Asterisk 1.6 und Sipgate Live

[Lazybone]

Hallo,

seit update auf Asterisk 1.6.2 funktioniert mein Sipgate Live zugang nicht mehr es kommt nur

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: OPTIONS
[Mar 21 21:35:49] NOTICE[7054]: chan_sip.c:11049 sip_reregister:    -- Re-registration for  [email protected]
       > doing dnsmgr_lookup for 'sipconnect.sipgate.de'
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 217.10.68.150:5060:
REGISTER sip:sipgate.de SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK7e8f7efe;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as2068cc26
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 104 REGISTER
User-Agent: Asterisk PBX 1.6.2.0~rc2-0ubuntu1.2
Expires: 120
Contact: <sip:[email protected]>
Content-Length: 0
dmveb2*CLI>

---
dmveb2*CLI>
<--- SIP read from UDP:217.10.68.150:5060 --->
SIP/2.0 404 not found (unknown domain)
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK7e8f7efe;rport=5060
From: <sip:[email protected]>;tag=as2068cc26
To: <sip:[email protected]>;tag=0648aaddc9a6dfd17d4c74b60bcf0df2.ff6b
Call-ID: [email protected]
CSeq: 104 REGISTER
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
[Mar 21 21:35:49] WARNING[7054]: chan_sip.c:17574 handle_response_register: Got 404 Not found on SIP register to service [email protected], giving up
Really destroying SIP dialog '[email protected]' Method: REGISTER

sip.conf

register => xxxxxxxx:[email protected]/xxxxxxxx ;Digitmedia Trunk

[sipgate.de]
type = friend
host = sipconnect.sipgate.de
outboundproxy=sipconnect.live.sipgate.de
port = 5060
username = xxxxxxxxxx
fromuser = xxxxxxxxxx
fromdomain = sipconnect.sipgate.de
secret = xxxxxxx
dtmfmode = rfc2833
insecure = port,invite
canreinvite = yes
registertimeout = 600
disallow=all
allow=alaw
allow=ulaw
context=ankommend

Woran könnte dies liegen? Bei Asterisk 1.4 funktioniert alles wunderbar.

Gruß
Lazybone

 

[Netview]

Einstellungen genau übernehmen insbesondere @sipconnect Verweis auf context [sipconnect]!

Ein canreinvit hat übrigens im peer nichts zu suchen, es sei denn, asterisk soll außen vor bleiben!

register => user:password[b]@sipconnect[/b]/sipconnect

[b][sipconnect][/b] 
type=peer
defaultuser=user
secret=password
host=sipconnect.sipgate.de
fromuser=user
fromdomain=sipconnect.sipgate.de
outboundproxy=sipconnect.live.sipgate.de
canreinvite=no
qualify=no
allow=alaw
allow=ulaw
insecure=port,invite
nat=no
dtmfmode=rfc2833