Problem mit eingehende 1und1 anrufe (SIP/2.0 404 Not Found?)

m.list · 15 Mrz 2006

Hallo,

hab ein problem mit eingehende Anrufe ueber 1und1 Nummern. Ausgehende Anrufe gehen jetzt, nach langen rumsuchen, klar.

Wenn ich meine 1und1 Nummer waehle dann hoere ich die Ansage "Kein anschluss unter dieser Nummer", sehe aber mit sip debug ip das der Anruf bei Asterisk ankommt aber es kommt nicht zum ausfuehren der Dial Plan.

Bin kein sip protocol experte aber die einzige zwei fehler meldungen die is sehe sind

"Ignoring this request"
"SIP/2.0 404 Not Found"

Sip.conf:

Code:

[general]
port = 5060
nat=yes
localnet=192.xx.xx.xx/255.255.255.0
externip=xxx-xxx.dyndns.org
bindaddr = 0.0.0.0
useragent=corny


disallow=all
allow=ulaw
allow=alaw
allow=gsm

canreinvite=no
context = from-sip

register => 49xxxxxxxxxx:[email protected]/49xxxxxxxxxx

[1und1-17]
type=peer
username=49xxxxxxxxxx
secret=geheim
host=sip.1und1.de
fromuser=49xxxxxxxxxx
fromdomain=1und1.de
nat=no
canreinvite=no
qualify=yes
insecure=very

extensions.conf

Code:

[general]

static=yes
writeprotect=yes

[globals]

FAMILY-VM=20@family-vm


;********************************************************************************
; [from-sip]
;
;********************************************************************************
[from-sip]
exten => 49xxxxxxxx17,1,Answer
exten => 49xxxxxxxx17,2,Wait(1)
exten => 49xxxxxxxx17,3,DateTime()
exten => 49xxxxxxxx17,4,Hangup

sip debug ip xx.xx.xx.xx

Code:

Sip read:
INVITE sip:[email protected] SIP/2.0
Record-Route: <sip:212.227.15.225;ftag=1144642072;lr=on>
Via: SIP/2.0/UDP 212.227.15.197;branch=z9hG4bKe09e7a9fe14739d519d9062b6af961b5
Via: SIP/2.0/UDP 212.227.15.225;branch=z9hG4bKb0fb.13045da.0
Via: SIP/2.0/UDP 212.227.15.197;branch=z9hG4bKf217d0b4d89a806dd06472691af158a6
Via: SIP/2.0/UDP sipgw01.bmcag.com:5060 ;received=62.206.6.140;branch=z9hG4bKterm-143981-49xxxxxxx35-+49xxxxxxxx17
From: 49xxxxxxx35 <sip:[email protected];user=phone>;tag=1144642072
To: +49xxxxxxxx17 <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Supported: timer
Session-Expires: 1800
Min-SE: 1800
Contact:  <sip:[email protected]:5060>
Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE
Max-Forwards: 14
Content-Type: application/sdp
Content-Length: 345
Diversion: <sip:[email protected];user=phone>;reason=additional

v=0
o=- 1777794 0 IN IP4 62.206.6.202
s=Cisco SDP 0
c=IN IP4 62.206.6.202
t=0 0
m=audio 19330 RTP/AVP 8 0 2 99 18 110
a=rtpmap:99 G726-24/8000
a=rtpmap:110 X-NSE/8000
a=fmtp:110 192-194,200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 110
a=X-cpar: a=rtpmap:110 X-NSE/8000
a=X-cpar: a=fmtp:110 192-194,200-202
a=X-cap: 2 image udptl t38

19 headers, 14 lines
Using latest request as basis request
Sending to 212.227.15.197 : 5060 (NAT)
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 99
Found RTP audio format 18
Found RTP audio format 110
Peer audio RTP is at port 62.206.6.202:19330
Found description format G726-24
Found description format X-NSE
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x31c (ulaw|alaw|g726|g729|speex)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing)
Found peer '1und1-17'
Looking for 49xxxxxxxx17 in from-sip
RDNIS is +49xxxxxxxx17
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 212.227.15.197;branch=z9hG4bKe09e7a9fe14739d519d9062b6af961b5
Via: SIP/2.0/UDP 212.227.15.225;branch=z9hG4bKb0fb.13045da.0
Via: SIP/2.0/UDP 212.227.15.197;branch=z9hG4bKf217d0b4d89a806dd06472691af158a6
Via: SIP/2.0/UDP sipgw01.bmcag.com:5060 ;received=62.206.6.140;branch=z9hG4bKterm-143981-49xxxxxxx35-+49xxxxxxxx17
From: 49xxxxxxx35 <sip:[email protected];user=phone>;tag=1144642072
To: +49xxxxxxxx17 <sip:[email protected];user=phone>;tag=as6dd45d3c
Call-ID: [email protected]
CSeq: 1 INVITE
User-Agent: corny
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
Content-Length: 0


 to 212.227.15.197:5060
cellar*CLI>

Sip read:
INVITE sip:[email protected] SIP/2.0
Record-Route: <sip:212.227.15.225;ftag=1144642072;lr=on>
Via: SIP/2.0/UDP 212.227.15.197;branch=z9hG4bK68696f4a446a65b3f397306985059cc7
Via: SIP/2.0/UDP 212.227.15.225;branch=z9hG4bKb0fb.13045da.1
Via: SIP/2.0/UDP 212.227.15.197;branch=z9hG4bKf217d0b4d89a806dd06472691af158a6
Via: SIP/2.0/UDP sipgw01.bmcag.com:5060 ;received=62.206.6.140;branch=z9hG4bKterm-143981-49xxxxxxx35-+49xxxxxxxx17
From: 49xxxxxxx35 <sip:[email protected];user=phone>;tag=1144642072
To: +49xxxxxxxx17 <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Supported: timer
Session-Expires: 1800
Min-SE: 1800
Contact:  <sip:[email protected]:5060>
Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE
Max-Forwards: 14
Content-Type: application/sdp
Content-Length: 345
Diversion: <sip:[email protected];user=phone>;reason=additional

v=0
o=- 1777794 0 IN IP4 62.206.6.202
s=Cisco SDP 0
c=IN IP4 62.206.6.202
t=0 0
m=audio 19330 RTP/AVP 8 0 2 99 18 110
a=rtpmap:99 G726-24/8000
a=rtpmap:110 X-NSE/8000
a=fmtp:110 192-194,200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 110
a=X-cpar: a=rtpmap:110 X-NSE/8000
a=X-cpar: a=fmtp:110 192-194,200-202
a=X-cap: 2 image udptl t38

19 headers, 14 lines
Ignoring this request
cellar*CLI>

Sip read:
Max-Forwards: 10
Via: SIP/2.0/UDP 212.227.15.197;branch=z9hG4bKe09e7a9fe14739d519d9062b6af961b5
Via: SIP/2.0/UDP 212.227.15.225;branch=z9hG4bKb0fb.13045da.0
From: 49xxxxxxx35 <sip:[email protected];user=phone>;tag=1144642072
Call-ID: [email protected]
To: +49xxxxxxxx17 <sip:[email protected];user=phone>;tag=as6dd45d3c
CSeq: 1 ACK
User-Agent: UI OpenSer
Content-Length: 0


10 headers, 0 lines
Destroying call '[email protected]'
cellar*CLI>

Sip read:
INVITE sip:[email protected] SIP/2.0
Record-Route: <sip:212.227.15.225;ftag=1144642072;lr=on>
Via: SIP/2.0/UDP 212.227.15.197;branch=z9hG4bK68696f4a446a65b3f397306985059cc7
Via: SIP/2.0/UDP 212.227.15.225;branch=z9hG4bKb0fb.13045da.1
Via: SIP/2.0/UDP 212.227.15.197;branch=z9hG4bKf217d0b4d89a806dd06472691af158a6
Via: SIP/2.0/UDP sipgw01.bmcag.com:5060 ;received=62.206.6.140;branch=z9hG4bKterm-143981-49xxxxxxx35-+49xxxxxxxx17
From: 49xxxxxxx35 <sip:[email protected];user=phone>;tag=1144642072
To: +49xxxxxxxx17 <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Supported: timer
Session-Expires: 1800
Min-SE: 1800
Contact:  <sip:[email protected]:5060>
Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE
Max-Forwards: 14
Content-Type: application/sdp
Content-Length: 345
Diversion: <sip:[email protected];user=phone>;reason=additional

v=0
o=- 1777794 0 IN IP4 62.206.6.202
s=Cisco SDP 0
c=IN IP4 62.206.6.202
t=0 0
m=audio 19330 RTP/AVP 8 0 2 99 18 110
a=rtpmap:99 G726-24/8000
a=rtpmap:110 X-NSE/8000
a=fmtp:110 192-194,200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 110
a=X-cpar: a=rtpmap:110 X-NSE/8000
a=X-cpar: a=fmtp:110 192-194,200-202
a=X-cap: 2 image udptl t38

19 headers, 14 lines
Using latest request as basis request
Sending to 212.227.15.197 : 5060 (NAT)
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 99
Found RTP audio format 18
Found RTP audio format 110
Peer audio RTP is at port 62.206.6.202:19330
Found description format G726-24
Found description format X-NSE
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x31c (ulaw|alaw|g726|g729|speex)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing)
Found peer '1und1-17'
Looking for 788517 in from-sip
RDNIS is +49xxxxxxxx17
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 212.227.15.197;branch=z9hG4bK68696f4a446a65b3f397306985059cc7
Via: SIP/2.0/UDP 212.227.15.225;branch=z9hG4bKb0fb.13045da.1
Via: SIP/2.0/UDP 212.227.15.197;branch=z9hG4bKf217d0b4d89a806dd06472691af158a6
Via: SIP/2.0/UDP sipgw01.bmcag.com:5060 ;received=62.206.6.140;branch=z9hG4bKterm-143981-49xxxxxxx35-+49xxxxxxxx17
From: 49xxxxxxx35 <sip:[email protected];user=phone>;tag=1144642072
To: +49xxxxxxxx17 <sip:[email protected];user=phone>;tag=as667cb319
Call-ID: [email protected]
CSeq: 1 INVITE
User-Agent: corny
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
Content-Length: 0


 to 212.227.15.197:5060
cellar*CLI>

Sip read:
ACK sip:[email protected] SIP/2.0
Max-Forwards: 10
Via: SIP/2.0/UDP 212.227.15.197;branch=z9hG4bK68696f4a446a65b3f397306985059cc7
Via: SIP/2.0/UDP 212.227.15.225;branch=z9hG4bKb0fb.13045da.1
From: 49xxxxxxx35 <sip:[email protected];user=phone>;tag=1144642072
Call-ID: [email protected]
To: +49xxxxxxxx17 <sip:[email protected];user=phone>;tag=as667cb319
CSeq: 1 ACK
User-Agent: UI OpenSer
Content-Length: 0


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

merkc · 2 Apr 2006

lies mal den kurs zuerst durch.... dann mit dem debugging beginnen.
ps: du hast im sip.conf nix definiert, wohin der eingehende anruf gehen soll... sprich du brauchst noch ein [49xxxx...] usw.

betateilchen · 2 Apr 2006

Code:
Code:

Looking for 788517 in from-sip

Das Problem ist, daß der 1und1 Server den Anruf mit der Kennung 788517 an Deinen Asterisk schickt. Das mußt Du auch in der extensions.conf behandeln. Da es hierfür keinen Eintrag gibt, wird der Anruf mit 404 abgebrochen.

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Problem mit eingehende 1und1 anrufe (SIP/2.0 404 Not Found?)

m.list

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betateilchen

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