,0 SIG->STK: InviteA, URL = sip:[email protected]:5060 SigRef=2 StkRef=0 RTP Port=5002, Line ID=0:
0,396,18-Jun-2007,13:42:34,1006011,SipEngine,0xc0e69aa0,10680,sg_engine.cpp,13,2,0,0 INT: CreateCallContext, VecSize=1, Line ID=0: Context=0xc0e03bb0
0,397,18-Jun-2007,13:42:34,1006011,SipEngine,0xc0e69aa0,775,Z:/ipphone1/application/sipsignalling/include/sg_engine_cfg.h,13,2,0,0 SIG->CC: SetConf_non_initiator() xxxxxxxxxxxxxxxxxxxxxxxxxxxxx 0 sig Cref 2
0,398,18-Jun-2007,13:42:34,1006012,SipEngine,0xc0e69aa0,10762,sg_engine.cpp,13,2,0,0 INT: CreateCallContext, CallId: 1
0,399,18-Jun-2007,13:42:34,1006012,SipEngine,0xc0e69aa0,2137,sg_engine.cpp,13,2,0,0 INT Eng::InviteA, sigcref = 2, stackcref = 1, Line ID=0:
0,400,18-Jun-2007,13:42:34,1006013,SipEngine,0xc0e69aa0,376,sg_events.cpp,13,2,0,0 STK->SIG: ProceedingA: Cref=2, Line ID=0:
0,401,18-Jun-2007,13:42:34,1006013,tIPP_SiPSig,0xc0f04770,136,sg_parser.cpp,13,2,0,0 SIG->SIG: sipProceeding:
0,402,18-Jun-2007,13:42:34,1006013,tIPP_SiPSig,0xc0f04770,383,sg_call_service_ext.cpp,13,2,0,0 SIG->CC: sgCallStateProceeding, Cref=2, Line ID=0:
0,403,18-Jun-2007,13:42:34,1006014,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 UI_Notify: state = ccCallSettingUp, reason = Normal : 2 : 0
0,404,18-Jun-2007,13:42:34,1006015,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 E_FM_wait: S_FM_new_call_outgoing ---> S_FM_primary_call_wait_state : : 0
0,405,18-Jun-2007,13:42:34,1006015,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 E_CC_call_proceeding: S_CC_call_setup ---> S_CC_call_proceeding : 2 : 0
0,406,18-Jun-2007,13:42:34,1006015,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 UI_Notify: state = ccCallProceeding, reason = Normal : 2 : 0
0,407,18-Jun-2007,13:42:34,1006017,SipEngine,0xc0e69aa0,940,sg_events.cpp,13,2,0,0 STK->SIG AuthenticationNeeded: Ref = 1, Realm = asterisk, Line ID=0:
0,408,18-Jun-2007,13:42:34,1006017,tIPP_SiPSig,0xc0f04770,257,sg_parser.cpp,13,2,0,0 SIG->SIG: sipAuthenticationNeeded:
0,409,18-Jun-2007,13:42:34,1006017,tIPP_SiPSig,0xc0f04770,2793,sg_call_service_ext.cpp,13,2,0,0 STK->SIG: sgCallAuthenticationNeeded, Line ID=0:
0,410,18-Jun-2007,13:42:34,1006017,tIPP_SiPSig,0xc0f04770,2848,sg_call_service_ext.cpp,13,2,0,0 SIG->STK: SetAuthentificationInfoA: Ref = 1, UserName = 1619, Password = 1234, Line ID=0
0,411,18-Jun-2007,13:42:34,1006021,SipEngine,0xc0e69aa0,7220,sg_engine.cpp,13,2,0,0 InternalEvSesStatus else
0,412,18-Jun-2007,13:42:34,1006032,SipEngine,0xc0e69aa0,7220,sg_engine.cpp,13,2,0,0 InternalEvSesStatus else
0,413,18-Jun-2007,13:42:34,1006033,tIPP_SiPSig,0xc0f04770,143,sg_parser.cpp,13,2,0,0 SIG->SIG: sipCallStatus:
0,414,18-Jun-2007,13:42:34,1006033,tIPP_SiPSig,0xc0f04770,410,sg_call_service_ext.cpp,13,2,0,0 SIG->CC: sgCallState 0xc0f04108
0,415,18-Jun-2007,13:42:34,1006033,tIPP_SiPSig,0xc0f04770,417,sg_call_service_ext.cpp,13,2,0,0 SIG->CC: sgCallState ,Cref=2, status code=180, Line ID=0: 0xc0eac6e0
0,416,18-Jun-2007,13:42:34,1006033,tIPP_SiPSig,0xc0f04770,726,sg_call_service_ext.cpp,13,2,0,0 SIG->CC: sgCallStateRinging, Cref=2, Line ID=0:
0,417,18-Jun-2007,13:42:34,1006033,tIPP_SiPSig,0xc0f04770,412,sg_parser.cpp,13,2,0,0 CC->SIG: sgRtpSourceChange:
0,418,18-Jun-2007,13:42:34,1006033,tIPP_SiPSig,0xc0f04770,1127,sg_call_service_int.cpp,13,2,0,0 SIG->RTP: mediaChange, Cref=2 type=1
0,419,18-Jun-2007,13:42:34,1006033,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpConnectReceive(C062E25C, 0, C0E09590)
0,420,18-Jun-2007,13:42:34,1006033,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpConnectTransmit(C062E25C)
0,421,18-Jun-2007,13:42:34,1006033,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 E_CC_call_ringing_remote: S_CC_call_proceeding ---> S_CC_call_ringing_remote : 2 : 0
0,422,18-Jun-2007,13:42:34,1006033,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 UI_Notify: state = ccCallRingingRemote, reason = Normal : 2 : 0
0,428,18-Jun-2007,13:42:37,1006337,SipEngine,0xc0e69aa0,640,sg_events.cpp,13,2,0,0 STK->SIG: sgMxEvAnswered: Cref=2, Status=0, Line ID=0:
0,429,18-Jun-2007,13:42:37,1006337,tIPP_SiPSig,0xc0f04770,157,sg_parser.cpp,13,2,0,0 SIG->SIG: sipAnswered:
0,430,18-Jun-2007,13:42:37,1006337,tIPP_SiPSig,0xc0f04770,142,sg_sdp_parser.cpp,13,2,0,0 sg_sdpGetCodecType: Num algoritms = 2
0,431,18-Jun-2007,13:42:37,1006337,tIPP_SiPSig,0xc0f04770,188,sg_sdp_parser.cpp,13,2,0,0 Algorithm=0 eG711U; Payload=0
0,432,18-Jun-2007,13:42:37,1006337,tIPP_SiPSig,0xc0f04770,188,sg_sdp_parser.cpp,13,2,0,0 Algorithm=7 eG711A; Payload=8
0,433,18-Jun-2007,13:42:37,1006337,tIPP_SiPSig,0xc0f04770,190,sg_sdp_parser.cpp,13,2,0,0 Set: G711U=1, G711A=1, G723=0, G729=0, G722=0
0,434,18-Jun-2007,13:42:37,1006337,tIPP_SiPSig,0xc0f04770,142,sg_sdp_parser.cpp,13,2,0,0 sg_sdpGetCodecType: Num algoritms = 2
0,435,18-Jun-2007,13:42:37,1006338,tIPP_SiPSig,0xc0f04770,188,sg_sdp_parser.cpp,13,2,0,0 Algorithm=0 eG711U; Payload=0
0,436,18-Jun-2007,13:42:37,1006338,tIPP_SiPSig,0xc0f04770,188,sg_sdp_parser.cpp,13,2,0,0 Algorithm=7 eG711A; Payload=8
0,437,18-Jun-2007,13:42:37,1006338,tIPP_SiPSig,0xc0f04770,190,sg_sdp_parser.cpp,13,2,0,0 Set: G711U=1, G711A=1, G723=0, G729=0, G722=0
0,438,18-Jun-2007,13:42:37,1006338,tIPP_SiPSig,0xc0f04770,2738,sg_call_service_ext.cpp,13,2,0,0 INT->INT - sgUpdateStream oldRx=0 newRx=2 : oldTx=0 newTx=2 : oldPort=0 newPort=16206
0,439,18-Jun-2007,13:42:37,1006338,tIPP_SiPSig,0xc0f04770,2740,sg_call_service_ext.cpp,13,2,0,0 INT->INT - sgUpdateStream : oldIP=0.0.0.0 newIP=192.168.11.66
0,440,18-Jun-2007,13:42:37,1006338,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpClose(C062E25C)
0,441,18-Jun-2007,13:42:37,1006338,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpDisconnectReceive(C062E25C)
0,444,18-Jun-2007,13:42:37,1006338,tIPP_SiPSig,0xc0f04770,233,rtp_component.c,0,4,0,!!!rtp_qdc_close(rtpSessionPtr: c05e0640): Invalid Session pointer parameter
0,445,18-Jun-2007,13:42:37,1006338,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpDisconnectReceive()
0,446,18-Jun-2007,13:42:37,1006338,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpStopRxTask() - SessionPtr = C062E25C
0,447,18-Jun-2007,13:42:37,1006338,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpStopRxTask()
0,448,18-Jun-2007,13:42:37,1006338,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpDisconnectTransmit(C062E25C)
0,449,18-Jun-2007,13:42:37,1006338,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpDisconnectTransmit()
0,452,18-Jun-2007,13:42:37,1006339,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpClose() Tx packets: 0 Rx packets: 0
0,453,18-Jun-2007,13:42:37,1006339,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpOpen()
0,454,18-Jun-2007,13:42:37,1006340,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpStartRxTask() - SessionPtr = C062E25C
0,455,18-Jun-2007,13:42:37,1006340,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpStartRxTask()
0,456,18-Jun-2007,13:42:37,1006340,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpOpen() - Session Ptr = c062e25c
0,457,18-Jun-2007,13:42:37,1006340,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpGetLocalAddress() RTP Port = 138A, RTCP Port = 138B
0,458,18-Jun-2007,13:42:37,1006340,tIPP_SiPSig,0xc0f04770,142,sg_sdp_parser.cpp,13,2,0,0 sg_sdpGetCodecType: Num algoritms = 2
0,459,18-Jun-2007,13:42:37,1006340,tIPP_SiPSig,0xc0f04770,188,sg_sdp_parser.cpp,13,2,0,0 Algorithm=0 eG711U; Payload=0
0,460,18-Jun-2007,13:42:37,1006340,tIPP_SiPSig,0xc0f04770,188,sg_sdp_parser.cpp,13,2,0,0 Algorithm=7 eG711A; Payload=8
0,461,18-Jun-2007,13:42:37,1006340,tIPP_SiPSig,0xc0f04770,190,sg_sdp_parser.cpp,13,2,0,0 Set: G711U=1, G711A=1, G723=0, G729=0, G722=0
0,462,18-Jun-2007,13:42:37,1006340,tIPP_SiPSig,0xc0f04770,142,sg_sdp_parser.cpp,13,2,0,0 sg_sdpGetCodecType: Num algoritms = 2
0,463,18-Jun-2007,13:42:37,1006340,tIPP_SiPSig,0xc0f04770,188,sg_sdp_parser.cpp,13,2,0,0 Algorithm=0 eG711U; Payload=0
0,464,18-Jun-2007,13:42:37,1006340,tIPP_SiPSig,0xc0f04770,188,sg_sdp_parser.cpp,13,2,0,0 Algorithm=7 eG711A; Payload=8
0,465,18-Jun-2007,13:42:37,1006340,tIPP_SiPSig,0xc0f04770,190,sg_sdp_parser.cpp,13,2,0,0 Set: G711U=1, G711A=1, G723=0, G729=0, G722=0
0,466,18-Jun-2007,13:42:37,1006340,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpSetRemoteAddress() IP = C0A80B42, Port = 3F4E
0,467,18-Jun-2007,13:42:37,1006340,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,!!!rtp_qdc_set_remote_address(rtpSessionPtr: c062e25c *UNKNOWN*): C0A80B42.16206
0,468,18-Jun-2007,13:42:37,1006340,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpSetRemoteAddress()
0,469,18-Jun-2007,13:42:37,1006340,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtcpSetRemoteAddress() IP = C0A80B42, Port = 3F4F0,47
0,18-Jun-2007,13:42:37,1006340,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtcpSetRemoteAddress()
0,471,18-Jun-2007,13:42:37,1006340,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpConnectReceive(C062E25C, 0, C0E09590)
0,472,18-Jun-2007,13:42:37,1006340,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpConnectTransmit(C062E25C)
0,473,18-Jun-2007,13:42:37,1006340,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpStartReceive() - ReceiveAudioType (2)
0,474,18-Jun-2007,13:42:37,1006340,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<=>rtp_qdc_set_remote_address(rtpSessionPtr: c062e25c): C0A80B42.16206
0,475,18-Jun-2007,13:42:37,1006340,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<=>rtp_qdc_set_local_address(rtpSessionPtr: c062e25c): C0A80BBC.5002
0,476,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpStartRxTask() - SessionPtr = C062E25C
0,477,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpStartRxTask()
0,478,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpStartReceive()
0,479,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpStartTransmit() - TransmitAudioType (2), PktSize (20)
0,480,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,Using silence suppression: 0
0,481,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpStartTransmit()
0,482,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,142,sg_sdp_parser.cpp,13,2,0,0 sg_sdpGetCodecType: Num algoritms = 2
0,483,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,188,sg_sdp_parser.cpp,13,2,0,0 Algorithm=0 eG711U; Payload=0
0,484,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,188,sg_sdp_parser.cpp,13,2,0,0 Algorithm=7 eG711A; Payload=8
0,485,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,190,sg_sdp_parser.cpp,13,2,0,0 Set: G711U=1, G711A=1, G723=0, G729=0, G722=0
0,486,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,142,sg_sdp_parser.cpp,13,2,0,0 sg_sdpGetCodecType: Num algoritms = 2
0,487,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,188,sg_sdp_parser.cpp,13,2,0,0 Algorithm=0 eG711U; Payload=0
0,488,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,188,sg_sdp_parser.cpp,13,2,0,0 Algorithm=7 eG711A; Payload=8
0,489,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,190,sg_sdp_parser.cpp,13,2,0,0 Set: G711U=1, G711A=1, G723=0, G729=0, G722=0
0,490,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,2760,sg_call_service_ext.cpp,13,2,0,0 INT->INT - sgNotifyDTMFCapability: DTMF in RTP will be used, Cref=2, Line ID=0:
0,491,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpSetRemoteAddress() IP = C0A80B42, Port = 3F4E
0,492,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<=>rtp_qdc_set_remote_address(rtpSessionPtr: c062e25c): C0A80B42.16206
0,493,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpSetRemoteAddress()
0,494,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtcpSetRemoteAddress() IP = C0A80B42, Port = 3F4F
0,495,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtcpSetRemoteAddress()
0,496,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpStartReceive() - ReceiveAudioType (2)
0,498,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpStartRxTask() - SessionPtr = C062E25C
0,499,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpStartRxTask()
0,500,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpStartReceive()
0,501,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpStartTransmit() - TransmitAudioType (2), PktSize (20)
0,502,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,Using silence suppression: 0
0,505,18-Jun-2007,13:42:37,1006341,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpStartTransmit()
0,506,18-Jun-2007,13:42:37,1006342,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpConnectReceive(C062E25C, 0, C0E09590)
0,508,18-Jun-2007,13:42:37,1006342,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpStartRxTask() - SessionPtr = C062E25C
0,509,18-Jun-2007,13:42:37,1006342,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpStartRxTask()
0,510,18-Jun-2007,13:42:37,1006342,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpConnectTransmit(C062E25C)
0,513,18-Jun-2007,13:42:37,1006342,tIPP_SiPSig,0xc0f04770,1207,sg_call_service_ext.cpp,13,2,0,0 SIG->CC: sgCallStateConnected, Cref=2, Line ID=0:
0,516,18-Jun-2007,13:42:37,1006342,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 H245 Complete Rx'd - call type is : 0
0,517,18-Jun-2007,13:42:37,1006342,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 E_CC_call_connected: S_CC_call_ringing_remote ---> S_CC_call_connected : 2 : 0
0,518,18-Jun-2007,13:42:37,1006342,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 E_FM_call_connected: S_FM_primary_call_wait_state ---> S_FM_primary_call_connected : : 0
0,554,18-Jun-2007,13:42:43,1006921,SipEngine,0xc0e69aa0,353,sg_events.cpp,13,2,0,0 STK->SIG TerminatedA: Cref=2, Line ID=0:
0,555,18-Jun-2007,13:42:43,1006922,SipEngine,0xc0e69aa0,9468,sg_engine.cpp,13,2,0,0 HandleCallTerminationFinal index =1
0,556,18-Jun-2007,13:42:43,1006922,SipEngine,0xc0e69aa0,11669,sg_engine.cpp,13,2,0,0 INT: DeleteCallContext, VecSize=0, Line ID=0: context=0xc0e03bb0
0,557,18-Jun-2007,13:42:43,1006922,tIPP_SiPSig,0xc0f04770,130,sg_parser.cpp,13,2,0,0 SIG->SIG: sipFarClear:
0,558,18-Jun-2007,13:42:43,1006922,tIPP_SiPSig,0xc0f04770,2041,sg_call_service_ext.cpp,13,2,0,0 SIG->CC: sgCallStateDisconnected, Cref=2, Line ID=0:
0,559,18-Jun-2007,13:42:43,1006923,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 E_CC_call_clear_remote: S_CC_call_connected ---> S_CC_call_cleared : 2 : 0
0,560,18-Jun-2007,13:42:43,1006923,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 UI_Notify: state = ccCallCleared, reason = Normal : 2 : 0
0,569,18-Jun-2007,13:42:43,1006924,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 ACT_call_cleared - : 2
0,571,18-Jun-2007,13:42:43,1006924,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 E_FM_call_cleared: S_FM_primary_call_connected ---> S_FM_primary_call_cleared : : 0
0,572,18-Jun-2007,13:42:43,1006924,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 CALLREC FREE = 2 : 0
0,573,18-Jun-2007,13:42:43,1006924,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 E_FM_wait: S_FM_primary_call_cleared ---> S_FM_idle : : 0
0,574,18-Jun-2007,13:42:43,1006924,tIPP_SiPSig,0xc0f04770,366,sg_parser.cpp,13,2,0,0 CC->SIG: sgCallDisconnectedConfirm:
0,575,18-Jun-2007,13:42:43,1006924,tIPP_SiPSig,0xc0f04770,3147,sg_engine.cpp,13,2,0,0 INT Eng::TerminateA, sigcref = 2, stackcref = 0, Line ID=0:
0,576,18-Jun-2007,13:42:43,1006924,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpClose(C062E25C)
0,577,18-Jun-2007,13:42:43,1006924,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpDisconnectReceive(C062E25C)
0,578,18-Jun-2007,13:42:43,1006924,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpDisconnectReceive()
0,579,18-Jun-2007,13:42:43,1006924,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpStopRxTask() - SessionPtr = C062E25C
0,580,18-Jun-2007,13:42:43,1006924,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpStopRxTask()
0,581,18-Jun-2007,13:42:43,1006924,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpDisconnectTransmit(C062E25C)
0,582,18-Jun-2007,13:42:43,1006924,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpDisconnectTransmit()
0,583,18-Jun-2007,13:42:43,1006925,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpClose() Tx packets: 290 Rx packets: 0
0,584,18-Jun-2007,13:42:43,1006925,tIPP_SiPSig,0xc0f04770,358,sg_callrec_mgr.cpp,13,2,0,0 INT: sgCallRec_Free: FREE cRef=2 numcRefs=0
0,585,18-Jun-2007,13:42:43,1006925,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 SendCallClearNotify - no dialog for line. Line: : 0
0,586,18-Jun-2007,13:42:43,1006925,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 SendCallClearNotify - no dialog for line. Line: : 0
0,587,18-Jun-2007,13:42:43,1006925,tRtpQdc,0xc0eeb4a0,213,rtp_component.c,15,4,0,(U)ObservationPeriodStat(0): NO VALID PACKETS WAS RECEIVED
0,588,18-Jun-2007,13:42:43,1006925,tRtpQdc,0xc0eeb4a0,213,rtp_component.c,15,4,0,!!!SPDAPI(Session=c062e25c, DspCh=SPDAPI_DSP_CHANNEL_NONE) !
0,589,18-Jun-2007,13:42:43,1006926,tRtpQdc,0xc0eeb4a0,213,rtp_component.c,15,4,0,[0] Mode 1 [bLastReport=1,nrOfPeriodsOneThrExc=1(RepsAcc=1)] Active=1 --> Send-Report=1
0,590,18-Jun-2007,13:42:43,1006926,tQDC,0xc0eec6c0,121,QDC.cpp,15,2,0,QDC msg queue rcv 144 bytes
0,591,18-Jun-2007,13:42:43,1006932,tQDC,0xc0eec6c0,92,CTransmitter_SIP.cpp,15,4,0,CTransmitter::SendTraps QcuAct=0 TrapAct=0 QcuIp=0 TrapIp=0 QcuPort=12010 TrapPort=162 TType=4
0,592,18-Jun-2007,13:42:43,1006932,tQDC,0xc0eec6c0,137,QDC.cpp,15,255,0,couldn't do report on QDC data<<<<<<<<<<<<< LATEST TRACE MESSAGE

was mich stutzig macht:

0,444,18-Jun-.....c05e0640): Invalid Session pointer parameter

klingt nicht sehr toll, weil ab da gehts dann mit den disconnect-anweisungen los


und hier die angerufene:

 ----------------------------------------------
Phone type: optiPoint 420 economy plus S
Application version: V6 R1.64.4
MAC address: 00:01:e3:28:ab:57
IP address: 192.168.11.189
Terminal no.: 1607
Time zone: 1
-----------------------------------------------
0,267,18-Jun-2007,13:42:46,1005571,SipEngine,0xc0e69ea0,10818,sg_engine.cpp,13,2,0,0 EXT: CreateCallContext, VecSize=1, Line ID=0: Context=0xc0e02da0
0,268,18-Jun-2007,13:42:46,1005571,SipEngine,0xc0e69ea0,775,Z:/ipphone1/application/sipsignalling/include/sg_engine_cfg.h,13,2,0,0 SIG->CC: SetConf_non_initiator() xxxxxxxxxxxxxxxxxxxxxxxxxxxxx 0 sig Cref 1008
0,269,18-Jun-2007,13:42:46,1005571,SipEngine,0xc0e69ea0,10921,sg_engine.cpp,13,2,0,0 EXT: CreateCallContext, CallId: 1
0,270,18-Jun-2007,13:42:46,1005574,SipEngine,0xc0e69ea0,230,sg_events.cpp,13,2,0,0 STK->SIG: sipNewCall: Cref=1008, ARef=0, Replaced Cref=0, Line ID=0:
0,271,18-Jun-2007,13:42:46,1005574,SipEngine,0xc0e69ea0,231,sg_events.cpp,13,2,0,0 STK->SIG: sipNewCall: alertURI=, info=, autoanswer=0, delay=0
0,272,18-Jun-2007,13:42:46,1005575,SipEngine,0xc0e69ea0,274,sg_events.cpp,13,2,0,0 STK->SIG: sipNewCall: R_Uname=6902, R_IP=192.168.11.66 R_Dname=Franz Haberfellner
0,273,18-Jun-2007,13:42:46,1005575,tIPP_SiPSig,0xc0f04770,120,sg_parser.cpp,13,2,0,0 SIG->SIG: sipNewCall:
0,274,18-Jun-2007,13:42:46,1005575,tIPP_SiPSig,0xc0f04770,325,sg_callrec_mgr.cpp,13,2,0,0 INT: sgCallRec_Alloc: ALLOC: cRef=1008, numcRefs=1
0,275,18-Jun-2007,13:42:46,1005575,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpOpen()
0,276,18-Jun-2007,13:42:46,1005575,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpStartRxTask() - SessionPtr = C062E25C
0,277,18-Jun-2007,13:42:46,1005575,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpStartRxTask()
0,278,18-Jun-2007,13:42:46,1005575,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpOpen() - Session Ptr = c062e25c
0,279,18-Jun-2007,13:42:46,1005575,tIPP_SiPSig,0xc0f04770,299,sg_call_service_ext.cpp,13,2,0,1008 sgCallNew About to send message to CC
0,280,18-Jun-2007,13:42:46,1005575,tIPP_SiPSig,0xc0f04770,300,sg_call_service_ext.cpp,13,2,0,0 sizeof(TsgCallStateIncoming)= 188
0,281,18-Jun-2007,13:42:46,1005575,tIPP_SiPSig,0xc0f04770,142,sg_sdp_parser.cpp,13,2,0,0 sg_sdpGetCodecType: Num algoritms = 2
0,282,18-Jun-2007,13:42:46,1005575,tIPP_SiPSig,0xc0f04770,188,sg_sdp_parser.cpp,13,2,0,0 Algorithm=0 eG711U; Payload=0
0,283,18-Jun-2007,13:42:46,1005575,tIPP_SiPSig,0xc0f04770,188,sg_sdp_parser.cpp,13,2,0,0 Algorithm=7 eG711A; Payload=8
0,284,18-Jun-2007,13:42:46,1005575,tIPP_SiPSig,0xc0f04770,190,sg_sdp_parser.cpp,13,2,0,0 Set: G711U=1, G711A=1, G723=0, G729=0, G722=0
0,285,18-Jun-2007,13:42:46,1005576,tIPP_SiPSig,0xc0f04770,142,sg_sdp_parser.cpp,13,2,0,0 sg_sdpGetCodecType: Num algoritms = 2
0,286,18-Jun-2007,13:42:46,1005576,tIPP_SiPSig,0xc0f04770,188,sg_sdp_parser.cpp,13,2,0,0 Algorithm=0 eG711U; Payload=0
0,287,18-Jun-2007,13:42:46,1005576,tIPP_SiPSig,0xc0f04770,188,sg_sdp_parser.cpp,13,2,0,0 Algorithm=7 eG711A; Payload=8
0,288,18-Jun-2007,13:42:46,1005576,tIPP_SiPSig,0xc0f04770,190,sg_sdp_parser.cpp,13,2,0,0 Set: G711U=1, G711A=1, G723=0, G729=0, G722=0
0,289,18-Jun-2007,13:42:46,1005576,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpSetRemoteAddress() IP = C0A80B42, Port = 281C
0,290,18-Jun-2007,13:42:46,1005576,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,!!!rtp_qdc_set_remote_address(rtpSessionPtr: c062e25c *UNKNOWN*): C0A80B42.10268
0,291,18-Jun-2007,13:42:46,1005576,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpSetRemoteAddress()
0,292,18-Jun-2007,13:42:46,1005576,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtcpSetRemoteAddress() IP = C0A80B42, Port = 281D
0,293,18-Jun-2007,13:42:46,1005576,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtcpSetRemoteAddress()
0,294,18-Jun-2007,13:42:46,1005576,tIPP_SiPSig,0xc0f04770,2760,sg_call_service_ext.cpp,13,2,0,0 INT->INT - sgNotifyDTMFCapability: DTMF in RTP will be used, Cref=1008, Line ID=0:
0,295,18-Jun-2007,13:42:46,1005577,tIPP_SiPSig,0xc0f04770,2125,sg_call_service_int.cpp,13,2,0,0 CODEC: g722_Try, Fallback: g723
0,296,18-Jun-2007,13:42:46,1005577,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 E_FM_new_incoming_call: S_FM_idle ---> S_FM_new_call_incoming : : 0
0,297,18-Jun-2007,13:42:46,1005577,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 CALLREC ALLOC = 1008 : 0
0,298,18-Jun-2007,13:42:46,1005577,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 E_CC_call_incoming: S_CC_idle ---> S_CC_call_incoming : 1008 : 0
0,299,18-Jun-2007,13:42:46,1005577,tIPP_SiPSig,0xc0f04770,352,sg_parser.cpp,13,2,0,0 CC->SIG: sgCallAcceptIC:
0,300,18-Jun-2007,13:42:46,1005578,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpGetLocalAddress() RTP Port = 138A, RTCP Port = 138B
0,301,18-Jun-2007,13:42:46,1005578,tIPP_SiPSig,0xc0f04770,720,sg_call_service_int.cpp,13,2,0,0 SIG->STK RingA: Cref=1008, Line ID=0:
0,303,18-Jun-2007,13:42:46,1005579,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 E_CC_wait: S_CC_call_incoming ---> S_CC_call_ringing_local : 1008 : 0
0,304,18-Jun-2007,13:42:46,1005579,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 E_FM_wait: S_FM_new_call_incoming ---> S_FM_primary_call_wait_state : : 0
0,305,18-Jun-2007,13:42:46,1005579,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 H245 Complete Rx'd - call type is : 0
0,310,18-Jun-2007,13:42:49,1005876,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 Failed to find callref for calltype: ccPreConsultationCall : 0
0,314,18-Jun-2007,13:42:49,1005877,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 E_FM_call_answer: S_FM_primary_call_wait_state ---> S_FM_primary_call_answer : : 0
0,315,18-Jun-2007,13:42:49,1005877,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 E_FML_FORCE: : S_FM_primary_call_answer ---> S_FM_primary_call_wait_state : : 0
0,316,18-Jun-2007,13:42:49,1005877,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 E_CC_call_answer: S_CC_call_ringing_local ---> S_CC_call_connecting : 1008 : 0
0,318,18-Jun-2007,13:42:49,1005877,tIPP_SiPSig,0xc0f04770,357,sg_parser.cpp,13,2,0,0 CC->SIG: sgCallConnect:
0,319,18-Jun-2007,13:42:49,1005877,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpGetLocalAddress() RTP Port = 138A, RTCP Port = 138B
0,320,18-Jun-2007,13:42:49,1005877,tIPP_SiPSig,0xc0f04770,845,sg_call_service_int.cpp,13,2,0,0 SIG->STK AcceptA: Cref=1008 Port=5002, Line ID=0:
0,321,18-Jun-2007,13:42:49,1005877,tIPP_SiPSig,0xc0f04770,872,sg_call_service_int.cpp,13,2,0,0 SIG->CC sgCallStateConnected: Cref=1008, Line ID=0:
0,322,18-Jun-2007,13:42:49,1005877,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpStartReceive() - ReceiveAudioType (2)
0,323,18-Jun-2007,13:42:49,1005877,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpStartReceive()
0,324,18-Jun-2007,13:42:49,1005877,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpStartTransmit() - TransmitAudioType (2), PktSize (20)
0,325,18-Jun-2007,13:42:49,1005877,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,Using silence suppression: 0
0,326,18-Jun-2007,13:42:49,1005877,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpStartTransmit()
0,327,18-Jun-2007,13:42:49,1005877,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpConnectReceive(C062E25C, 0, C0DFF240)
0,328,18-Jun-2007,13:42:49,1005877,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<=>rtp_qdc_set_remote_address(rtpSessionPtr: c062e25c): C0A80B42.10268
0,329,18-Jun-2007,13:42:49,1005877,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<=>rtp_qdc_set_local_address(rtpSessionPtr: c062e25c): C0A80BBD.5002
0,330,18-Jun-2007,13:42:49,1005877,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpStartRxTask() - SessionPtr = C062E25C
0,331,18-Jun-2007,13:42:49,1005877,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpStartRxTask()
0,332,18-Jun-2007,13:42:49,1005877,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpConnectTransmit(C062E25C)
0,333,18-Jun-2007,13:42:49,1005881,SipEngine,0xc0e69ea0,294,sg_events.cpp,13,2,0,0 STK->SIG AckReceivedA: Cref=1008 
0,334,18-Jun-2007,13:42:49,1005881,tIPP_SiPSig,0xc0f04770,228,sg_parser.cpp,13,2,0,0 SIG->SIG: sipAckReceived:
0,335,18-Jun-2007,13:42:49,1005881,tIPP_SiPSig,0xc0f04770,2094,sg_call_service_ext.cpp,13,2,0,0 STK->SIG - sgAckReceived with SDP for re-invite, Callref = 1008
0,336,18-Jun-2007,13:42:49,1005881,tIPP_SiPSig,0xc0f04770,306,sg_parser.cpp,13,2,0,0 SIG->SIG: sipRTPStatsUpdateDetails:
0,337,18-Jun-2007,13:42:49,1005881,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 UI_Notify: state = ccCallConnecting, reason = Normal : 1008 : 0
0,343,18-Jun-2007,13:42:49,1005882,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 E_CC_call_connected: S_CC_call_connecting ---> S_CC_call_connected : 1008 : 0
0,344,18-Jun-2007,13:42:49,1005882,tIPP_SiPSig,0xc0f04770,412,sg_parser.cpp,13,2,0,0 CC->SIG: sgRtpSourceChange:
0,345,18-Jun-2007,13:42:49,1005882,tIPP_SiPSig,0xc0f04770,1127,sg_call_service_int.cpp,13,2,0,0 SIG->RTP: mediaChange, Cref=1008 type=1
0,346,18-Jun-2007,13:42:49,1005882,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpConnectReceive(C062E25C, 0, C0DFF240)
0,348,18-Jun-2007,13:42:49,1005882,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpStartRxTask() - SessionPtr = C062E25C
0,349,18-Jun-2007,13:42:49,1005882,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpStartRxTask()
0,350,18-Jun-2007,13:42:49,1005882,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpConnectTransmit(C062E25C)
0,353,18-Jun-2007,13:42:49,1005882,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 E_FM_call_connected: S_FM_primary_call_wait_state ---> S_FM_primary_call_connected : : 0
0,354,18-Jun-2007,13:42:49,1005882,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 UI_Notify: state = ccCallConnected, reason = Normal : 1008 : 0
0,355,18-Jun-2007,13:42:49,1005884,t_dsp_tx_0,0xc0efaf40,213,rtp_component.c,15,4,0,RTP!!Txed packet!*********************************************
0,356,18-Jun-2007,13:42:49,1005886,t_dsp_tx_0,0xc0efaf40,213,rtp_component.c,15,4,0,RTP!!Txed packet!*********************************************
0,357,18-Jun-2007,13:42:49,1005888,t_dsp_tx_0,0xc0efaf40,213,rtp_component.c,15,4,0,RTP!!Txed packet!*********************************************
0,358,18-Jun-2007,13:42:49,1005890,t_dsp_tx_0,0xc0efaf40,213,rtp_component.c,15,4,0,RTP!!Txed packet!*********************************************
0,359,18-Jun-2007,13:42:50,1005967,tRtpQdc,0xc0eeb4a0,213,rtp_component.c,15,4,0,RtpQdc_FirstStart_Timers(0) was called 
0,364,18-Jun-2007,13:42:55,1006440,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 Failed to find callref for calltype: ccWaitingCall : 0
0,365,18-Jun-2007,13:42:55,1006440,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 Failed to find callref for calltype: ccConsultationCall : 0
0,366,18-Jun-2007,13:42:55,1006440,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 E_FM_call_clear_local: S_FM_primary_call_connected ---> S_FM_local_clear : : 0
0,367,18-Jun-2007,13:42:55,1006440,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 E_FML_FORCE: : S_FM_local_clear ---> S_FM_primary_call_wait_state : : 0
0,368,18-Jun-2007,13:42:55,1006440,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 E_CC_call_clear_local: S_CC_call_connected ---> S_CC_call_clearing : 1008 : 0
0,369,18-Jun-2007,13:42:55,1006440,tIPP_SiPSig,0xc0f04770,362,sg_parser.cpp,13,2,0,0 CC->SIG: sgCallDisconnect:
0,370,18-Jun-2007,13:42:55,1006440,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 UI_Notify: state = ccCallClearing, reason = Normal : 1008 : 0
0,371,18-Jun-2007,13:42:55,1006465,tIPP_SiPSig,0xc0f04770,2041,sg_call_service_ext.cpp,13,2,0,0 SIG->CC: sgCallStateDisconnected, Cref=1008, Line ID=0:
0,372,18-Jun-2007,13:42:55,1006465,tIPP_SiPSig,0xc0f04770,695,sg_call_service_int.cpp,13,2,0,0 SIG->STK: CallTerminateA: Cref=1008, Reason=Normal, Line ID=0:
0,373,18-Jun-2007,13:42:55,1006465,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 E_CC_call_clear_remote: S_CC_call_clearing ---> S_CC_call_cleared : 1008 : 0
0,374,18-Jun-2007,13:42:55,1006465,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 UI_Notify: state = ccCallCleared, reason = Normal : 1008 : 0
0,375,18-Jun-2007,13:42:55,1006465,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 ACT_call_cleared - : 1008
0,377,18-Jun-2007,13:42:55,1006465,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 E_FM_call_cleared: S_FM_primary_call_wait_state ---> S_FM_primary_call_cleared : : 0
0,378,18-Jun-2007,13:42:55,1006465,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 CALLREC FREE = 1008 : 0
0,379,18-Jun-2007,13:42:55,1006465,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 E_FM_wait: S_FM_primary_call_cleared ---> S_FM_idle : : 0
0,380,18-Jun-2007,13:42:55,1006465,tIPP_SiPSig,0xc0f04770,366,sg_parser.cpp,13,2,0,0 CC->SIG: sgCallDisconnectedConfirm:
0,381,18-Jun-2007,13:42:55,1006465,tIPP_SiPSig,0xc0f04770,3147,sg_engine.cpp,13,2,0,0 INT Eng::TerminateA, sigcref = 1008, stackcref = 1, Line ID=0:
0,382,18-Jun-2007,13:42:55,1006465,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpClose(C062E25C)
0,383,18-Jun-2007,13:42:55,1006465,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpDisconnectReceive(C062E25C)
0,384,18-Jun-2007,13:42:55,1006465,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpDisconnectReceive()
0,385,18-Jun-2007,13:42:55,1006465,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpStopRxTask() - SessionPtr = C062E25C
0,386,18-Jun-2007,13:42:55,1006465,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpStopRxTask()
0,387,18-Jun-2007,13:42:55,1006465,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,-->sgRtpDisconnectTransmit(C062E25C)
0,388,18-Jun-2007,13:42:55,1006465,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpDisconnectTransmit()
0,389,18-Jun-2007,13:42:55,1006466,SipEngine,0xc0e69ea0,3191,sg_engine.cpp,13,2,0,0 INT Eng::InternalTerminateA, stackcref = 1, Line ID=0:
0,390,18-Jun-2007,13:42:55,1006466,tIPP_SiPSig,0xc0f04770,213,rtp_component.c,15,4,0,<--sgRtpClose() Tx packets: 279 Rx packets: 0
0,391,18-Jun-2007,13:42:55,1006466,tIPP_SiPSig,0xc0f04770,358,sg_callrec_mgr.cpp,13,2,0,0 INT: sgCallRec_Free: FREE cRef=1008 numcRefs=0
0,392,18-Jun-2007,13:42:55,1006468,SipEngine,0xc0e69ea0,353,sg_events.cpp,13,2,0,0 STK->SIG TerminatedA: Cref=1008, Line ID=0:
0,393,18-Jun-2007,13:42:55,1006469,SipEngine,0xc0e69ea0,11669,sg_engine.cpp,13,2,0,0 INT: DeleteCallContext, VecSize=0, Line ID=0: context=0xc0e02da0
0,394,18-Jun-2007,13:42:55,1006469,tIPP_SiPSig,0xc0f04770,130,sg_parser.cpp,13,2,0,0 SIG->SIG: sipFarClear:
0,395,18-Jun-2007,13:42:55,1006469,tIPP_SiPSig,0xc0f04770,2016,sg_call_service_ext.cpp,13,2,0,0 INT sgFarCallClear - No call record allocated, Cref=1008, Line ID=0:
0,396,18-Jun-2007,13:42:55,1006469,tIPP_SiPSig,0xc0f04770,2024,sg_call_service_ext.cpp,13,2,0,0 SIG->CC: sgCallStateDisconnected, Cref=0, Line ID=0:
0,397,18-Jun-2007,13:42:55,1006469,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 SendCallClearNotify - no dialog for line. Line: : 0
0,398,18-Jun-2007,13:42:55,1006469,tSIPTelCtl,0xc0ea56b0,408,cc_tracing.c,12,4,0,0 SendCallClearNotify - no dialog for line. Line: : 0
0,399,18-Jun-2007,13:42:55,1006470,tRtpQdc,0xc0eeb4a0,213,rtp_component.c,15,4,0,(U)ObservationPeriodStat(0): NO VALID PACKETS WAS RECEIVED
0,400,18-Jun-2007,13:42:55,1006470,tRtpQdc,0xc0eeb4a0,213,rtp_component.c,15,4,0,!!!SPDAPI(Session=c062e25c, DspCh=SPDAPI_DSP_CHANNEL_NONE) !
0,401,18-Jun-2007,13:42:55,1006470,tRtpQdc,0xc0eeb4a0,213,rtp_component.c,15,4,0,[0] Mode 1 [bLastReport=1,nrOfPeriodsOneThrExc=1(RepsAcc=1)] Active=1 --> Send-Report=1
0,402,18-Jun-2007,13:42:55,1006470,tQDC,0xc0eec6c0,121,QDC.cpp,15,2,0,QDC msg queue rcv 144 bytes
0,403,18-Jun-2007,13:42:55,1006474,tQDC,0xc0eec6c0,92,CTransmitter_SIP.cpp,15,4,0,CTransmitter::SendTraps QcuAct=0 TrapAct=0 QcuIp=0 TrapIp=0 QcuPort=12010 TrapPort=162 TType=4
0,404,18-Jun-2007,13:42:55,1006474,tQDC,0xc0eec6c0,137,QDC.cpp,15,255,0,couldn't do report on QDC data<<<<<<<<<<<<< LATEST TRACE MESSAGE

 

[Dirk07]

Gateway IP Address unter SIP Details rausgenommen (kann mir wer sagen für was die is?)

Dort kommt z.B. ein SBC (SessionBorderController-Adresse) rein, wenn der Registrar und/oder Proxy nicht im normalen DNS aufgelöst werden kann. Dann schickt das Phone alle Anfragen unaufgelöst zu dieser Adresse. Die Adresse muss sich dnn um die Auflösung kümmern. Funktioniert aber nur im Zusammenhang mit dem Outbound-proxy ...


Einzurichten ist :

Ip-Addresse
Subnet und
Default-Route

Die TerminalNummer
und / oder TerminalName ( jenachdem wie du dich registrierst [Register by Name] )
Routing: Server
Outbound Proxy = OFF
Default OutboundDomain = <leer>
Sip-Server : IP vom Asterisk Port oder DNS-Name-sofern aufgelöst: 0 oder 5060
Port 0 = DNS-SRV
Registrar : IP vom Asterisk Port oder DNS-Name-sofern aufgelöst: 0 oder 5060
Port 0 = DNS-SRV
Gateway: 0.0.0.0
Sip-Port 5060
RTP Port :z.B: 5004
ServerTyp : Other
SIP Transport = UDP
SIP Realm = <realm vom Asterisk>
Sip-UserID = < steht in der sip.conf>
Sip-Passwort= < steht in der sip.conf>
...

Sprache : G711 preferrred
Compression G729
RTP Packet Size = Auto

Damit sollte es eigentlich funktionieren
Canreinvite auf NO = jedenfalls in der V1.4 ( evtl den Asterisk neu starten )

 

[Telephonmann]

hab ich alles so eingestellt,
das realm hat er automatisch auf asterisk gesetzt, da weiß ich jetzt auch nicht was das tut oder wo ich das finde auf dem server....
hast du ne idee?

achja ich hab da noch ne frage: was ist mit dem rtp port, kann da jedes tele dan selben benutzen? ich denke mal schon oder gibt es da gegenteilige erfahrungen?

 

[DeSan]

Kann jedes Phone den gleichen benutzen - macht ja auch Sinn bei Firewalls
Den Realm muss ich mal nachschauen, wo der noch einmal gesetzt war beim Server. Solange der Server bei der unauth. Msg. den Realm mitbringt, nehmen die Optie den mitgelieferten - der Realm gehört mit zur Authentifizierung. Wenn keiner übertragen wird, muss der im Phone eingetragene Realm 100% mit der der Anlage übereinstimmen.

Btw: Läuft es jetzt ?

 

[Telephonmann]

den realm hab ich jetzt in der sip.conf und auf den telefonen eingetragen.
am verhalten der telefone hat sich nichts verändert (-> es funkt noch immer nicht)

was mir heute aufgefallen ist:
wenn ich bei den optis canreinvite=no setzte, kann ich plötzlich von anderen ip-phones ein gespräch führen, untereinander haben sie füe einen bruchteil einer sekunde auch eine sprachverbindung bevor ich im asterisk dann seh dass sich einer mit "internal server error" verabschiedet.
umgekehrt geht dann wiederum kein gespräch über die isdn-karte
*verwirrung*

ich werd das morgen mal tracen und hier posten, viell kommen wir gemeinsam dahinter.

bis hierher schonmal danke an alle für die hilfe

 

[DeSan]

Poste doch einmal bitte dein generellen Teil deiner sip.conf und einmal den Teil von einer Rufnummer des Optie's. Vielleicht liegt dort der Hund begraben ...

Ach ja : hast du das Phone normal einrichtet oder liegt die Leitung auf einer Taste ?

 

[Telephonmann]

[general]
bindport = 5060
bindaddr = 192.168.11.66
context = default
realm=asterisk
srvlookup=yes
language=de
disallow=gsm

[1607]
callerid=6901
canreinvite=no
type=friend
secret=1234
host=dynamic
callgroup=4
pickupgroup=4
mailbox=1607

[1619]
callerid=6902
canreinvite=no
type=friend
secret=1234
host=dynamic
callgroup=4
pickupgroup=4
mailbox=1619

hier der auszug aus der sip.conf; zur info: die callerids brauch ich für die TK-Anlage hinter der ISDN karte.

@ DeSan: Entschuldige aber ich habe deine Frage nicht verstanden. Was meinst du mit Leitung auf einer Taste?

 

[Dirk07]

Hier einmal mein Auschnitt, wo die Phones ohne Probleme laufen ( V1.4 Asterisk )

################# sip.conf

[general]
context=default
bindaddr=192.168.0.44
bindport=5060
nat=yes
srvlookup=yes
language=de
dtmfmode=auto
;pedantic=yes
notifymimetype=text/plain
qualify=2000
disallow=all
allow=alaw

...


[7777]
callerid="Wesenberg" <7777>
host=dynamic
domain=192.168.0.44
user=7777
secret=7777
type=peer
mailbox=7777
nat=no
canreinvite=yes

...

############### extension.conf

;----------------------------------------------------------------------------

[lokal]

exten => _777X,1,NoCDR()
exten => _777X,n,Dial,SIP/${EXTEN}|55|Ttr
exten => _777x,n,Hangup
...


;--------------------------------------------------------------------------

[default]
include => lokal
...

 

[Telephonmann]

danke für die config, hab ich übernommen:

sip.conf:
[general]
context = default
bindaddr = 192.168.11.66
bindport = 5060
srvlookup=yes
nat=yes
language=de
dtmfmode=auto
;pedantic=yes
notifymimetype=text/plain
qualify=2000
disallow=all
allow=alaw

[1607]
callerid=6901
canreinvite=yes
type=peer
secret=1234
host=dynamic
domain=192.168.11.66
nat=no
callgroup=4
pickupgroup=4
mailbox=1607

[1619]
callerid=6902
canreinvite=yes
type=peer
secret=1234
host=dynamic
domain=192.168.11.66
nat=no
callgroup=4
pickupgroup=4
mailbox=1619

__________________________________

extensions.conf:

exten => 1607,1,NoCDR()
exten => 1607,n,Dial,SIP/${EXTEN}|30|Ttr
exten => 1607,n,voicemail,${EXTEN}
exten => 1619,1,NoCDR()
exten => 1619,n,Dial,SIP/${EXTEN}|30|Ttr
exten => 1619,n,voicemail,${EXTEN}

hat aber leider nichts geändert.
werde als nächstes testen ob es an der asterisk-version liegt