Keine Telefonate möglich / SIP to SIP

Hi,

hab mich heute zu ersten Mal mit Asterisk und meiner Fritzbox beschäftigt.
Installation ist sauber durchgelaufen und ich habe auch schon die ersten beiden Lektionen von betateilchen "durchgeackert"

Nun zu meinem Problem.
Ich habe 3 SIP Clients eingerichtet und haben sich auch registriert.

1x FritzBox Internetnummer
1x X-Lite auf PC
1x SJ Phone auf HTC Kaiser

Den SIP Port habe ich auf 5061 gestellt, da ja sonst die FB Probleme macht.

Leider kann ich keinen Ruf zu den einzelnen Clients aufbauen.
Dann kommt die Fehlermeldung "Declined to talk 603"

Die extensions.conf habe ich komplett von betateilchen, sodass die Grundkonfig erst einmal laufen sollte.

Die Asterisk Konsole gibt bei Anrufversuch von 30 auf 31 folgendes aus.



<------------->
--- (0 headers 0 lines) Nat keepalive ---

(none)*CLI>
<--- SIP read from 192.168.1.20:46172 --->



<------------->
--- (0 headers 1 lines) ---

(none)*CLI>
<--- SIP read from 192.168.1.24:5060 --->


<------------->
--- (0 headers 0 lines) Nat keepalive ---

Really destroying SIP dialog '[email protected]' Method: ACK

(none)*CLI>
<--- SIP read from 192.168.1.20:46172 --->



<------------->
--- (0 headers 1 lines) ---

(none)*CLI>
<--- SIP read from 192.168.1.24:5060 --->


<------------->
--- (0 headers 0 lines) Nat keepalive ---

(none)*CLI>
<--- SIP read from 192.168.1.24:5060 --->
INVITE sip:[email protected] SIP/2.0
To: <sip:[email protected]>
From: "unknown"<sip:[email protected]>;tag=952478825748
Via: SIP/2.0/UDP 192.168.1.24;rport;branch=z9hG4bKc0a801180000013f47e7ff3500007f8200000264
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Max-Forwards: 70
User-Agent: SJphone/1.60.303c (SJ Labs)
Content-Length: 266
Content-Type: application/sdp

v=0
o=- 3415375285 3415375285 IN IP4 192.168.1.24
s=SJphone
c=IN IP4 192.168.1.24
t=0 0
a=direction:active
m=audio 49206 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16

<------------->
--- (11 headers 12 lines) ---
Sending to 192.168.1.24 : 5060 (NAT)

Using INVITE request as basis request - [email protected]

<--- Reliably Transmitting (NAT) to 192.168.1.24:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.24;branch=z9hG4bKc0a801180000013f47e7ff3500007f8200000264;received=192.168.1.24;rport=5060
From: "unknown"<sip:[email protected]>;tag=952478825748
To: <sip:[email protected]>;tag=as022693d6
Call-ID: [email protected]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
A
llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4ddcb1d8"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
Found user '30'

(none)*CLI>
<--- SIP read from 192.168.1.24:5060 --->
ACK sip:[email protected] SIP/2.0
To: <sip:[email protected]>;tag=as022693d6
From: "unknown"<sip:[email protected]>;tag=952478825748
Via: SIP/2.0/UDP 192.168.1.24;rport;branch=z9hG4bKc0a801180000013f47e7ff3500007f8200000264
Call-ID: [email protected]
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0


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

(none)*CLI>
<--- SIP read from 192.168.1.24:5060 --->
INVITE sip:[email protected] SIP/2.0
To: <sip:[email protected]>
From: "unknown"<sip:[email protected]>;tag=952478825748
Via: SIP/2.0/UDP 192.168.1.24;rport;branch=z9hG4bKc0a801180000013f47e7ff36000034e900000265
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]>
Max-Forwards: 70
User-Agent: SJphone/1.60.303c (SJ Labs)
Content-Length: 266
Content-Type: application/sdp
Proxy-Authorization: Digest username="30",realm="asterisk",nonce="4ddcb1d8",uri="sip:[email protected]",response="817916cc2dc79dcc9c390c098e1f03e7",algorithm=MD5

v=0
o=- 3415375285 3415375285 IN IP4 192.168.1.24
s=SJphone
c=IN IP4 192.168.1.24
t=0 0
a=direction:active
m=audio 49206 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16

<------------->
--- (12 headers 12 lines) ---
Sending to 192.168.1.24 : 5060 (NAT)
Using INVITE request as basis request - [email protected]

Found user '30'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.24:49206
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (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 192.168.1.24:49206
Looking for 31 in default (domain 192.168.1.1)
list_route: hop: <sip:[email protected]>

<--- Transmitting (NAT) to 192.168.1.24:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.24;branch=z9hG4bKc0a801180000013f47e7ff36000034e900000265;received=192.168.1.24;rport=5060
From: "unknown"<sip:[email protected]>;tag=952478825748
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]:5061>
Content-Length: 0


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

[Mar 24 19:20:44] WARNING[3986]: pbx.c:1817 pbx_extension_helper: No application 'NoCDR' for extension (default, 31, 1)

== Spawn extension (default, 31, 1) exited non-zero on 'SIP/30-005f10e8'

Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)

(none)*CLI>
<--- Reliably Transmitting (NAT) to 192.168.1.24:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.24;branch=z9hG4bKc0a801180000013f47e7ff36000034e900000265;received=192.168.1.24;rport=5060
From: "unknown"<sip:[email protected]>;tag=952478825748
To: <sip:[email protected]>;tag=as465e8751
Call-ID: [email protected]
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]:5061>
Content-Length: 0


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

(none)*CLI>
<--- SIP read from 192.168.1.24:5060 --->
ACK sip:[email protected] SIP/2.0
To: <sip:[email protected]>;tag=as465e8751
From: "unknown"<sip:[email protected]>;tag=952478825748
Via: SIP/2.0/UDP 192.168.1.24;rport;branch=z9hG4bKc0a801180000013f47e7ff36000034e900000265
Call-ID: [email protected]
CSeq: 2 ACK
Max-Forwards: 70
Content-Length: 0


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

(none)*CLI>






Laut Asterisk sind die Clients auch angemeldet

Vielleicht kann mir jemand dabei weiterhelfen, dann langsam sehe ich nicht mehr durch.

Danke im Voraus

Kevin

@dynamic Danke für die schnell Hilfe.

Hab die extension.conf und die sip.conf wiederhergestellt.
Und nun hab ich schon mal ne Rufsignalisierung.

Hab zwar noch keine Sprachverbindung, aber dass werde ich mir morgen mal ansehen.

Gruß
kevin