[Gelöst] Keine ausgehenden Anrufe möglich

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robinsonR

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Ich habe eine Flynumber eingerichtet, mit Hinweisen von hier und hier. Eingehende Anrufe funktionieren auch; nur ausgehende Anrufe wollen nicht klappen. Hier mein SIP-Debug:
Code: 
pbx*CLI> 
Reliably Transmitting (no NAT) to 10.0.1.165:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.250:5060;branch=z9hG4bK60c70d61
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as795f81fd
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(11.5.0)
Date: Wed, 30 Oct 2013 11:12:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.0.1.165:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.250:5060;branch=z9hG4bK60c70d61
From: "Unknown" <sip:[email protected]>;tag=as795f81fd
To: <sip:[email protected]:5060>;tag=3049948853
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Contact: <sip:[email protected]:5060>
Supported: replaces
Allow-Events: message-summary, refer
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp,application/dtmf-relay,application/simple-message-summary,message/sipfrag
Accept-Encoding: identity
Accept-Language: en
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

<--- SIP read from UDP:10.0.1.165:5060 --->
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.1.165:5060;branch=z9hG4bKc588cfd17673d04fdd1cf56ee45dc229;rport
From: "Gigaset" <sip:[email protected]>;tag=420259779
To: <sip:[email protected];user=phone>
Call-ID: 612709245@10_0_1_165
CSeq: 2 INVITE
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
User-Agent: S450 IP/022270000000
Supported: replaces
Allow-Events: message-summary, refer
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 347

v=0
o=41445002909 5018 1 IN IP4 10.0.1.165
s=Mapping
c=IN IP4 10.0.1.165
t=0 0
m=audio 5018 RTP/AVP 0 8 96 97 2 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
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
<------------->
--- (14 headers 15 lines) ---
Sending to 10.0.1.165:5060 (NAT)
Sending to 10.0.1.165:5060 (NAT)
Using INVITE request as basis request - 612709245@10_0_1_165
Found peer '61' for '61' from 10.0.1.165:5060
  == Using SIP VIDEO TOS bits 136
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G726-32 for ID 96
Found audio description format AAL2-G726-32 for ID 97
Found audio description format G726-32 for ID 2
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g722|h263|h263p|h264), peer - audio=(ulaw|alaw|g726|g729|g726aal2)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.1.165:5018
Peer doesn't provide video
Looking for 01141xxxxxxxx in from-internal (domain pbx.meinedomain.com)
list_route: hop: <sip:[email protected]:5060>

<--- Transmitting (no NAT) to 10.0.1.165:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.1.165:5060;branch=z9hG4bKc588cfd17673d04fdd1cf56ee45dc229;received=10.0.1.165;rport=5060
From: "Gigaset" <sip:[email protected]>;tag=420259779
To: <sip:[email protected];user=phone>
Call-ID: 612709245@10_0_1_165
CSeq: 2 INVITE
Server: FPBX-2.10.1(11.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
    -- Executing [01141xxxxxxxx@from-internal:1] Macro("SIP/61-000000b3", "user-callerid,LIMIT,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/61-000000b3", "AMPUSER=61") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/61-000000b3", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/61-000000b3", "1?Set(REALCALLERIDNUM=61)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/61-000000b3", "AMPUSER=61") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/61-000000b3", "AMPUSERCIDNAME=Gigaset") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/61-000000b3", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/61-000000b3", "AMPUSERCID=61") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/61-000000b3", "CALLERID(all)="Gigaset" <61>") in new stack
    -- Executing [s@macro-user-callerid:9] GotoIf("SIP/61-000000b3", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:10] ExecIf("SIP/61-000000b3", "1?Set(GROUP(concurrency_limit)=61)") in new stack
    -- Executing [s@macro-user-callerid:11] GosubIf("SIP/61-000000b3", "7?sub-ccss,s,1(from-internal,01141xxxxxxxx)") in new stack
    -- Executing [s@sub-ccss:1] ExecIf("SIP/61-000000b3", "0?Return()") in new stack
    -- Executing [s@sub-ccss:2] Set("SIP/61-000000b3", "CCSS_SETUP=TRUE") in new stack
    -- Executing [s@sub-ccss:3] GosubIf("SIP/61-000000b3", "0?monitor_config,1(from-internal,01141xxxxxxxx):monitor_default,1(from-internal,01141xxxxxxxx)") in new stack
    -- Executing [monitor_default@sub-ccss:1] GotoIf("SIP/61-000000b3", "0?is_exten") in new stack
    -- Executing [monitor_default@sub-ccss:2] StackPop("SIP/61-000000b3", "") in new stack
    -- Executing [monitor_default@sub-ccss:3] Return("SIP/61-000000b3", "FALSE") in new stack
    -- Executing [s@macro-user-callerid:12] ExecIf("SIP/61-000000b3", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:13] GotoIf("SIP/61-000000b3", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,26)
    -- Executing [s@macro-user-callerid:26] Set("SIP/61-000000b3", "CALLERID(number)=61") in new stack
    -- Executing [s@macro-user-callerid:27] Set("SIP/61-000000b3", "CALLERID(name)=Gigaset") in new stack
    -- Executing [s@macro-user-callerid:28] Set("SIP/61-000000b3", "CHANNEL(language)=de") in new stack
    -- Executing [01141xxxxxxxx@from-internal:2] Set("SIP/61-000000b3", "MOHCLASS=default") in new stack
    -- Executing [01141xxxxxxxx@from-internal:3] Set("SIP/61-000000b3", "_NODEST=") in new stack
    -- Executing [01141xxxxxxxx@from-internal:4] Gosub("SIP/61-000000b3", "sub-record-check,s,1(out,01141xxxxxxxx,)") in new stack
    -- Executing [s@sub-record-check:1] GotoIf("SIP/61-000000b3", "1?check") in new stack
    -- Goto (sub-record-check,s,6)
    -- Executing [s@sub-record-check:6] Set("SIP/61-000000b3", "__MON_FMT=wav") in new stack
    -- Executing [s@sub-record-check:7] GotoIf("SIP/61-000000b3", "1?next") in new stack
    -- Goto (sub-record-check,s,10)
    -- Executing [s@sub-record-check:10] ExecIf("SIP/61-000000b3", "0?Return()") in new stack
    -- Executing [s@sub-record-check:11] GotoIf("SIP/61-000000b3", "0?out,1") in new stack
    -- Executing [s@sub-record-check:12] Set("SIP/61-000000b3", "__REC_STATUS=INITIALIZED") in new stack
    -- Executing [s@sub-record-check:13] ExecIf("SIP/61-000000b3", "0?Set(__REC_POLICY_MODE=)") in new stack
    -- Executing [s@sub-record-check:14] Set("SIP/61-000000b3", "NOW=1383131574") in new stack
    -- Executing [s@sub-record-check:15] Set("SIP/61-000000b3", "__DAY=30") in new stack
    -- Executing [s@sub-record-check:16] Set("SIP/61-000000b3", "__MONTH=10") in new stack
    -- Executing [s@sub-record-check:17] Set("SIP/61-000000b3", "__YEAR=2013") in new stack
    -- Executing [s@sub-record-check:18] Set("SIP/61-000000b3", "__TIMESTR=20131030-121254") in new stack
    -- Executing [s@sub-record-check:19] Set("SIP/61-000000b3", "__FROMEXTEN=61") in new stack
    -- Executing [s@sub-record-check:20] Set("SIP/61-000000b3", "__CALLFILENAME=out-01141xxxxxxxx-61-20131030-121254-1383131574.179") in new stack
    -- Executing [s@sub-record-check:21] Goto("SIP/61-000000b3", "out,1") in new stack
    -- Goto (sub-record-check,out,1)
    -- Executing [out@sub-record-check:1] ExecIf("SIP/61-000000b3", "1?Set(__REC_POLICY_MODE=dontcare)") in new stack
    -- Executing [out@sub-record-check:2] GosubIf("SIP/61-000000b3", "0?record,1(exten,01141xxxxxxxx,61)") in new stack
    -- Executing [out@sub-record-check:3] Return("SIP/61-000000b3", "") in new stack
    -- Executing [01141xxxxxxxx@from-internal:5] Macro("SIP/61-000000b3", "dialout-trunk,3,01141xxxxxxxx,") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/61-000000b3", "DIAL_TRUNK=3") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/61-000000b3", "0?sub-pincheck,s,1()") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/61-000000b3", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/61-000000b3", "DIAL_NUMBER=01141xxxxxxxx") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/61-000000b3", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/61-000000b3", "OUTBOUND_GROUP=OUT_3") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/61-000000b3", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/61-000000b3", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/61-000000b3", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/61-000000b3", "outbound-callerid,3") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/61-000000b3", "0?Set(CALLERPRES()=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/61-000000b3", "0?Set(REALCALLERIDNUM=61)") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/61-000000b3", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/61-000000b3", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/61-000000b3", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/61-000000b3", "TRUNKOUTCID=001xxxxxxx") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/61-000000b3", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/61-000000b3", "1?Set(CALLERID(all)=001xxxxxxx)") in new stack
    -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/61-000000b3", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/61-000000b3", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/61-000000b3", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
    -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/61-000000b3", "0?sub-flp-3,s,1()") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/61-000000b3", "OUTNUM=01141xxxxxxxx") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/61-000000b3", "custom=SIP/flynumber") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/61-000000b3", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
    -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/61-000000b3", "0?Set(DIAL_TRUNK_OPTIONS=M(confirm))") in new stack
    -- Executing [s@macro-dialout-trunk:17] Macro("SIP/61-000000b3", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/61-000000b3", "") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/61-000000b3", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/61-000000b3", "1?Set(CONNECTEDLINE(num,i)=01141xxxxxxxx)") in new stack
    -- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/61-000000b3", "1?Set(CONNECTEDLINE(name,i)=CID:001xxxxxxx)") in new stack
    -- Executing [s@macro-dialout-trunk:21] GotoIf("SIP/61-000000b3", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:22] Dial("SIP/61-000000b3", "SIP/flynumber/01141xxxxxxxx,300,") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 16056
Adding codec 100003 (ulaw) to SDP
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 46.19.209.10:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP mei.ne.ip.adresse:5060;branch=z9hG4bK2575e9e6;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as50c5b3aa
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: FPBX-2.10.1(11.5.0)
Date: Wed, 30 Oct 2013 11:12:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 248523058 248523058 IN IP4 mei.ne.ip.adresse
s=Asterisk PBX 11.5.0
c=IN IP4 mei.ne.ip.adresse
t=0 0
m=audio 16056 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/flynumber/01141xxxxxxxx
Retransmitting #1 (NAT) to 46.19.209.10:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP mei.ne.ip.adresse:5060;branch=z9hG4bK2575e9e6;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as50c5b3aa
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: FPBX-2.10.1(11.5.0)
Date: Wed, 30 Oct 2013 11:12:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 248523058 248523058 IN IP4 mei.ne.ip.adresse
s=Asterisk PBX 11.5.0
c=IN IP4 mei.ne.ip.adresse
t=0 0
m=audio 16056 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #2 (NAT) to 46.19.209.10:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP mei.ne.ip.adresse:5060;branch=z9hG4bK2575e9e6;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as50c5b3aa
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: FPBX-2.10.1(11.5.0)
Date: Wed, 30 Oct 2013 11:12:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 248523058 248523058 IN IP4 mei.ne.ip.adresse
s=Asterisk PBX 11.5.0
c=IN IP4 mei.ne.ip.adresse
t=0 0
m=audio 16056 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #3 (NAT) to 46.19.209.10:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP mei.ne.ip.adresse:5060;branch=z9hG4bK2575e9e6;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as50c5b3aa
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: FPBX-2.10.1(11.5.0)
Date: Wed, 30 Oct 2013 11:12:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 248523058 248523058 IN IP4 mei.ne.ip.adresse
s=Asterisk PBX 11.5.0
c=IN IP4 mei.ne.ip.adresse
t=0 0
m=audio 16056 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #4 (NAT) to 46.19.209.10:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP mei.ne.ip.adresse:5060;branch=z9hG4bK2575e9e6;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as50c5b3aa
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: FPBX-2.10.1(11.5.0)
Date: Wed, 30 Oct 2013 11:12:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 248523058 248523058 IN IP4 mei.ne.ip.adresse
s=Asterisk PBX 11.5.0
c=IN IP4 mei.ne.ip.adresse
t=0 0
m=audio 16056 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:10.0.1.165:5060 --->
REGISTER sip:pbx.meinedomain.com SIP/2.0
Via: SIP/2.0/UDP 10.0.1.165:5060;branch=z9hG4bKfe73c812786d3f2c94d27565a8851b1;rport
From: "Gigaset" <sip:[email protected]>;tag=3922554777
To: "Gigaset" <sip:[email protected]>
Call-ID: 2617431517@10_0_1_165
CSeq: 57 REGISTER
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
User-Agent: S450 IP/022270000000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 10.0.1.165:5060 (NAT)
Sending to 10.0.1.165:5060 (NAT)

<--- Transmitting (NAT) to 10.0.1.165:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.1.165:5060;branch=z9hG4bKfe73c812786d3f2c94d27565a8851b1;received=10.0.1.165;rport=5060
From: "Gigaset" <sip:[email protected]>;tag=3922554777
To: "Gigaset" <sip:[email protected]>;tag=as1c21d8fd
Call-ID: 2617431517@10_0_1_165
CSeq: 57 REGISTER
Server: FPBX-2.10.1(11.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1be374ca"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2617431517@10_0_1_165' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:10.0.1.165:5060 --->
REGISTER sip:pbx.meinedomain.com SIP/2.0
Via: SIP/2.0/UDP 10.0.1.165:5060;branch=z9hG4bKe7751f0ba915cbadfbaa7fcdfda9697c;rport
From: "Gigaset" <sip:[email protected]>;tag=3922554777
To: "Gigaset" <sip:[email protected]>
Call-ID: 2617431517@10_0_1_165
CSeq: 58 REGISTER
Contact: <sip:[email protected]:5060>
Authorization: Digest username="61", realm="asterisk", algorithm=MD5, uri="sip:pbx.meinedomain.com", nonce="1be374ca", response="269aa9dac2c7060c21ffd2edd1eb85e1"
Max-Forwards: 70
User-Agent: S450 IP/022270000000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 10.0.1.165:5060 (NAT)
Reliably Transmitting (no NAT) to 10.0.1.165:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.250:5060;branch=z9hG4bK22ab9c76
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as024ac867
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(11.5.0)
Date: Wed, 30 Oct 2013 11:13:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (NAT) to 10.0.1.165:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.165:5060;branch=z9hG4bKe7751f0ba915cbadfbaa7fcdfda9697c;received=10.0.1.165;rport=5060
From: "Gigaset" <sip:[email protected]>;tag=3922554777
To: "Gigaset" <sip:[email protected]>;tag=as1c21d8fd
Call-ID: 2617431517@10_0_1_165
CSeq: 58 REGISTER
Server: FPBX-2.10.1(11.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 180
Contact: <sip:[email protected]:5060>;expires=180
Date: Wed, 30 Oct 2013 11:13:02 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]:5060' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 10.0.1.165:5060:
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.250:5060;branch=z9hG4bK36d54e0a
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as4639c760
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 NOTIFY
User-Agent: FPBX-2.10.1(11.5.0)
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 85

Messages-Waiting: no
Message-Account: sip:*[email protected]
Voice-Message: 0/0 (0/0)

---
Scheduling destruction of SIP dialog '2617431517@10_0_1_165' in 32000 ms (Method: REGISTER)
Retransmitting #1 (no NAT) to 10.0.1.165:5060:
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.250:5060;branch=z9hG4bK36d54e0a
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as4639c760
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 NOTIFY
User-Agent: FPBX-2.10.1(11.5.0)
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 85

Messages-Waiting: no
Message-Account: sip:*[email protected]
Voice-Message: 0/0 (0/0)

---

<--- SIP read from UDP:10.0.1.165:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 10.0.1.250:5060;branch=z9hG4bK22ab9c76
From: "Unknown" <sip:[email protected]>;tag=as024ac867
To: <sip:[email protected]:5060>;tag=1438810799
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Contact: <sip:[email protected]:5060>
Allow-Events: message-summary, refer
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

<--- SIP read from UDP:10.0.1.165:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.250:5060;branch=z9hG4bK36d54e0a
From: "Unknown" <sip:[email protected]>;tag=as4639c760
To: <sip:[email protected]:5060>;tag=2148649155
Call-ID: [email protected]:5060
CSeq: 102 NOTIFY
Contact: <sip:[email protected]:5060>
Supported: replaces
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: NOTIFY

<--- SIP read from UDP:10.0.1.165:5060 --->
CANCEL sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.1.165:5060;branch=z9hG4bKc588cfd17673d04fdd1cf56ee45dc229;rport
From: "Gigaset" <sip:[email protected]>;tag=420259779
To: <sip:[email protected];user=phone>
Call-ID: 612709245@10_0_1_165
CSeq: 2 CANCEL
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
User-Agent: S450 IP/022270000000
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 10.0.1.165:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 10.0.1.165:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.0.1.165:5060;branch=z9hG4bKc588cfd17673d04fdd1cf56ee45dc229;received=10.0.1.165;rport=5060
From: "Gigaset" <sip:[email protected]>;tag=420259779
To: <sip:[email protected];user=phone>;tag=as46837400
Call-ID: 612709245@10_0_1_165
CSeq: 2 INVITE
Server: FPBX-2.10.1(11.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 10.0.1.165:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.165:5060;branch=z9hG4bKc588cfd17673d04fdd1cf56ee45dc229;received=10.0.1.165;rport=5060
From: "Gigaset" <sip:[email protected]>;tag=420259779
To: <sip:[email protected];user=phone>;tag=as46837400
Call-ID: 612709245@10_0_1_165
CSeq: 2 CANCEL
Server: FPBX-2.10.1(11.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
  == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/61-000000b3' in macro 'dialout-trunk'
  == Spawn extension (from-internal, 01141xxxxxxxx, 5) exited non-zero on 'SIP/61-000000b3'
    -- Executing [h@from-internal:1] Hangup("SIP/61-000000b3", "") in new stack
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/61-000000b3'
Retransmitting #1 (no NAT) to 10.0.1.165:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.0.1.165:5060;branch=z9hG4bKc588cfd17673d04fdd1cf56ee45dc229;received=10.0.1.165;rport=5060
From: "Gigaset" <sip:[email protected]>;tag=420259779
To: <sip:[email protected];user=phone>;tag=as46837400
Call-ID: 612709245@10_0_1_165
CSeq: 2 INVITE
Server: FPBX-2.10.1(11.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.0.1.165:5060 --->
ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.1.165:5060;branch=z9hG4bKc588cfd17673d04fdd1cf56ee45dc229;rport
From: "Gigaset" <sip:[email protected]>;tag=420259779
To: <sip:[email protected];user=phone>;tag=as46837400
Call-ID: 612709245@10_0_1_165
CSeq: 2 ACK
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
User-Agent: S450 IP/022270000000
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '612709245@10_0_1_165' Method: ACK

<--- SIP read from UDP:10.0.1.165:5060 --->
ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.1.165:5060;branch=z9hG4bKc588cfd17673d04fdd1cf56ee45dc229;rport
From: "Gigaset" <sip:[email protected]>;tag=420259779
To: <sip:[email protected];user=phone>;tag=as46837400
Call-ID: 612709245@10_0_1_165
CSeq: 2 ACK
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
User-Agent: S450 IP/022270000000
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
pbx*CLI> sip set debug off
SIP Debugging Disabled
pbx*CLI>
Wo liegt der Hund begraben?

Der Provider hat Konfigurationen umgestellt. Damit sind die oben angeführten Tipps obsolet. Jetzt funktioniert alles so wie es sollte. Ist sehr zu empfehlen.
 
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