[asterisk-users] Malformed/Missing URL Problem with Cisco Callmanager 4.1

[David Schmitt]

Hi

I want to use Asterisk as a Voicemail Box for my Callmanager Users
The Link between Cisco Callmanager and Asterisk has to be SIP (according 
to 
http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration)
The Voicemail Part on Asterisk is running perfect via a IAX Softphone 
but not via the SIP Channel (SIP Trunk in Cisco words)
The Callmanager Box and the Asterisk Box are on the same Subnet/VLAN -> 
there is no Firewall or something else between them

I am always getting this Error on the Asterisk CLI :

<-- SIP read from 10.200.16.52:5060:
SIP/2.0 400 Bad Request - 'Malformed/Missing URL'
Via: SIP/2.0/UDP 10.200.16.72:5060;branch=z9hG4bK0b0171ec;rport
From: "asterisk" <sip:asterisk at 10.200.16.72>;tag=as027c0ecb
To: <sip:callmanagertest.firm.country>
Call-ID: 16e90f962661be6c29731b3b28af3067 at 10.200.16.72
CSeq: 102 OPTIONS
Content-Length: 0


--- (7 headers 0 lines)---
Destroying call '16e90f962661be6c29731b3b28af3067 at 10.200.16.72'

Asterisk Versions I tried : 1.2.7 - 1.2.10
Callmanager Versions I tried : 4.1 - 4.2.1sr1a

Changing the Version of Asterisk or Callmanager doesn't help.
So I think the Problem is in my Asterisk SIP Trunk Configuration.

At the moment the configuration looks like :

[general]
context=default
allowguest=no
realm=tds.de
bindport=5060
bindaddr=10.200.16.72
srvlookup=no
autodomain=yes
domain=firm.country
domain=10.200.16.52
vmexten=voicemail
videosupport=no
disallow=all
allow=ulaw
allow=alaw
relaxdtmf=yes
rtptimeout=60
rtpholdtimeout=300
useragent=Asterisk
dtmfmode=rfc2833
sipdebug=yes
notifyringing=yes

[default]
include => callmanager2-1
include => callmanager2-2

[callmanager2-1]
type=friend
context=default
host=callmanagertest.firm.country
dtmfmode=rfc2833
port=5060
insecure=port,invite
disallow=all
allow=ulaw
allow=alaw
nat=no
canreinvite=yes
username=phone
fromuser=phone
qualify=yes

[callmanager2-2]
type=friend
context=default
host=callmanagertest.firm.country
dtmfmode=rfc2833
port=5060
insecure=port,invite
disallow=all
allow=ulaw
allow=alaw
nat=no
canreinvite=yes
username=phone
fromuser=phone
qualify=yes



Has anyone any Idea ? :)  or perhaps some Sample Configuration Files of 
such a scenario ?

Many thanks
David