Probleme mit Dialin Nummer von didww.com und Asterisk

[Chimbo2004]

Ich habe bei Didww.com 2 Dids (1 Buenos Aires und eine Caracas/Venezuela)
bis gestern haben beide anstandslos gearbeitet,und seit gestern funzt die DID
aus Caracas nicht mehr obwohl ich an der Konfiguration nichts gemacht habe.

Die sip.conf schaut so aus:

[didww]
type=peer
host=204.11.194.34
context=didww-ven


[didwwargentina]
type=peer
host=204.11.194.38
context=didww-arg

extensions.conf schaut so aus:

[didww-ven]
exten =>5821233xxxx,1,NoOp(${CALLERID(num)})
exten =>5821233xxxx,n,Goto(micha,${EXTEN},1)
exten =>5821233xxxx,n,Hangup

[didww-arg]
exten =>5411525xxx,1,NoOp(${CALLERID(num)})
exten =>5411525xxx,n,Goto(micha,${EXTEN},1)
exten =>5411525xxx,n,Hangup

Die argentinsche wie gesagt ohne Probleme und wenn auf der venezolanischen angerufen wird dann es sieht es mit dem sip debug so aus:

Reliably Transmitting (NAT) to 190.254.147.220:5060:
OPTIONS sip:[email protected]:5060;transport=udp;user=phone SIP/2.0
Via: SIP/2.0/UDP 82.197.159.207:5060;branch=z9hG4bK0e3c87cb;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as5e0829c9
To: <sip:[email protected]:5060;transport=udp;user=phone>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: chimbo2004
Date: Sun, 25 Apr 2010 01:14:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
v90460*CLI>
<--- SIP read from UDP://190.254.147.220:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.197.159.207:5060;branch=z9hG4bK0e3c87cb;rport
From: "asterisk" <sip:[email protected]>;tag=as5e0829c9
To: <sip:[email protected]:5060;transport=udp;user=phone>;tag=405c194fcf3fca3c
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Grandstream BT200 1.1.6.32
Contact: <sip:[email protected]:5060;transport=udp;user=phone>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Supported: replaces, timer
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: OPTIONS
v90460*CLI>
<--- SIP read from UDP://204.11.194.38:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 204.11.194.38:5060;branch=z9hG4bK3aaea210;rport
From: "1102268800" <sip:[email protected]>;tag=as2e35e7c0
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: DIDWW
Max-Forwards: 70
Remote-Party-ID: "1102268800" <sip:[email protected]>;privacy=off;screen=no
Date: Sun, 25 Apr 2010 00:12:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 411

v=0
o=root 27126 27126 IN IP4 204.11.194.38
s=session
c=IN IP4 204.11.194.38
t=0 0
m=audio 19834 RTP/AVP 0 8 18 4 111 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
--- (15 headers 19 lines) ---
  == Using SIP RTP CoS mark 5
Sending to 204.11.194.38 : 5060 (no NAT)
Using INVITE request as basis request - [email protected]
No user '1102268800' in SIP users list
Found peer 'didwwargentina' for '1102268800' from 204.11.194.38:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 111
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format G726-32 for ID 111
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=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=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 204.11.194.38:19834
Looking for 5821233xxxx in didww-arg (domain 82.197.159.207)
v90460*CLI>
<--- Reliably Transmitting (no NAT) to 204.11.194.38:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 204.11.194.38:5060;branch=z9hG4bK3aaea210;received=204.11.194.38;rport=5060
From: "1102268800" <sip:[email protected]>;tag=as2e35e7c0
To: <sip:[email protected]>;tag=as2eca6b82
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: chimbo2004
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
v90460*CLI>
<--- SIP read from UDP://204.11.194.38:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 204.11.194.38:5060;branch=z9hG4bK3aaea210;rport
From: "1102268800" <sip:[email protected]>;tag=as2e35e7c0
To: <sip:[email protected]>;tag=as2eca6b82
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: DIDWW
Max-Forwards: 70
Remote-Party-ID: "1102268800" <sip:[email protected]>;privacy=off;screen=no
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: ACK
v90460*CLI>
<--- SIP read from UDP://213.172.34.198:15222 --->

<------------->
v90460*CLI>

Liegt das Problem am Asterisk oder bei didww.com,die sagen naemlich es sei alles bestens.

Edit Guard-X: Bitte nächstes mal "Code"- statt "Quote"-Tags verwenden!