[Problem] Keine ausgehenden Gespräche möglich

robinsonR

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Seit einiger Zeit habe ich eine Nummer von www.flynumber.com. Bis vor kurzem ging alles tipptopp. Jetzt kann ich plötzlich nicht mehr raus telefonieren. Eingehende Anrufe kommen aber an. Über ein Softphone (Jitsi) klappt die Telefoniererei zwar, aber ich möchte ja meinen Asterisk verwenden. Leider kann ich nicht sagen, was sich an der Konfiguration geändert hat, seit es nicht mehr klappt. Meiner Meinung nach eigentlich nichts. ;-)
Leider konnte mir der Support nicht weiterhelfen, darum frage ich hier mal nach, ob jemand aus dem Debug schlau wird.
Code:
Audio is at 9332
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 100019 (slin) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to [Provider-IP]:5060:
INVITE sip:[angerufene Nummer]@[SIP-Provider] SIP/2.0
Via: SIP/2.0/UDP [meine ext. IP]:5060;branch=z9hG4bK6b28541a;rport
Max-Forwards: 70
From: "[ICH]" <sip:[ausgehende Nummer]@[SIP-Provider]>;tag=as7ac3cb44
To: <sip:[angerufene Nummer]@[SIP-Provider]>
Contact: <sip:[ausgehende Nummer]@[meine ext. IP]:5060>
Call-ID: 13b5b7[EXT][email protected][SIP-Provider]
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.5.0)
Date: Tue, 02 Sep 2014 00:44:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326

v=0
o=root 546951438 546951438 IN IP4 [meine ext. IP]
s=Asterisk PBX 11.5.0
c=IN IP4 [meine ext. IP]
t=0 0
m=audio 9332 RTP/AVP 0 8 9 10 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:10 L16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/flynumber/[angerufene Nummer]

<--- SIP read from UDP:[Provider-IP]:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP [meine ext. IP]:5060;branch=z9hG4bK6b28541a;rport=5060
From: "[ICH]" <sip:[ausgehende Nummer]@[SIP-Provider]>;tag=as7ac3cb44
To: <sip:[angerufene Nummer]@[SIP-Provider]>;tag=88f8d25d306df673868d5fcf[EXT]d8252e.ba38
Call-ID: [email protected][SIP-Provider]
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="[SIP-Provider]", nonce="****************"
Server: hedgehog v6.8p6332
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to [Provider-IP]:5060:
ACK sip:[angerufene Nummer]@[SIP-Provider] SIP/2.0
Via: SIP/2.0/UDP [meine ext. IP]:5060;branch=z9hG4bK6b28541a;rport
Max-Forwards: 70
From: "[ICH]" <sip:[ausgehende Nummer]@[SIP-Provider]>;tag=as7ac3cb44
To: <sip:[angerufene Nummer]@[SIP-Provider]>;tag=88f8d25d306df673868d5fcf[EXT]d8252e.ba38
Contact: <sip:[ausgehende Nummer]@[meine ext. IP]:5060>
Call-ID: 13b5b7[EXT][email protected][SIP-Provider]
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.5.0)
Content-Length: 0


---
Audio is at 9332
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 100019 (slin) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to [Provider-IP]:5060:
INVITE sip:[angerufene Nummer]@[SIP-Provider] SIP/2.0
Via: SIP/2.0/UDP [meine ext. IP]:5060;branch=z9hG4bK40f407ae;rport
Max-Forwards: 70
From: "[ICH]" <sip:[ausgehende Nummer]@[SIP-Provider]>;tag=as7ac3cb44
To: <sip:[angerufene Nummer]@[SIP-Provider]>
Contact: <sip:[ausgehende Nummer]@[meine ext. IP]:5060>
Call-ID: 13b5b7[EXT][email protected][SIP-Provider]
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0(11.5.0)
Proxy-Authorization: Digest username="[username]", realm="[SIP-Provider]", algorithm=MD5, uri="sip:[angerufene Nummer]@[SIP-Provider]", nonce="*******************=", response="*********************"
Date: Tue, 02 Sep 2014 00:44:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326

v=0
o=root 546951438 546951439 IN IP4 [meine ext. IP]
s=Asterisk PBX 11.5.0
c=IN IP4 [meine ext. IP]
t=0 0
m=audio 9332 RTP/AVP 0 8 9 10 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:10 L16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #1 (NAT) to [Provider-IP]:5060:
INVITE sip:[angerufene Nummer]@[SIP-Provider] SIP/2.0
Via: SIP/2.0/UDP [meine ext. IP]:5060;branch=z9hG4bK40f407ae;rport
Max-Forwards: 70
From: "[ICH]" <sip:[ausgehende Nummer]@[SIP-Provider]>;tag=as7ac3cb44
To: <sip:[angerufene Nummer]@[SIP-Provider]>
Contact: <sip:[ausgehende Nummer]@[meine ext. IP]:5060>
Call-ID: 13b5b7[EXT][email protected][SIP-Provider]
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0(11.5.0)
Proxy-Authorization: Digest username="[username]", realm="[SIP-Provider]", algorithm=MD5, uri="sip:[angerufene Nummer]@[SIP-Provider]", nonce="*******************=", response="*******************"
Date: Tue, 02 Sep 2014 00:44:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326

v=0
o=root 546951438 546951439 IN IP4 [meine ext. IP]
s=Asterisk PBX 11.5.0
c=IN IP4 [meine ext. IP]
t=0 0
m=audio 9332 RTP/AVP 0 8 9 10 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:10 L16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #2 (NAT) to [Provider-IP]:5060:
INVITE sip:[angerufene Nummer]@[SIP-Provider] SIP/2.0
Via: SIP/2.0/UDP [meine ext. IP]:5060;branch=z9hG4bK40f407ae;rport
Max-Forwards: 70
From: "[ICH]" <sip:[ausgehende Nummer]@[SIP-Provider]>;tag=as7ac3cb44
To: <sip:[angerufene Nummer]@[SIP-Provider]>
Contact: <sip:[ausgehende Nummer]@[meine ext. IP]:5060>
Call-ID: [email protected][SIP-Provider]
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0(11.5.0)
Proxy-Authorization: Digest username="[username]", realm="[SIP-Provider]", algorithm=MD5, uri="sip:[angerufene Nummer]@[SIP-Provider]", nonce="*******************", response="*******************"
Date: Tue, 02 Sep 2014 00:44:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326

v=0
o=root 546951438 546951439 IN IP4 [meine ext. IP]
s=Asterisk PBX 11.5.0
c=IN IP4 [meine ext. IP]
t=0 0
m=audio 9332 RTP/AVP 0 8 9 10 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:10 L16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Really destroying SIP dialog '[email protected]_0_1_249' Method: REGISTER

<--- SIP read from UDP:10.0.1.178:52035 --->

<------------->
Retransmitting #3 (NAT) to [Provider-IP]:5060:
INVITE sip:[angerufene Nummer]@[SIP-Provider] SIP/2.0
Via: SIP/2.0/UDP [meine ext. IP]:5060;branch=z9hG4bK40f407ae;rport
Max-Forwards: 70
From: "[ICH]" <sip:[ausgehende Nummer]@[SIP-Provider]>;tag=as7ac3cb44
To: <sip:[angerufene Nummer]@[SIP-Provider]>
Contact: <sip:[ausgehende Nummer]@[meine ext. IP]:5060>
Call-ID: 13b5b7[EXT][email protected][SIP-Provider]
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0(11.5.0)
Proxy-Authorization: Digest username="[username]", realm="[SIP-Provider]", algorithm=MD5, uri="sip:[angerufene Nummer]@[SIP-Provider]", nonce="*******************", response="*******************"
Date: Tue, 02 Sep 2014 00:44:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326

v=0
o=root 546951438 546951439 IN IP4 [meine ext. IP]
s=Asterisk PBX 11.5.0
c=IN IP4 [meine ext. IP]
t=0 0
m=audio 9332 RTP/AVP 0 8 9 10 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:10 L16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:[EXT-IP]:5060 --->
CANCEL sip:011[angerufene Nummer]@[Asterisk IP] SIP/2.0
Via: SIP/2.0/UDP [EXT-IP]:5060;branch=z9hG4bK-d4e65540
From: "Office" <sip:[EXT]@[Asterisk IP]>;tag=36fb1545eb54942co0
To: <sip:011[angerufene Nummer]@[Asterisk IP]>
Call-ID: [email protected][EXT-IP]
CSeq: 102 CANCEL
Max-Forwards: 70
Authorization: Digest username="[EXT]",realm="asterisk",nonce="*******************",uri="sip:011[angerufene Nummer]@[Asterisk IP]",algorithm=MD5,response="*******************"
User-Agent: Linksys/SPA921-5.1.8
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to [EXT-IP]:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to [EXT-IP]:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP [EXT-IP]:5060;branch=z9hG4bK-d4e65540;received=[EXT-IP]
From: "Office" <sip:[EXT]@[Asterisk IP]>;tag=36fb1545eb54942co0
To: <sip:011[angerufene Nummer]@[Asterisk IP]>;tag=as54e2ed33
Call-ID: [email protected][EXT-IP]
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


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

<--- Transmitting (no NAT) to [EXT-IP]:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP [EXT-IP]:5060;branch=z9hG4bK-d4e65540;received=[EXT-IP]
From: "Office" <sip:[EXT]@[Asterisk IP]>;tag=36fb1545eb54942co0
To: <sip:011[angerufene Nummer]@[Asterisk IP]>;tag=as54e2ed33
Call-ID: [email protected][EXT-IP]
CSeq: 102 CANCEL
Server: FPBX-2.11.0(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 '13b5b7[EXT][email protected][SIP-Provider]' in 32000 ms (Method: INVITE)
  == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/[EXT]-00000051' in macro 'dialout-trunk'
  == Spawn extension (from-internal, 011[angerufene Nummer], 5) exited non-zero on 'SIP/[EXT]-00000051'
    -- Executing [[email protected]:1] Hangup("SIP/[EXT]-00000051", "") in new stack
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/[EXT]-00000051'
Retransmitting #4 (NAT) to [Provider-IP]:5060:
INVITE sip:[angerufene Nummer]@[SIP-Provider] SIP/2.0
Via: SIP/2.0/UDP [meine ext. IP]:5060;branch=z9hG4bK40f407ae;rport
Max-Forwards: 70
From: "[ICH]" <sip:[ausgehende Nummer]@[SIP-Provider]>;tag=as7ac3cb44
To: <sip:[angerufene Nummer]@[SIP-Provider]>
Contact: <sip:[ausgehende Nummer]@[meine ext. IP]:5060>
Call-ID: 13b5b7[EXT][email protected][SIP-Provider]
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0(11.5.0)
Proxy-Authorization: Digest username="[username]", realm="[SIP-Provider]", algorithm=MD5, uri="sip:[angerufene Nummer]@[SIP-Provider]", nonce="*******************", response="*******************"
Date: Tue, 02 Sep 2014 00:44:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326

v=0
o=root 546951438 546951439 IN IP4 [meine ext. IP]
s=Asterisk PBX 11.5.0
c=IN IP4 [meine ext. IP]
t=0 0
m=audio 9332 RTP/AVP 0 8 9 10 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:10 L16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:[EXT-IP]:5060 --->
ACK sip:011[angerufene Nummer]@[Asterisk IP] SIP/2.0
Via: SIP/2.0/UDP [EXT-IP]:5060;branch=z9hG4bK-d4e65540
From: "Office" <sip:[EXT]@[Asterisk IP]>;tag=36fb1545eb54942co0
To: <sip:011[angerufene Nummer]@[Asterisk IP]>;tag=as54e2ed33
Call-ID: [email protected][EXT-IP]
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="[EXT]",realm="asterisk",nonce="*******************",uri="sip:011[angerufene Nummer]@[Asterisk IP]",algorithm=MD5,response="*******************"
Contact: "Office" <sip:[EXT]@[EXT-IP]:5060>
User-Agent: Linksys/SPA921-5.1.8
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '[email protected][EXT-IP]' Method: ACK
pbx*CLI> sip set debug off
 

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