Trixbox kann Gespräch nicht weitervermitteln

[cohibnig]

Hallo Leute!

Ich habe folgendes Problem. Ich habe mir einen Sipgate Account angelegt mit folgenden Einstellungen:

sip.conf:

; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn't, try adding "nat=1" to each peer definition to
; solve translation problems.

[general]

port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
language=de
qualify=no
allow=alaw
allow=g729
allow=gsm
allow=slinear
srvlookup=yes
canreinvite=yes
; If you need to answer unauthenticated calls, you should change this
; next line to 'from-trunk', rather than 'from-sip-external'.
; You'll know this is happening if when you call in you get a message
; saying "The number you have dialed is not in service. Please check the
; number and try again."
context=from-sip-external ; Send unknown SIP callers to this context
callerid=Unknown
tos=0x68

; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf


sip_additional.conf


register=326xxxx:[email protected]/326xxxx

[10]
username=10
type=friend
secret=xxx
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=never
mailbox=10@device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=xxx <10>

[20]
username=20
type=friend
secret=xxx
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=never
mailbox=20@device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=xxx <20>

[Sipgate_outs]
username=326xxxx
type=friend
secret=xxxxxxxx
qualify=yes
nat=yes
insecure=very
host=sipgate.at
fromuser=326xxxx
fromdomain=sipgate.at


Untereinander ist es kein Problem zu telefonieren. Rufe ich nach außen, gehen auch, zwar läutet es bei mir länger, bevor der andere ein Klingenl hört. Nur erreichbar bin ich nicht. Es kommt immer, daß das Gespräch nicht vermittelt werden kann.

Genauer:


-- Executing NoOp("SIP/326xxxx-22cc", "Received incoming SIP connection from unknown peer to 326xxxx") in new stack
-- Executing Set("SIP/326xxxx-22cc", "DID=326xxxx") in new stack
-- Executing Goto("SIP/326xxxx-22cc", "s|1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing GotoIf("SIP/326xxxx-22cc", "0?from-trunk|326xxxx|1") in new stack
-- Executing Set("SIP/326xxxx-22cc", "TIMEOUT(absolute)=15") in new stack
-- Channel will hangup at 2006-08-27 07:36:37 UTC.
-- Executing Answer("SIP/326xxxx-22cc", "") in new stack
-- Executing Wait("SIP/326xxxx-22cc", "2") in new stack
-- Executing Playback("SIP/326xxxx-22cc", "ss-noservice") in new stack
-- Playing 'ss-noservice' (language 'de')
-- Executing NoOp("SIP/326xxxx-22cc", "Hangup") in new stack
-- Executing Set("SIP/326xxxx-22cc", "DID=s") in new stack
-- Executing Goto("SIP/326xxxx-22cc", "s|1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing GotoIf("SIP/326xxxx-22cc", "0?from-trunk|s|1") in new stack
-- Executing Set("SIP/326xxxx-22cc", "TIMEOUT(absolute)=15") in new stack
-- Channel will hangup at 2006-08-27 07:36:45 UTC.
-- Executing Answer("SIP/326xxxx-22cc", "") in new stack




asterisk1*CLI> sip show registry
Host Username Refresh State
sipgate.at:5060 326xxxx 105 Registered


asterisk1*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
Sipgate_outs/326xxxx 217.116.119.252 N 5060 OK (56 ms)
20/20 192.168.0.131 D 5060 Unmonitored
10/10 192.168.0.132 D 5060 Unmonitored
3 sip peers [3 online , 0 offline]

Die Trixbox nimmt das Gespräch an, da eine Gebühr beim Anrufenden angezeigt wird. Nur er leitet das Gespräch nicht zum Telefon weiter.

Kann mir wer weiterhelfen.
Danke.

Gruß

Juergen