SIP / ENUM

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jstocker

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Irgendwie schaff ich es nicht einen annonymen SIP call in die weite Welt zu machen. Die Fehlermeldungen sehen wie folgt aus.

Code: 
  -- Executing EnumLookup("SIP/p200-b0e6", "493039833444") in new stack
ENUM got '1'
    -- Executing Dial("SIP/p200-b0e6", "SIP/[email protected]|30") in new stack
    -- parse_srv: SRV mapped to host sip.snom.com, port 5082
    -- Called [email][email protected][/email]
Apr 29 23:09:18 NOTICE[1503]: chan_sip.c:6880 handle_response: Failed to authenticate on INVITE to '"P200@Asterisk" <sip:[email protected]>;tag=as4ec63708'
  == Spawn extension (sip-in-local, **493039833444, 2) exited non-zero on 'SIP/p200-b0e6'
    -- Got SIP response 481 "Call/Transaction does not exist" back from 217.115.141.99

Der Name "P200@Asterisk" kommt direkt vom IP Phone und hat nix mit Asterisk selber zu tun. Die IP 84.137.232.146 ist zur Zeit die tatsächliche IP des Asterisk Servers und von draußen durch 5060 erreichbar.

Extensions.conf
Code: 
exten => _**.,1,EnumLookup(${EXTEN:2})
exten => _**.,2,Dial(${ENUM},30)
exten => _**.,3,Congestion

Die ENUM Auflösung klappt einwandfrei, wie man an obiger Ausgabe sehen kann. Ich kann nun gar nichts mit der Fehlermeldung anfangen. Die Warnung stellt für mich auch Rätsel dar: Wieso will da sich das Telefon authentifizieren und wie soll das gehen, denn die beiden kennen sich ja nicht, das ist ja auch der Sinn eines ENUM Anrufes.

Kann mir jemand da weiter helfen?

Ich habe folgende Konfiguration

Internet --> FW + Asterisk -> 10.x.x.x -> Hardphone

Da Asterisk ebenfalls auf beiden Netzen horcht stellt sich nun die Frage des NAT. Wenn ich einen SIP call zu Asterisk mache (ueber 10.x.x.x) dieser dann einen Outcall ins Internet macht fungiert er hier ja hoffentlich als Gateway und zwingt das SIP Phone nicht die Verbindung selber aufzubauen (ich gehe jedenfalls ganz stark davon aus). Also sollte ich hier kein NAT Problem haben.
 
Ein 'sip debug' waere hier sehr hilfreich. Ausserdem waere es gut zu wissen, welche Asterisk-Version du verwendest.
 
Here you go...

Asterisk 1.0.7


Code: 
Sip read: 
INVITE sip:**[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK_00D0E9014CA1_T16547F5A
Session-Expires: 120
From: "P200@Asterisk" <sip:[email protected]:5060>;tag=00D0E9014CA1_T1218104444
To: <sip:**[email protected]:5060>
Call-ID: [email protected]
CSeq: 684307913 INVITE
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,INFO
Supported: timer,replaces
User-Agent: P200 02.09
Content-Type: application/sdp
Content-Length: 224

v=0
o=p200 471593064 471593064 IN IP4 10.1.1.3
s=P200 02.09
c=IN IP4 10.1.1.3
t=0 0
m=audio 8000 RTP/AVP 0 18 4
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000/1
a=sendrecv

14 headers, 11 lines
Using latest request as basis request
Sending to 10.1.1.3 : 5060 (non-NAT)
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK_00D0E9014CA1_T16547F5A
From: "P200@Asterisk" <sip:[email protected]:5060>;tag=00D0E9014CA1_T1218104444
To: <sip:**[email protected]:5060>;tag=as4f40f49e
Call-ID: [email protected]
CSeq: 684307913 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:**[email protected]>
Proxy-Authenticate: Digest realm="asterisk", nonce="317bd9b8"
Content-Length: 0


 to 10.1.1.3:5060
Scheduling destruction of call '[email protected]' in 15000 ms
Found user 'p200'


Sip read: 
ACK sip:**[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK_00D0E9014CA1_T16547F5A
From: "P200@Asterisk" <sip:[email protected]:5060>;tag=00D0E9014CA1_T1218104444
To: <sip:**[email protected]:5060>;tag=as4f40f49e
Call-ID: [email protected]
CSeq: 684307913 ACK
User-Agent: P200 02.09
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
Content-Length: 0


10 headers, 0 lines


Sip read: 
INVITE sip:**[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK_00D0E9014CA1_T15DE7F51
Session-Expires: 120
From: "P200@Asterisk" <sip:[email protected]:5060>;tag=00D0E9014CA1_T1218104444
To: <sip:**[email protected]:5060>
Call-ID: [email protected]
CSeq: 684307914 INVITE
Proxy-Authorization: Digest username="p200", realm="asterisk", nonce="317bd9b8", opaque="", uri="sip:**[email protected]:5060", response="f782a310c1849f462bfb763b55e90400"
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,INFO
Supported: timer,replaces
User-Agent: P200 02.09
Content-Type: application/sdp
Content-Length: 224

v=0
o=p200 471593064 471593064 IN IP4 10.1.1.3
s=P200 02.09
c=IN IP4 10.1.1.3
t=0 0
m=audio 8000 RTP/AVP 0 18 4
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000/1
a=sendrecv

15 headers, 11 lines
Using latest request as basis request
Sending to 10.1.1.3 : 5060 (non-NAT)
Found user 'p200'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 4
Peer audio RTP is at port 10.1.1.3:8000
Found description format PCMU
Found description format G729
Found description format G723
Capabilities: us - 0x4 (ulaw), peer - audio=0x105 (g723|ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing)
Looking for **493039833444 in sip-in-local
list_route: hop: <sip:[email protected]:5060>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK_00D0E9014CA1_T15DE7F51
From: "P200@Asterisk" <sip:[email protected]:5060>;tag=00D0E9014CA1_T1218104444
To: <sip:**[email protected]:5060>;tag=as3a12d869
Call-ID: [email protected]
CSeq: 684307914 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:**[email protected]>
Content-Length: 0
 to 10.1.1.3:5060
    -- Executing EnumLookup("SIP/p200-f348", "493039833444") in new stack
Urgent handler
Urgent handler
ENUM got '1'
    -- Executing Dial("SIP/p200-f348", "SIP/[email protected]|30") in new stack
    -- parse_srv: SRV mapped to host sip.snom.com, port 5082
We're at 84.137.243.197 port 19316
Answering/Requesting with root capability 0x4 (ulaw)
Answering with capability 0x2 (gsm)
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:[email protected]:5082 SIP/2.0
Via: SIP/2.0/UDP 84.137.243.197:5060;branch=z9hG4bK68de096b
From: "P200@Asterisk" <sip:[email protected]>;tag=as062fb784
To: <sip:[email protected]:5082>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 30 Apr 2005 09:54:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 731 731 IN IP4 84.137.243.197
s=session
c=IN IP4 84.137.243.197
t=0 0
m=audio 19316 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 - - - -
 (no NAT) to 217.115.141.99:5082
    -- Called [email][email protected][/email]
Urgent handler


Sip read: 
SIP/2.0 407 Proxy Authorization Required
v: SIP/2.0/UDP 84.137.243.197:5060;branch=z9hG4bK68de096b
f: "P200@Asterisk" <sip:[email protected]>;tag=as062fb784
t: <sip:[email protected]:5082>;tag=wkwisk5kxi
i: [email protected]

CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="default", nonce="994dd6e729dac4e077a227a7e77bb394", opaque="", stale=TRUE, algorithm=MD5
l: 0


8 headers, 0 lines
Transmitting:
ACK sip:[email protected]:5082 SIP/2.0
Via: SIP/2.0/UDP 84.137.243.197:5060;branch=z9hG4bK68de096b
From: "P200@Asterisk" <sip:[email protected]>;tag=as062fb784
To: <sip:[email protected]:5082>;tag=wkwisk5kxi
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 217.115.141.99:5082
We're at 84.137.243.197 port 19316
Answering/Requesting with root capability 0x4 (ulaw)
Answering with capability 0x2 (gsm)
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting:
INVITE sip:[email protected]:5082 SIP/2.0
Via: SIP/2.0/UDP 84.137.243.197:5060;branch=z9hG4bK44df73f3
From: "P200@Asterisk" <sip:[email protected]>;tag=as062fb784
To: <sip:[email protected]:5082>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="", realm="default", algorithm=MD5, uri="sip:[email protected]:5082", nonce="994dd6e729dac4e077a227a7e77bb394", response="afa4b70d50f3fe1b3c9068bfa8a4d48c", opaque=""
Date: Sat, 30 Apr 2005 09:54:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 731 732 IN IP4 84.137.243.197
s=session
c=IN IP4 84.137.243.197
t=0 0
m=audio 19316 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 - - - -
 (no NAT) to 217.115.141.99:5082
Sip read: 
SIP/2.0 407 Proxy Authorization Required
v: SIP/2.0/UDP 84.137.243.197:5060;branch=z9hG4bK44df73f3
f: "P200@Asterisk" <sip:[email protected]>;tag=as062fb784
t: <sip:[email protected]:5082>;tag=rzmlmd0t1d
i: [email protected]
CSeq: 103 INVITE
Proxy-Authenticate: Digest realm="default", nonce="e6700a020a0003e6e46abd1a3ff398d7", opaque="", stale=TRUE, algorithm=MD5
l: 0


8 headers, 0 lines
Transmitting:
ACK sip:[email protected]:5082 SIP/2.0
Via: SIP/2.0/UDP 84.137.243.197:5060;branch=z9hG4bK44df73f3
From: "P200@Asterisk" <sip:[email protected]>;tag=as062fb784
To: <sip:[email protected]:5082>;tag=rzmlmd0t1d
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 217.115.141.99:5082
We're at 84.137.243.197 port 19316
Answering/Requesting with root capability 0x4 (ulaw)
Answering with capability 0x2 (gsm)
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting:
INVITE sip:[email protected]:5082 SIP/2.0
Via: SIP/2.0/UDP 84.137.243.197:5060;branch=z9hG4bK00dbdcd4
From: "P200@Asterisk" <sip:[email protected]>;tag=as062fb784
To: <sip:[email protected]:5082>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="", realm="default", algorithm=MD5, uri="sip:[email protected]:5082", nonce="e6700a020a0003e6e46abd1a3ff398d7", response="954d22ec0c5ee679182fdf6b0704bc9f", opaque=""
Date: Sat, 30 Apr 2005 09:54:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 731 733 IN IP4 84.137.243.197
s=session
c=IN IP4 84.137.243.197
t=0 0

m=audio 19316 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 - - - -
 (no NAT) to 217.115.141.99:5082


Sip read: 
SIP/2.0 407 Proxy Authorization Required
v: SIP/2.0/UDP 84.137.243.197:5060;branch=z9hG4bK00dbdcd4
f: "P200@Asterisk" <sip:[email protected]>;tag=as062fb784
t: <sip:[email protected]:5082>;tag=64fvi35m9h
i: [email protected]
CSeq: 104 INVITE
Proxy-Authenticate: Digest realm="default", nonce="e15d731ddcbf6ee617eeed25c1ddfeee", opaque="", stale=TRUE, algorithm=MD5
l: 0


8 headers, 0 lines
Transmitting:
ACK sip:[email protected]:5082 SIP/2.0
Via: SIP/2.0/UDP 84.137.243.197:5060;branch=z9hG4bK00dbdcd4
From: "P200@Asterisk" <sip:[email protected]>;tag=as062fb784
To: <sip:[email protected]:5082>;tag=64fvi35m9h
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 104 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 217.115.141.99:5082
Apr 30 11:54:42 NOTICE[731]: chan_sip.c:6880 handle_response: Failed to authenticate on INVITE to '"P200@Asterisk" <sip:[email protected]>;tag=as062fb784'
Urgent handler
Urgent handler


Sip read: 
CANCEL sip:**[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK_00D0E9014CA1_T15DE7F51
From: "P200@Asterisk" <sip:[email protected]:5060>;tag=00D0E9014CA1_T1218104444
To: <sip:**[email protected]:5060>
Call-ID: [email protected]
CSeq: 684307914 CANCEL
User-Agent: P200 02.09
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
Content-Length: 0


10 headers, 0 lines
Sending to 10.1.1.3 : 5060 (non-NAT)
Reliably Transmitting (no NAT):
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK_00D0E9014CA1_T15DE7F51
From: "P200@Asterisk" <sip:[email protected]:5060>;tag=00D0E9014CA1_T1218104444
To: <sip:**[email protected]:5060>;tag=as3a12d869
Call-ID: [email protected]
CSeq: 684307914 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:**[email protected]>
Content-Length: 0


 to 10.1.1.3:5060
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK_00D0E9014CA1_T15DE7F51
From: "P200@Asterisk" <sip:[email protected]:5060>;tag=00D0E9014CA1_T1218104444
To: <sip:**[email protected]:5060>;tag=as3a12d869
Call-ID: [email protected]
CSeq: 684307914 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:**[email protected]>
Content-Length: 0


 to 10.1.1.3:5060
Reliably Transmitting:
CANCEL sip:[email protected]:5082 SIP/2.0
Via: SIP/2.0/UDP 84.137.243.197:5060;branch=z9hG4bK00dbdcd4
From: "P200@Asterisk" <sip:[email protected]>;tag=as062fb784
To: <sip:[email protected]:5082>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 104 CANCEL
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="", realm="default", algorithm=MD5, uri="sip:[email protected]:5082", nonce="e6700a020a0003e6e46abd1a3ff398d7", response="3224127ffd897fdae81f25788c5c904c", opaque=""
Content-Length: 0

 (no NAT) to 217.115.141.99:5082
Scheduling destruction of call '[email protected]' in 15000 ms


Sip read: 
SIP/2.0 481 Call/Transaction does not exist
v: SIP/2.0/UDP 84.137.243.197:5060;branch=z9hG4bK00dbdcd4
f: "P200@Asterisk" <sip:[email protected]>;tag=as062fb784

t: <sip:[email protected]:5082>
i: [email protected]
CSeq: 104 CANCEL
l: 0


7 headers, 0 lines
    -- Got SIP response 481 "Call/Transaction does not exist" back from 217.115.141.99
Destroying call '[email protected]'


Sip read: 
ACK sip:**[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK_00D0E9014CA1_T15DE7F51
From: "P200@Asterisk" <sip:[email protected]:5060>;tag=00D0E9014CA1_T1218104444
To: <sip:**[email protected]:5060>;tag=as3a12d869
Call-ID: [email protected]
CSeq: 684307914 ACK
User-Agent: P200 02.09
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
Content-Length: 0


10 headers, 0 lines
Destroying call '[email protected]'


Sip read: 
ACK sip:**[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK_00D0E9014CA1_T42A7F750
From: "P200@Asterisk" <sip:[email protected]:5060>;tag=00D0E9014CA1_T1218104444
To: <sip:**[email protected]:5060>;tag=as3a12d869
Call-ID: [email protected]
CSeq: 684307914 ACK
User-Agent: P200 02.09
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
Content-Length: 0


10 headers, 0 lines
Destroying call '[email protected]'
 
Wie ich es mir schon gedacht habe erwartet der snom-Server ein Passwort und da kannst du natuerlich nicht das richtige liefern. Da sollte snom evtl. mal den Server richtig konfigurieren. ;)
 
Um antworten zu können musst du eingeloggt sein.

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