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[Problem] ALL IP Anschluss mit Asterisk: Keine eingehenden Gespräche.

December 23, 2013, 5:27 am
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Hallo zusammen,

ich möchte mich hier auch mal mit einem kleinen Problem einklinken.
Ich denke ich sehe den Fehler vor lauter Bäumen nicht.

Also folgender Aufbau:
Ich hatte einen Speedport w724 an dem Funktionierte alles wie es soll. Diesen habe ich nun gegen ein HS 300 und pfsense(router) ausgetauscht. Hinter der Pfsense steht ein Asterisk in Version "11.2-cert2".

Diesen habe ich nun mit der Registrierung der Telekom Accounts betraut. Klappt soweit auch, nur leider bekommt man wenn man anrufen möchte ein 401 Unauthorized.
Abgehend kann ich ohne Probleme telefonieren.

Anbei mal meine sip.conf

Code:
[general]
port=5060
bindaddr=0.0.0.0
localnet=192.168.20.0/255.255.255.0
srvlookup=yes
nat=auto_force_rport,auto_comedia
session-timers=refuse
context=call-outside
maxexpirey=600
defaultexpirey=240


register => 042127xxxxx8:passwort:zugangsnummer-0001@t-online.de@tel.t-online.de/27xxxxx8~240
register => 042124xxxxxx:passwort:zugangsnummer-0001@t-online.de@tel.t-online.de/24xxxxxx~240
register => 042127xxxxx9:passwort:zugangsnummer-0001@t-online.de@tel.t-online.de/27xxxxx9~240

[telekom_out]
type=peer
defaultuser=zugangsnummer-0001@tel.t-online.de
secret=passwort
host=tel.t-online.de
fromdomain=tel.t-online.de
insecure=port,invite
directmedia=no

[telekom_in]
type=peer
allowguest=yes
context=telekom_in
insecure=port,invite
host=tel.t-online.de
fromdomain=tel.t-online.de
directmedia=no
qualify=yes

[10]
callerid="Wohnzimmer" <10>
secret=fon1
type=friend
host=dynamic

[20]
callerid="Buero 1" <20>
secret=fon2
type=friend
host=dynamic

[30]
callerid="Buero 2" <30>
secret=fon3
type=friend
host=dynamic
Der entsprechende Schnippsel aus der Extensions:

Code:
context telekom_in {
        27xxxxx8 => {
                Dial(SIP/20&SIP/30,30,rtT);
        }

        27xxxxx9  => {
                Dial(SIP/20&SIP/30,30,rtT);
        }

}
Und einen SIP debug von einem Anruf:

Code:
asterisk*CLI> sip set debug on
SIP Debugging enabled

<--- SIP read from UDP:217.0.19.166:5060 --->
INVITE sip:27xxxxx8@192.168.20.9:5060 SIP/2.0
Via: SIP/2.0/UDP 217.0.19.166:5060;branch=z9hG4bK96673175bc043964baecc7660356b0ce
Via: SIP/2.0/TCP 62.155.0.194:5060;branch=z9hG4bK4dc4f931b7f80f790f08379311e95202.cd53748e
Record-Route: <sip:LEITASP21-forward@217.0.19.166:5060;lr>, <sip:LEITASP21-forward@62.155.4.66:5060;transport=udp;lr>
Max-Forwards: 54
To: <sip:27xxxxx8@192.168.20.9:5060>
From: <sip:+49151xxxxxxxx@tel.t-online.de;user=phone>;tag=d6d9ad08
Call-ID: 629716a749c9a560@62.155.0.194
Contact: <sip:TASP002@62.155.0.194:5060>
Supported: histinfo,199
CSeq: 1210661 INVITE
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REGISTER, UPDATE
P-Asserted-Identity: <sip:+49151xxxxxxxx@tel.t-online.de;user=phone;rn_source=1>
P-Called-Party-ID: <sip:+4942127xxxxx8@tel.t-online.de:5060;transport=tcp;user=phone;rn_source=1>
Session-ID: 52B83395-00B2A457@ynxc2pcu-245
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 304

v=0
o=- 5054820131123135901 359923749 IN IP4 217.0.16.106
s=Phone Call via hiQ9200 SIPCA
c=IN IP4 217.0.1.2
t=0 0
m=audio 10850 RTP/AVP 8 100
b=AS:87
b=RS:1087
b=RR:3262
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sqn: 0
a=sendrecv
a=ptime:20
<------------->
--- (18 headers 16 lines) ---
Sending to 217.0.19.166:5060 (no NAT)
Using INVITE request as basis request - 629716a749c9a560@62.155.0.194
Found peer 'telekom_out' for '+49151xxxxxxxx' from 217.0.19.166:5060
Found RTP audio format 8
Found RTP audio format 100
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 100
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 217.0.1.2:10850
Looking for 27xxxxx8 in call-outside (domain 192.168.20.9)
list_route: hop: <sip:LEITASP21-forward@217.0.19.166:5060;lr>
list_route: hop: <sip:LEITASP21-forward@62.155.4.66:5060;transport=udp;lr>

<--- Transmitting (no NAT) to 217.0.19.166:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.0.19.166:5060;branch=z9hG4bK96673175bc043964baecc7660356b0ce;received=217.0.19.166
Via: SIP/2.0/TCP 62.155.0.194:5060;branch=z9hG4bK4dc4f931b7f80f790f08379311e95202.cd53748e
Record-Route: <sip:LEITASP21-forward@217.0.19.166:5060;lr>, <sip:LEITASP21-forward@62.155.4.66:5060;transport=udp;lr>
From: <sip:+49151xxxxxxxx@tel.t-online.de;user=phone>;tag=d6d9ad08
To: <sip:27xxxxx8@192.168.20.9:5060>
Call-ID: 629716a749c9a560@62.155.0.194
CSeq: 1210661 INVITE
Server: Asterisk PBX 11.2-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:27xxxxx8@192.168.20.9:5060>
Content-Length: 0


<------------>
Audio is at 10590
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 217.0.19.166:5060:
INVITE sip:27xxxxx8@tel.t-online.de SIP/2.0
Via: SIP/2.0/UDP 192.168.20.9:5060;branch=z9hG4bK61661c70
Max-Forwards: 70
From: "042127xxxxx8" <sip:042127xxxxx8@tel.t-online.de>;tag=as74737e1e
To: <sip:27xxxxx8@tel.t-online.de>
Contact: <sip:042127xxxxx8@192.168.20.9:5060>
Call-ID: 2c73a712102c80495c88f3d31516da07@tel.t-online.de
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.2-cert2
Date: Mon, 23 Dec 2013 12:59:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 1008507934 1008507934 IN IP4 192.168.20.9
s=Asterisk PBX 11.2-cert2
c=IN IP4 192.168.20.9
t=0 0
m=audio 10590 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:217.0.19.166:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.20.9:5060;rport=6635;received=91.58.250.191;branch=z9hG4bK61661c70
To: <sip:27xxxxx8@tel.t-online.de>;tag=d91bdd71
From: 042127xxxxx8 <sip:042127xxxxx8@tel.t-online.de>;tag=as74737e1e
Call-ID: 2c73a712102c80495c88f3d31516da07@tel.t-online.de
CSeq: 102 INVITE
WWW-Authenticate: Digest algorithm=MD5, nonce="ddc4e95dddc4e95d8f7cdac8c155e5065c57c4b8bb637628a4882a92ff2f0274", realm="tel.t-online.de"
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 217.0.19.166:5060:
ACK sip:27xxxxx8@tel.t-online.de SIP/2.0
Via: SIP/2.0/UDP 192.168.20.9:5060;branch=z9hG4bK61661c70
Max-Forwards: 70
From: "042127xxxxx8" <sip:042127xxxxx8@tel.t-online.de>;tag=as74737e1e
To: <sip:27xxxxx8@tel.t-online.de>;tag=d91bdd71
Contact: <sip:042127xxxxx8@192.168.20.9:5060>
Call-ID: 2c73a712102c80495c88f3d31516da07@tel.t-online.de
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.2-cert2
Content-Length: 0


---
Audio is at 10590
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 217.0.19.166:5060:
INVITE sip:27xxxxx8@tel.t-online.de SIP/2.0
Via: SIP/2.0/UDP 192.168.20.9:5060;branch=z9hG4bK01658c52
Max-Forwards: 70
From: "042127xxxxx8" <sip:042127xxxxx8@tel.t-online.de>;tag=as74737e1e
To: <sip:27xxxxx8@tel.t-online.de>
Contact: <sip:042127xxxxx8@192.168.20.9:5060>
Call-ID: 2c73a712102c80495c88f3d31516da07@tel.t-online.de
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.2-cert2
Authorization: Digest username="551121360633-0001@tel.t-online.de", realm="tel.t-online.de", algorithm=MD5, uri="sip:27xxxxx8@tel.t-online.de", nonce="ddc4e95dddc4e95d8f7cdac8c155e5065c57c4b8bb637628a4882a92ff2f0274", response="2db298b5fbf4d497f42b048556c306fc"
Date: Mon, 23 Dec 2013 12:59:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 1008507934 1008507935 IN IP4 192.168.20.9
s=Asterisk PBX 11.2-cert2
c=IN IP4 192.168.20.9
t=0 0
m=audio 10590 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:217.0.19.166:5060 --->
SIP/2.0 100 Rufaufbau
Via: SIP/2.0/UDP 192.168.20.9:5060;rport=6635;received=91.58.250.191;branch=z9hG4bK01658c52
To: <sip:27xxxxx8@tel.t-online.de>
From: 042127xxxxx8 <sip:042127xxxxx8@tel.t-online.de>;tag=as74737e1e
Call-ID: 2c73a712102c80495c88f3d31516da07@tel.t-online.de
CSeq: 103 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '57014c886bbec7c9@217.0.19.166' Method: ACK
Really destroying SIP dialog '3e285d4f951092bf@62.155.2.195' Method: ACK

<--- SIP read from UDP:217.0.19.166:5060 --->
INVITE sip:27xxxxx8@192.168.20.9:5060 SIP/2.0
Via: SIP/2.0/UDP 217.0.19.166:5060;branch=z9hG4bK9d4e3a0d8e6a31224c8c0b7a0cd8c041.157676ed
Max-Forwards: 70
To: <sip:27xxxxx8@192.168.20.9:5060>
From: +4942127xxxxx8 <sip:+4942127xxxxx8@tel.t-online.de;user=phone;rn_source=1>;tag=47296f1b
Call-ID: cedd47b8545b9308@217.0.19.166
Contact: <sip:LEITASP21@217.0.19.166:5060>
Supported: histinfo
CSeq: 1910022 INVITE
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PUBLISH, REFER, REGISTER, SUBSCRIBE
P-Asserted-Identity: <sip:+4942127xxxxx8@tel.t-online.de;user=phone>
P-Called-Party-ID: <sip:+4942127xxxxx8@tel.t-online.de;rn_source=1>
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 308

v=0
o=- 1008507934 1008507935 IN IP4 217.0.19.166
s=Asterisk PBX 11.2-cert2
c=IN IP4 217.0.0.69
t=0 0
m=audio 15576 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (15 headers 14 lines) ---
Sending to 217.0.19.166:5060 (no NAT)
Using INVITE request as basis request - cedd47b8545b9308@217.0.19.166
Found peer 'telekom_out' for '+4942127xxxxx8' from 217.0.19.166:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(gsm|ulaw|alaw)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 217.0.0.69:15576
Looking for 27xxxxx8 in call-outside (domain 192.168.20.9)
list_route: hop: <sip:LEITASP21@217.0.19.166:5060>

<--- Transmitting (no NAT) to 217.0.19.166:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.0.19.166:5060;branch=z9hG4bK9d4e3a0d8e6a31224c8c0b7a0cd8c041.157676ed;received=217.0.19.166
From: +4942127xxxxx8 <sip:+4942127xxxxx8@tel.t-online.de;user=phone;rn_source=1>;tag=47296f1b
To: <sip:27xxxxx8@192.168.20.9:5060>
Call-ID: cedd47b8545b9308@217.0.19.166
CSeq: 1910022 INVITE
Server: Asterisk PBX 11.2-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:27xxxxx8@192.168.20.9:5060>
Content-Length: 0


<------------>
Audio is at 12424
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 217.0.19.166:5060:
INVITE sip:27xxxxx8@tel.t-online.de SIP/2.0
Via: SIP/2.0/UDP 192.168.20.9:5060;branch=z9hG4bK4180b01d
Max-Forwards: 70
From: "042127xxxxx8" <sip:042127xxxxx8@tel.t-online.de>;tag=as3455ace9
To: <sip:27xxxxx8@tel.t-online.de>
Contact: <sip:042127xxxxx8@192.168.20.9:5060>
Call-ID: 153849d734242edc47b4c9f20657eabe@tel.t-online.de
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.2-cert2
Date: Mon, 23 Dec 2013 12:59:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 1500640814 1500640814 IN IP4 192.168.20.9
s=Asterisk PBX 11.2-cert2
c=IN IP4 192.168.20.9
t=0 0
m=audio 12424 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 - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:217.0.19.166:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.20.9:5060;rport=6635;received=91.58.250.191;branch=z9hG4bK4180b01d
To: <sip:27xxxxx8@tel.t-online.de>;tag=f5fec82a
From: 042127xxxxx8 <sip:042127xxxxx8@tel.t-online.de>;tag=as3455ace9
Call-ID: 153849d734242edc47b4c9f20657eabe@tel.t-online.de
CSeq: 102 INVITE
WWW-Authenticate: Digest algorithm=MD5, nonce="8ebbaf648ebbaf64dc039cf1922aa27e2c3d2bc50881dba9dba35e5b420d6baf", realm="tel.t-online.de"
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 217.0.19.166:5060:
ACK sip:27xxxxx8@tel.t-online.de SIP/2.0
Via: SIP/2.0/UDP 192.168.20.9:5060;branch=z9hG4bK4180b01d
Max-Forwards: 70
From: "042127xxxxx8" <sip:042127xxxxx8@tel.t-online.de>;tag=as3455ace9
To: <sip:27xxxxx8@tel.t-online.de>;tag=f5fec82a
Contact: <sip:042127xxxxx8@192.168.20.9:5060>
Call-ID: 153849d734242edc47b4c9f20657eabe@tel.t-online.de
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.2-cert2
Content-Length: 0


---
Audio is at 12424
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 217.0.19.166:5060:
INVITE sip:27xxxxx8@tel.t-online.de SIP/2.0
Via: SIP/2.0/UDP 192.168.20.9:5060;branch=z9hG4bK0865de84
Max-Forwards: 70
From: "042127xxxxx8" <sip:042127xxxxx8@tel.t-online.de>;tag=as3455ace9
To: <sip:27xxxxx8@tel.t-online.de>
Contact: <sip:042127xxxxx8@192.168.20.9:5060>
Call-ID: 153849d734242edc47b4c9f20657eabe@tel.t-online.de
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.2-cert2
Authorization: Digest username="551121360633-0001@tel.t-online.de", realm="tel.t-online.de", algorithm=MD5, uri="sip:27xxxxx8@tel.t-online.de", nonce="8ebbaf648ebbaf64dc039cf1922aa27e2c3d2bc50881dba9dba35e5b420d6baf", response="c201146469a42249ac9c574d2aad92b0"
Date: Mon, 23 Dec 2013 12:59:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 1500640814 1500640815 IN IP4 192.168.20.9
s=Asterisk PBX 11.2-cert2
c=IN IP4 192.168.20.9
t=0 0
m=audio 12424 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 - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:217.0.19.166:5060 --->
SIP/2.0 100 Rufaufbau
Via: SIP/2.0/UDP 192.168.20.9:5060;rport=6635;received=91.58.250.191;branch=z9hG4bK0865de84
To: <sip:27xxxxx8@tel.t-online.de>
From: 042127xxxxx8 <sip:042127xxxxx8@tel.t-online.de>;tag=as3455ace9
Call-ID: 153849d734242edc47b4c9f20657eabe@tel.t-online.de
CSeq: 103 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:217.0.19.166:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.20.9:5060;rport=6635;received=91.58.250.191;branch=z9hG4bK0865de84
To: <sip:27xxxxx8@tel.t-online.de>;tag=9d8c6058
From: 042127xxxxx8 <sip:042127xxxxx8@tel.t-online.de>;tag=as3455ace9
Call-ID: 153849d734242edc47b4c9f20657eabe@tel.t-online.de
Contact: <sip:LEITASP21@217.0.19.166:5060>
CSeq: 103 INVITE
Reason: SIP;cause=486;text="Network Determined User Busy - Originating";reason=4860009
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 217.0.19.166:5060:
ACK sip:27xxxxx8@tel.t-online.de SIP/2.0
Via: SIP/2.0/UDP 192.168.20.9:5060;branch=z9hG4bK0865de84
Max-Forwards: 70
From: "042127xxxxx8" <sip:042127xxxxx8@tel.t-online.de>;tag=as3455ace9
To: <sip:27xxxxx8@tel.t-online.de>;tag=9d8c6058
Contact: <sip:042127xxxxx8@192.168.20.9:5060>
Call-ID: 153849d734242edc47b4c9f20657eabe@tel.t-online.de
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.2-cert2
Content-Length: 0


---
Scheduling destruction of SIP dialog 'cedd47b8545b9308@217.0.19.166' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to 217.0.19.166:5060 --->
SIP/2.0 486 Busy here
Via: SIP/2.0/UDP 217.0.19.166:5060;branch=z9hG4bK9d4e3a0d8e6a31224c8c0b7a0cd8c041.157676ed;received=217.0.19.166
From: +4942127xxxxx8 <sip:+4942127xxxxx8@tel.t-online.de;user=phone;rn_source=1>;tag=47296f1b
To: <sip:27xxxxx8@192.168.20.9:5060>;tag=as52327bf9
Call-ID: cedd47b8545b9308@217.0.19.166
CSeq: 1910022 INVITE
Server: Asterisk PBX 11.2-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0


<------------>
Really destroying SIP dialog '153849d734242edc47b4c9f20657eabe@tel.t-online.de' Method: INVITE

<--- SIP read from UDP:217.0.19.166:5060 --->
ACK sip:27xxxxx8@192.168.20.9:5060 SIP/2.0
Via: SIP/2.0/UDP 217.0.19.166:5060;branch=z9hG4bK9d4e3a0d8e6a31224c8c0b7a0cd8c041.157676ed
Max-Forwards: 70
To: <sip:27xxxxx8@192.168.20.9:5060>;tag=as52327bf9
From: +4942127xxxxx8 <sip:+4942127xxxxx8@tel.t-online.de;user=phone;rn_source=1>;tag=47296f1b
Call-ID: cedd47b8545b9308@217.0.19.166
CSeq: 1910022 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:217.0.19.166:5060 --->
SIP/2.0 486 Besetzt (0)
Via: SIP/2.0/UDP 192.168.20.9:5060;rport=6635;received=91.58.250.191;branch=z9hG4bK01658c52
To: <sip:27xxxxx8@tel.t-online.de>;tag=2aeeaf87
From: 042127xxxxx8 <sip:042127xxxxx8@tel.t-online.de>;tag=as74737e1e
Call-ID: 2c73a712102c80495c88f3d31516da07@tel.t-online.de
Contact: <sip:LEITASP21@217.0.19.166:5060>
CSeq: 103 INVITE
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PUBLISH, REFER, REGISTER, SUBSCRIBE
Server: Asterisk PBX 11.2-cert2
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 217.0.19.166:5060:
ACK sip:27xxxxx8@tel.t-online.de SIP/2.0
Via: SIP/2.0/UDP 192.168.20.9:5060;branch=z9hG4bK01658c52
Max-Forwards: 70
From: "042127xxxxx8" <sip:042127xxxxx8@tel.t-online.de>;tag=as74737e1e
To: <sip:27xxxxx8@tel.t-online.de>;tag=2aeeaf87
Contact: <sip:042127xxxxx8@192.168.20.9:5060>
Call-ID: 2c73a712102c80495c88f3d31516da07@tel.t-online.de
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.2-cert2
Content-Length: 0


---
Scheduling destruction of SIP dialog '629716a749c9a560@62.155.0.194' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to 217.0.19.166:5060 --->
SIP/2.0 486 Busy here
Via: SIP/2.0/UDP 217.0.19.166:5060;branch=z9hG4bK96673175bc043964baecc7660356b0ce;received=217.0.19.166
Via: SIP/2.0/TCP 62.155.0.194:5060;branch=z9hG4bK4dc4f931b7f80f790f08379311e95202.cd53748e
From: <sip:+49151xxxxxxxx@tel.t-online.de;user=phone>;tag=d6d9ad08
To: <sip:27xxxxx8@192.168.20.9:5060>;tag=as227278db
Call-ID: 629716a749c9a560@62.155.0.194
CSeq: 1210661 INVITE
Server: Asterisk PBX 11.2-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0


<------------>
Really destroying SIP dialog '2c73a712102c80495c88f3d31516da07@tel.t-online.de' Method: INVITE

<--- SIP read from UDP:217.0.19.166:5060 --->
ACK sip:27xxxxx8@192.168.20.9:5060 SIP/2.0
Via: SIP/2.0/UDP 217.0.19.166:5060;branch=z9hG4bK96673175bc043964baecc7660356b0ce
Max-Forwards: 54
To: <sip:27xxxxx8@192.168.20.9:5060>;tag=as227278db
From: <sip:+49151xxxxxxxx@tel.t-online.de;user=phone>;tag=d6d9ad08
Call-ID: 629716a749c9a560@62.155.0.194
CSeq: 1210661 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
asterisk*CLI> sip set debug off
SIP Debugging Disabled

Soweit so gut. Vielleicht hat einer für mich einen kleine Schlag auf den Hinterkopf übrig :)

Danke & Gruß
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