Gesprächsabbruch

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Gesprächsabbruch

Postby wasserhydrant » Mon Feb 28, 2011 2:02 pm

Hi

Ich beschäftige mich seit kurzer Zeit mit der 3CX und habe folgendes Problem.
Ich habe eine 3CX VOIP PBX mit Snom apparaten und habe folgendes Phänomen:

+) Ich baue einen Call auf (nach extern)
+) nach 20 Sekunden bricht mir das Audio auf meinem Apparat ab, der Call jedoch geht weiter
+) Auf der gegenseite wird der call automatisch beendet.

Anschließend versuche ich es erneut und dann gehts.

Hat jemand eine Idee was das sein kann?!

Vielen Dank im Voraus!!!
wasserhydrant
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Posts: 4
Joined: Mon Feb 28, 2011 1:58 pm

Re: Gesprächsabbruch

Postby groundhog » Mon Feb 28, 2011 2:35 pm

Stelle bitte mal die Protokollierung der 3CX auf umfassend und stelle mal das Protokoll eines solchen Gesprächst hier ein.
Veiser Gebäudetechnik GmbH
Ihr ITK Infrastruktur-Dienstleister im Rheinland
http://www.Veiser.de
groundhog
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3CX Valued Professional
 
Posts: 602
Joined: Sat Jul 04, 2009 10:49 am
Location: Neuss-Norf, Germany

Re: Gesprächsabbruch

Postby wasserhydrant » Mon Feb 28, 2011 2:44 pm

Hi

Hier das Gespräch:

Sent to udp:192.168.9.99:5060 at 28/2/2011 14:40:03:953 (1192 bytes):

INVITE sip:344@192.168.9.99;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.4.12:1026;branch=z9hG4bK-0gecjj8rystb;rport
From: "test" <sip:1378@192.168.9.99>;tag=545f3q5rg4
To: <sip:344@192.168.9.99;user=phone>
Call-ID: 3c2aa0e3dd57-aeod6li1h0oq
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:1378@192.168.4.12:1026;line=coojin6h>;reg-id=1
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom370/7.3.30
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 452

v=0
o=root 815747460 815747460 IN IP4 192.168.4.12
s=call
c=IN IP4 192.168.4.12
t=0 0
m=audio 57146 RTP/AVP 0 8 9 99 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:qoCawlV2sA6/IERKqG9hCxgbkDb1KMNLMqpQH6J8
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv



--------------------------------------------------------------------------------

Received from udp:192.168.9.99:5060 at 28/2/2011 14:40:04:239 (476 bytes):

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.4.12:1026;branch=z9hG4bK-0gecjj8rystb;rport=1026
Proxy-Authenticate: Digest nonce="414d535c037c36b256:fb291a490c7064fb4eaa2b85fc20d071",algorithm=MD5,realm="3CXPhoneSystem"
To: <sip:344@192.168.9.99;user=phone>;tag=6f654b59
From: "test"<sip:1378@192.168.9.99>;tag=545f3q5rg4
Call-ID: 3c2aa0e3dd57-aeod6li1h0oq
CSeq: 1 INVITE
User-Agent: 3CXPhoneSystem 9.0.15776.0
Content-Length: 0




--------------------------------------------------------------------------------

Sent to udp:192.168.9.99:5060 at 28/2/2011 14:40:04:246 (386 bytes):

ACK sip:344@192.168.9.99;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.4.12:1026;branch=z9hG4bK-0gecjj8rystb;rport
From: "test" <sip:1378@192.168.9.99>;tag=545f3q5rg4
To: <sip:344@192.168.9.99;user=phone>;tag=6f654b59
Call-ID: 3c2aa0e3dd57-aeod6li1h0oq
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:1378@192.168.4.12:1026;line=coojin6h>;reg-id=1
Content-Length: 0




--------------------------------------------------------------------------------

Sent to udp:192.168.9.99:5060 at 28/2/2011 14:40:04:259 (1416 bytes):

INVITE sip:344@192.168.9.99;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.4.12:1026;branch=z9hG4bK-tuebztdbe28d;rport
From: "test" <sip:1378@192.168.9.99>;tag=545f3q5rg4
To: <sip:344@192.168.9.99;user=phone>
Call-ID: 3c2aa0e3dd57-aeod6li1h0oq
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:1378@192.168.4.12:1026;line=coojin6h>;reg-id=1
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom370/7.3.30
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Authorization: Digest username="1378",realm="3CXPhoneSystem",nonce="414d535c037c36b256:fb291a490c7064fb4eaa2b85fc20d071",uri="sip:344@192.168.9.99;user=phone",response="57c395cc96123924a79334ea1b476a5b",algorithm=MD5
Content-Type: application/sdp
Content-Length: 452

v=0
o=root 815747460 815747460 IN IP4 192.168.4.12
s=call
c=IN IP4 192.168.4.12
t=0 0
m=audio 57146 RTP/AVP 0 8 9 99 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:qoCawlV2sA6/IERKqG9hCxgbkDb1KMNLMqpQH6J8
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv



--------------------------------------------------------------------------------

Received from udp:192.168.9.99:5060 at 28/2/2011 14:40:04:376 (276 bytes):

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.4.12:1026;branch=z9hG4bK-tuebztdbe28d;rport=1026
To: <sip:344@192.168.9.99;user=phone>
From: "test" <sip:1378@192.168.9.99>;tag=545f3q5rg4
Call-ID: 3c2aa0e3dd57-aeod6li1h0oq
CSeq: 2 INVITE
Content-Length: 0




--------------------------------------------------------------------------------

Received from udp:192.168.9.99:5060 at 28/2/2011 14:40:04:723 (735 bytes):

SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.4.12:1026;branch=z9hG4bK-tuebztdbe28d;rport=1026
Contact: <sip:344@192.168.9.99;user=phone>
To: <sip:344@192.168.9.99;user=phone>;tag=d043a60e
From: "test"<sip:1378@192.168.9.99>;tag=545f3q5rg4
Call-ID: 3c2aa0e3dd57-aeod6li1h0oq
CSeq: 2 INVITE
Content-Type: application/sdp
User-Agent: 3CXPhoneSystem 9.0.15776.0
Content-Length: 320

v=0
o=3cxPS 183291084800 503400366081 IN IP4 192.168.9.99
s=3cxPS Audio call
c=IN IP4 192.168.9.99
t=0 0
m=audio 7004 RTP/AVP 0 8 3 18 101
c=IN IP4 192.168.9.99
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv



--------------------------------------------------------------------------------

Received from udp:192.168.9.99:5060 at 28/2/2011 14:40:07:922 (829 bytes):

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.4.12:1026;branch=z9hG4bK-tuebztdbe28d;rport=1026
Contact: <sip:344@192.168.9.99:5060>
To: <sip:344@192.168.9.99;user=phone>;tag=d043a60e
From: "test"<sip:1378@192.168.9.99>;tag=545f3q5rg4
Call-ID: 3c2aa0e3dd57-aeod6li1h0oq
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhoneSystem 9.0.15776.0
Content-Length: 320

v=0
o=3cxPS 183291084800 503400366081 IN IP4 192.168.9.99
s=3cxPS Audio call
c=IN IP4 192.168.9.99
t=0 0
m=audio 7004 RTP/AVP 0 8 3 18 101
c=IN IP4 192.168.9.99
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv



--------------------------------------------------------------------------------

Sent to udp:192.168.9.99:5060 at 28/2/2011 14:40:07:951 (380 bytes):

ACK sip:344@192.168.9.99:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.12:1026;branch=z9hG4bK-lmycmbc2p959;rport
From: "test" <sip:1378@192.168.9.99>;tag=545f3q5rg4
To: <sip:344@192.168.9.99;user=phone>;tag=d043a60e
Call-ID: 3c2aa0e3dd57-aeod6li1h0oq
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:1378@192.168.4.12:1026;line=coojin6h>;reg-id=1
Content-Length: 0




--------------------------------------------------------------------------------

Sent to udp:192.168.9.99:5060 at 28/2/2011 14:40:32:166 (552 bytes):

BYE sip:344@192.168.9.99:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.12:1026;branch=z9hG4bK-1wrtq3x3hceg;rport
From: "test" <sip:1378@192.168.9.99>;tag=545f3q5rg4
To: <sip:344@192.168.9.99;user=phone>;tag=d043a60e
Call-ID: 3c2aa0e3dd57-aeod6li1h0oq
CSeq: 3 BYE
Max-Forwards: 70
Contact: <sip:1378@192.168.4.12:1026;line=coojin6h>;reg-id=1
User-Agent: snom370/7.3.30
RTP-RxStat: Total_Rx_Pkts=1360,Rx_Pkts=1360,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=1360,Tx_Pkts=1360,Remote_Tx_Pkts=0
Content-Length: 0




--------------------------------------------------------------------------------

Received from udp:192.168.9.99:5060 at 28/2/2011 14:40:32:305 (359 bytes):

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.4.12:1026;branch=z9hG4bK-1wrtq3x3hceg;rport=1026
Contact: <sip:344@192.168.9.99:5060>
To: <sip:344@192.168.9.99;user=phone>;tag=d043a60e
From: "test"<sip:1378@192.168.9.99>;tag=545f3q5rg4
Call-ID: 3c2aa0e3dd57-aeod6li1h0oq
CSeq: 3 BYE
User-Agent: 3CXPhoneSystem 9.0.15776.0
Content-Length: 0


Vielen Dank im Voraus!!
wasserhydrant
New User
 
Posts: 4
Joined: Mon Feb 28, 2011 1:58 pm

Re: Gesprächsabbruch

Postby wasserhydrant » Mon Feb 28, 2011 2:47 pm

also das obere war vom Apparat direkt aus dem SIP Trace

das hier ist aus der Anlage:

14:40:31.020 [CM503008]: Call(344): Call is terminated
14:40:17.192 Currently active calls - 1: [344]
14:40:06.770 Session 140224 of leg C:344.1 is confirmed
14:40:06.645 [CM503007]: Call(344): Device joined: sip:p004728.sil@sip.sil.at:5060
14:40:06.630 [CM503007]: Call(344): Device joined: sip:1378@192.168.4.12:1026;line=coojin6h
14:40:06.630 [MS210001] C:344.2:Answer received. RTP connection[unsecure]: 213.235.242.210:16906(16907)
14:40:06.630 Remote SDP is set for legC:344.2
14:40:03.442 [MS210003] C:344.1:Answer provided. Connection(transcoding mode[unsecure]):192.168.9.99:7004(7005)
14:40:03.442 [MS210001] C:344.2:Answer received. RTP connection[unsecure]: 213.235.242.210:16906(16907)
14:40:03.442 Remote SDP is set for legC:344.2
14:40:03.442 [CM505003]: Provider:[SILVERSERVER] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [silver.sip] PBX contact: [sip:p004728.sil@192.168.9.99:5060]
14:40:03.442 [CM503002]: Call(344): Alerting sip:p004728.sil@sip.sil.at:5060
14:40:03.145 [CM503025]: Call(344): Calling VoIPline:018174846344@(Ln.10006@SILVERSERVER)@[Dev:sip:p004728.sil@sip.sil.at:5060]
14:40:03.145 [MS210002] C:344.2:Offer provided. Connection(transcoding mode): 192.168.9.99:7006(7007)
14:40:03.099 [CM503004]: Call(344): Route 2: Unknown:344@(Ln.10002@LANCOM1724)@[Dev:sip:10002@192.168.1.3:8905]
14:40:03.099 [CM503004]: Call(344): Route 1: VoIPline:018174846344@(Ln.10006@SILVERSERVER)@[Dev:sip:p004728.sil@sip.sil.at:5060]
14:40:03.099 [CM503010]: Making route(s) to <sip:344@192.168.9.99;user=phone>
14:40:03.099 [MS210000] C:344.1:Offer received. RTP connection: 192.168.4.12:57146(57147)
14:40:03.099 Remote SDP is set for legC:344.1
14:40:03.099 [CM505001]: Ext.1378: Device info: Device Identified: [Man: Snom;Mod: 3xx series;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [snom370/7.3.30] PBX contact: [sip:1378@192.168.9.99:5060]
14:40:03.099 [CM503001]: Call(344): Incoming call from Ext.1378 to <sip:344@192.168.9.99;user=phone>
14:40:03.099 [CM500002]: Info on incoming INVITE:
INVITE sip:344@192.168.9.99;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.4.12:1026;branch=z9hG4bK-tuebztdbe28d;rport=1026
Max-Forwards: 70
Contact: <sip:1378@192.168.4.12:1026;line=coojin6h>;reg-id=1
To: <sip:344@192.168.9.99;user=phone>
From: "test"<sip:1378@192.168.9.99>;tag=545f3q5rg4
Call-ID: 3c2aa0e3dd57-aeod6li1h0oq
CSeq: 2 INVITE
Session-Expires: 3600;refresher=uas
Min-SE: 90
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Proxy-Authorization: Digest username="1378",realm="3CXPhoneSystem",nonce="414d535c037c36b256:fb291a490c7064fb4eaa2b85fc20d071",uri="sip:344@192.168.9.99;user=phone",response="57c395cc96123924a79334ea1b476a5b",algorithm=MD5
Supported: timer, 100rel, replaces, from-change
User-Agent: snom370/7.3.30
Allow-Events: talk, hold, refer, call-info
Content-Length: 0
P-Key-Flags: resolution="31x13", keys="4"
wasserhydrant
New User
 
Posts: 4
Joined: Mon Feb 28, 2011 1:58 pm

Re: Gesprächsabbruch

Postby groundhog » Tue Mar 01, 2011 9:35 am

Tritt das Problem wechselseitig auf?

An der Anlage scheint es zumindest auf den ersten Blick nicht zu liegen, denn die Vermittlung ist ja sauber erledigt worden. Keine Ahnung ob da noch was wegen der Lizenzkontrolle "gefummel" wird, aber der dann folgende RTP-Stream wird nur von den Endgeräten abgewickelt.

Was mir jetzt nicht so klar ist, ist die Aufteilung Deiner Netzwerke. Da sind ja mindestens 2 Netzwerke aktiv. Kannst Du vielleicht dazu etwas sagen?
Veiser Gebäudetechnik GmbH
Ihr ITK Infrastruktur-Dienstleister im Rheinland
http://www.Veiser.de
groundhog
3CX Valued Professional
3CX Valued Professional
 
Posts: 602
Joined: Sat Jul 04, 2009 10:49 am
Location: Neuss-Norf, Germany

Re: Gesprächsabbruch

Postby wasserhydrant » Tue Mar 01, 2011 10:40 am

Hey

Danke für die Antworten! Habe das Problem bereits gefunden. Es liegt am SIP Provider, mit einem anderen geht es einwandfrei.

LG
Steffen
wasserhydrant
New User
 
Posts: 4
Joined: Mon Feb 28, 2011 1:58 pm


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