David Schmitt
2006-Jul-27 02:40 UTC
[asterisk-users] Malformed/Missing URL Problem with Cisco Callmanager 4.1
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@10.200.16.72>;tag=as027c0ecb To: <sip:callmanagertest.firm.country> Call-ID: 16e90f962661be6c29731b3b28af3067@10.200.16.72 CSeq: 102 OPTIONS Content-Length: 0 --- (7 headers 0 lines)--- Destroying call '16e90f962661be6c29731b3b28af3067@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