backbone.ch / oneweb calls kommen nicht durch aber an

[onemarkus]

--- (0 headers 0 lines) Nat keepalive ---
debian*CLI>
<-- SIP read from 212.117.200.148:5060:
INVITE sip:[email protected] SIP/2.0
Record-Route: <sip:212.117.200.148;ftag=16c2f114b15b8647810a5a8c2e3ed609;lr>
Via: SIP/2.0/UDP 212.117.200.148;branch=z9hG4bKa57b.d69cdf36a003bca26d1c9c0ec4e0a677.0
Via: SIP/2.0/UDP 212.117.200.148:5061;branch=z9hG4bK872de7ffc07688d2eee02d724244cf54;rport=5061
Max-Forwards: 16
From: <sip:[email protected]>;tag=16c2f114b15b8647810a5a8c2e3ed609
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 200 INVITE
Contact: Anonymous <sip:212.117.200.148:5061>
Expires: 300
User-Agent: Sippy
cisco-GUID: 1554835545-3932951002-2355397326-190043270
h323-conf-id: 1554835545-3932951002-2355397326-190043270
Content-Length: 395
Content-Type: application/sdp

v=0
o=CiscoSystemsSIP-GW-UserAgent 2363 2990 IN IP4 212.117.200.80
s=SIP Call
t=0 0
m=audio 16644 RTP/AVP 0 8 18 2 98 99 3 101
c=IN IP4 212.117.200.80
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:98 G726-24/8000
a=rtpmap:99 G726-16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

--- (16 headers 16 lines)---
Using INVITE request as basis request - [email protected]
Sending to 212.117.200.148 : 5060 (non-NAT)
Found peer '41325117466'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 98
Found RTP audio format 99
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 212.117.200.80:16644
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format G726-32
Found description format G726-24
Found description format G726-16
Found description format GSM
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x11e (gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 41325117466 in default (domain 83.216.218.3)
Reliably Transmitting (no NAT) to 212.117.200.148:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 212.117.200.148;branch=z9hG4bKa57b.d69cdf36a003bca26d1c9c0ec4e0a677.0;received=212.117.200.148
Via: SIP/2.0/UDP 212.117.200.148:5061;branch=z9hG4bK872de7ffc07688d2eee02d724244cf54;rport=5061
From: <sip:[email protected]>;tag=16c2f114b15b8647810a5a8c2e3ed609
To: <sip:[email protected]>;tag=as56ea7bb4
Call-ID: [email protected]
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0


---
debian*CLI>
<-- SIP read from 212.117.200.148:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 212.117.200.148;branch=z9hG4bKa57b.d69cdf36a003bca26d1c9c0ec4e0a677.0
From: <sip:[email protected]>;tag=16c2f114b15b8647810a5a8c2e3ed609
Call-ID: [email protected]
To: <sip:[email protected]>;tag=as56ea7bb4
CSeq: 200 ACK
User-Agent: Sip EXpress router (0.9.4 (i386/freebsd))
Content-Length: 0


--- (8 headers 0 lines)---
Destroying call '[email protected]'
debian*CLI>
<-- SIP read from 212.236.16.118:1029:

Wenn ich mit meinem Handy meine CH Nummer Anrufe kommt das die Nummer nicht vergeben ist.

Das ist der log vom Asterisk Server

 

[onemarkus]

BEHOBEN ---> Hatte falschen IN TAG

 

[betateilchen]

Schau mal in Dein SIP Log - da steht:

Looking for [COLOR="Red"]41325117466[/COLOR] in default (domain 83.216.218.3)

und 41325117466 wird in Deiner sip.conf nicht gefunden...

Reliably Transmitting (no NAT) to 212.117.200.148:5060:
[COLOR="Red"]SIP/2.0 404 Not Found[/COLOR]

edit - du warst mit deiner Fehlerbehebung schneller als ich mit meiner Erklärung :wink: