SIP -> ISDN | Abgehende Rufe werden beim Abheben automatisch beendet

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JayKa

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Gestern habe ich mir TrixBox das erste mal installiert. Nach einer weile hat dann das nötigste Funktioniert. Also vor allem ankommende und abgehende Rufe gingen Problemlos.
Seit heute kann ich keinen Ruf mehr tätigen. Weder intern noch extern.

System:
Lancom 1823 als ISDN-Gateway an einem ISDN-Anlagen-Anschluss
Asterix mit TrixBox und FreePBX als TK-Anlage
snom[1] 360 als abgehendes Telefon
weiteres snom[2] 360 als annehmendes Telefon

Also folgender Problemverlauf als Beispiel:

mit snom[1] wähle ich die externe Nummer des snom[2]. Das Klingeln funktioniert korrekt.
Sobald ich mit snom[2] das Gespräch annehme, wird die Verbindung für eine Sekunde aufgebaut (Connected) und gleich drauf von alleine abgebaut (Disconnected) im Display des snom[1] erscheind kurz darauf (Declined)

Dieses Problem bestand gestern noch nicht. Ich weiß aber leider auch nicht, was ich seit dem verstellt habe.
Es tritt auch unabhängig davon auf, wen ich anrufe. Also intern, extern oder auch ganz andere/fremde Nummern.

Folgenden Trace habe ich mit meinem snom[1] aufgenommen:
Code: 
Received from udp:192.168.0.2:5060 at 1/3/2007 18:00:11:040 (504 bytes):

OPTIONS sip:[email protected]:2051;line=nuqrdfij SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK6a8797a9;rport
From: "Unknown" <sip:[email protected]>;tag=as1edeb4c1
To: <sip:[email protected]:2051;line=nuqrdfij>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 01 Mar 2007 17:00:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


Sent to udp:192.168.0.2:5060 at 1/3/2007 18:00:11:050 (587 bytes):

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK6a8797a9;rport=5060
From: "Unknown" <sip:[email protected]>;tag=as1edeb4c1
To: <sip:[email protected]:2051;line=nuqrdfij>
Call-ID: [email protected]
CSeq: 102 OPTIONS
Contact: <sip:[email protected]:2051;line=nuqrdfij>;flow-id=1
User-Agent: snom360/6.5.2
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Content-Length: 0


Sent to udp:192.168.0.2:5060 at 1/3/2007 18:00:23:660 (1209 bytes):

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.88:2051;branch=z9hG4bK-fj6tubzf80fq;rport
From: "Julian" <sip:[email protected]>;tag=d8uqlf2aq0
To: <sip:[email protected];user=phone>
Call-ID: 3c2c06ed8165-s94voadrzn0f@snom360-000413236880
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:[email protected]:2051;line=nuqrdfij>;flow-id=1
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom360/6.5.2
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 475

v=0
o=root 1981643683 1981643683 IN IP4 192.168.0.88
s=call
c=IN IP4 192.168.0.88
t=0 0
m=audio 57218 RTP/AVP 0 8 9 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:cTP13uOiXi6vUhI1lD4hvxUvll7+dVGmfvfywc5t
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 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=encryption:optional
a=sendrecv

Received from udp:192.168.0.2:5060 at 1/3/2007 18:00:23:750 (515 bytes):

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.88:2051;branch=z9hG4bK-fj6tubzf80fq;received=192.168.0.88;rport=2051
From: "Julian" <sip:[email protected]>;tag=d8uqlf2aq0
To: <sip:[email protected];user=phone>;tag=as3024370c
Call-ID: 3c2c06ed8165-s94voadrzn0f@snom360-000413236880
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5e56dba9"
Content-Length: 0


Sent to udp:192.168.0.2:5060 at 1/3/2007 18:00:23:750 (397 bytes):

ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.88:2051;branch=z9hG4bK-fj6tubzf80fq;rport
From: "Julian" <sip:[email protected]>;tag=d8uqlf2aq0
To: <sip:[email protected];user=phone>;tag=as3024370c
Call-ID: 3c2c06ed8165-s94voadrzn0f@snom360-000413236880
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:[email protected]:2051;line=nuqrdfij>;flow-id=1
Content-Length: 0


Sent to udp:192.168.0.2:5060 at 1/3/2007 18:00:23:760 (1385 bytes):

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.88:2051;branch=z9hG4bK-b0szicv0g1sb;rport
From: "Julian" <sip:[email protected]>;tag=d8uqlf2aq0
To: <sip:[email protected];user=phone>
Call-ID: 3c2c06ed8165-s94voadrzn0f@snom360-000413236880
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:[email protected]:2051;line=nuqrdfij>;flow-id=1
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom360/6.5.2
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Authorization: Digest username="88",realm="asterisk",nonce="5e56dba9",uri="sip:[email protected];user=phone",response="738a9f190547f51bb6452ee40b8fb76e",algorithm=md5
Content-Type: application/sdp
Content-Length: 475

v=0
o=root 1981643683 1981643683 IN IP4 192.168.0.88
s=call
c=IN IP4 192.168.0.88
t=0 0
m=audio 57218 RTP/AVP 0 8 9 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:cTP13uOiXi6vUhI1lD4hvxUvll7+dVGmfvfywc5t
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 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=encryption:optional
a=sendrecv

Received from udp:192.168.0.2:5060 at 1/3/2007 18:00:23:830 (435 bytes):

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.88:2051;branch=z9hG4bK-b0szicv0g1sb;received=192.168.0.88;rport=2051
From: "Julian" <sip:[email protected]>;tag=d8uqlf2aq0
To: <sip:[email protected];user=phone>
Call-ID: 3c2c06ed8165-s94voadrzn0f@snom360-000413236880
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0


Received from udp:192.168.0.2:5060 at 1/3/2007 18:00:24:640 (850 bytes):

SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.88:2051;branch=z9hG4bK-b0szicv0g1sb;received=192.168.0.88;rport=2051
From: "Julian" <sip:[email protected]>;tag=d8uqlf2aq0
To: <sip:[email protected];user=phone>;tag=as6de8523a
Call-ID: 3c2c06ed8165-s94voadrzn0f@snom360-000413236880
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 357

v=0
o=root 2684 2684 IN IP4 192.168.0.2
s=session
c=IN IP4 192.168.0.2
t=0 0
m=audio 12212 RTP/AVP 4 3 0 8 2 18 101
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

Received from udp:192.168.0.2:5060 at 1/3/2007 18:00:25:620 (451 bytes):

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.88:2051;branch=z9hG4bK-b0szicv0g1sb;received=192.168.0.88;rport=2051
From: "Julian" <sip:[email protected]>;tag=d8uqlf2aq0
To: <sip:[email protected];user=phone>;tag=as6de8523a
Call-ID: 3c2c06ed8165-s94voadrzn0f@snom360-000413236880
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0


Received from udp:192.168.0.2:5060 at 1/3/2007 18:00:34:910 (452 bytes):

SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.0.88:2051;branch=z9hG4bK-b0szicv0g1sb;received=192.168.0.88;rport=2051
From: "Julian" <sip:[email protected]>;tag=d8uqlf2aq0
To: <sip:[email protected];user=phone>;tag=as6de8523a
Call-ID: 3c2c06ed8165-s94voadrzn0f@snom360-000413236880
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0


Sent to udp:192.168.0.2:5060 at 1/3/2007 18:00:34:920 (397 bytes):

ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.88:2051;branch=z9hG4bK-b0szicv0g1sb;rport
From: "Julian" <sip:[email protected]>;tag=d8uqlf2aq0
To: <sip:[email protected];user=phone>;tag=as6de8523a
Call-ID: 3c2c06ed8165-s94voadrzn0f@snom360-000413236880
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:[email protected]:2051;line=nuqrdfij>;flow-id=1
Content-Length: 0

und mit meinem snom[2]:

Code: 
Received from udp:192.168.0.2:5060 at 1/3/2007 18:00:11:230 (504 bytes):

OPTIONS sip:[email protected]:2054;line=y4mfi3nr SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK6b8abde4;rport
From: "Unknown" <sip:[email protected]>;tag=as5eadbaef
To: <sip:[email protected]:2054;line=y4mfi3nr>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 01 Mar 2007 17:00:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


Sent to udp:192.168.0.2:5060 at 1/3/2007 18:00:11:240 (587 bytes):

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK6b8abde4;rport=5060
From: "Unknown" <sip:[email protected]>;tag=as5eadbaef
To: <sip:[email protected]:2054;line=y4mfi3nr>
Call-ID: [email protected]
CSeq: 102 OPTIONS
Contact: <sip:[email protected]:2054;line=y4mfi3nr>;flow-id=1
User-Agent: snom360/6.5.2
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Content-Length: 0


Received from udp:192.168.0.2:5060 at 1/3/2007 18:00:25:750 (1035 bytes):

INVITE sip:[email protected]:2054;line=y4mfi3nr SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK6fddee64;rport
From: "09132782888" <sip:[email protected]>;tag=as43260ac3
To: <sip:[email protected]:2054;line=y4mfi3nr>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 01 Mar 2007 17:00:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 488

v=0
o=root 2684 2684 IN IP4 192.168.0.2
s=session
c=IN IP4 192.168.0.2
t=0 0
m=audio 19426 RTP/AVP 0 4 3 8 111 5 10 7 18 110 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

Sent to udp:192.168.0.2:5060 at 1/3/2007 18:00:25:790 (495 bytes):

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK6fddee64;rport=5060
From: "09132782888" <sip:[email protected]>;tag=as43260ac3
To: <sip:[email protected]:2054;line=y4mfi3nr>;tag=8wanpksykc
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:2054;line=y4mfi3nr>;flow-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Content-Length: 0


Received from udp:192.168.0.2:5060 at 1/3/2007 18:00:25:980 (1035 bytes):

INVITE sip:[email protected]:2054;line=y4mfi3nr SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK6fddee64;rport
From: "09132782888" <sip:[email protected]>;tag=as43260ac3
To: <sip:[email protected]:2054;line=y4mfi3nr>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 01 Mar 2007 17:00:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 488

v=0
o=root 2684 2684 IN IP4 192.168.0.2
s=session
c=IN IP4 192.168.0.2
t=0 0
m=audio 19426 RTP/AVP 0 4 3 8 111 5 10 7 18 110 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

Sent to udp:192.168.0.2:5060 at 1/3/2007 18:00:25:980 (495 bytes):

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK6fddee64;rport=5060
From: "09132782888" <sip:[email protected]>;tag=as43260ac3
To: <sip:[email protected]:2054;line=y4mfi3nr>;tag=8wanpksykc
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:2054;line=y4mfi3nr>;flow-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Content-Length: 0


Sent to udp:192.168.0.2:5060 at 1/3/2007 18:00:26:290 (495 bytes):

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK6fddee64;rport=5060
From: "09132782888" <sip:[email protected]>;tag=as43260ac3
To: <sip:[email protected]:2054;line=y4mfi3nr>;tag=8wanpksykc
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:2054;line=y4mfi3nr>;flow-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Content-Length: 0


Sent to udp:192.168.0.2:5060 at 1/3/2007 18:00:27:300 (495 bytes):

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK6fddee64;rport=5060
From: "09132782888" <sip:[email protected]>;tag=as43260ac3
To: <sip:[email protected]:2054;line=y4mfi3nr>;tag=8wanpksykc
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:2054;line=y4mfi3nr>;flow-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Content-Length: 0


Sent to udp:192.168.0.2:5060 at 1/3/2007 18:00:29:300 (495 bytes):

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK6fddee64;rport=5060
From: "09132782888" <sip:[email protected]>;tag=as43260ac3
To: <sip:[email protected]:2054;line=y4mfi3nr>;tag=8wanpksykc
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:2054;line=y4mfi3nr>;flow-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Content-Length: 0


Sent to udp:192.168.0.2:5060 at 1/3/2007 18:00:29:900 (909 bytes):

SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK6fddee64;rport=5060
From: "09132782888" <sip:[email protected]>;tag=as43260ac3
To: <sip:[email protected]:2054;line=y4mfi3nr>;tag=8wanpksykc
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:2054;line=y4mfi3nr>;flow-id=1
User-Agent: snom360/6.5.2
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 600460347 600460348 IN IP4 192.168.0.98
s=call
c=IN IP4 192.168.0.98
t=0 0
m=audio 55006 RTP/AVP 0 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:+nzkNWSXYn0yePMbxb7x5jwhjF32xeohz/uv1rhS
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=encryption:optional
a=sendrecv

Received from udp:192.168.0.2:5060 at 1/3/2007 18:00:30:090 (418 bytes):

ACK sip:[email protected]:2054;line=y4mfi3nr SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK6f30e770;rport
From: "09132782888" <sip:[email protected]>;tag=as43260ac3
To: <sip:[email protected]:2054;line=y4mfi3nr>;tag=8wanpksykc
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


Received from udp:192.168.0.2:5060 at 1/3/2007 18:00:30:340 (418 bytes):

BYE sip:[email protected]:2054;line=y4mfi3nr SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK2f90fe38;rport
From: "09132782888" <sip:[email protected]>;tag=as43260ac3
To: <sip:[email protected]:2054;line=y4mfi3nr>;tag=8wanpksykc
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


Received from udp:192.168.0.2:5060 at 1/3/2007 18:00:30:810 (418 bytes):

BYE sip:[email protected]:2054;line=y4mfi3nr SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK2f90fe38;rport
From: "09132782888" <sip:[email protected]>;tag=as43260ac3
To: <sip:[email protected]:2054;line=y4mfi3nr>;tag=8wanpksykc
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


Received from udp:192.168.0.2:5060 at 1/3/2007 18:00:30:810 (418 bytes):

BYE sip:[email protected]:2054;line=y4mfi3nr SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK2f90fe38;rport
From: "09132782888" <sip:[email protected]>;tag=as43260ac3
To: <sip:[email protected]:2054;line=y4mfi3nr>;tag=8wanpksykc
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


Sent to udp:192.168.0.2:5060 at 1/3/2007 18:00:30:840 (527 bytes):

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK2f90fe38;rport=5060
From: "09132782888" <sip:[email protected]>;tag=as43260ac3
To: <sip:[email protected]:2054;line=y4mfi3nr>;tag=8wanpksykc
Call-ID: [email protected]
CSeq: 103 BYE
Contact: <sip:[email protected]:2054;line=y4mfi3nr>;flow-id=1
User-Agent: snom360/6.5.2
RTP-RxStat: Total_Rx_Pkts=18,Rx_Pkts=18,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=17,Tx_Pkts=17,Remote_Tx_Pkts=0
Content-Length: 0

Ich hoffe ihr könnt mir weiterhelfen, hänge schon den ganzen Tag an dem Problem fest.
 
Gelöst :)

Habe das Problem gelöst.
Ich weiß nicht genau an was es lag. Aber jetzt habe ich von FreePBX ein upgrade von 2.2.0rc3 auf 2.2.1 durchgeführt und seit dem läuft wieder alles perfekt :)
Woran es jetzt genau lag, weiß ich leider nicht :(
Meint ihr es könnte an unkompilierten Änderungen in den config-Dateien liegen?
 
Um antworten zu können musst du eingeloggt sein.

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