hi there,
i want to use freepbx in combination with my lancom 1823 voip router. the router is connected via isdn to the cable network. i want to configure freepbx so that it uses the two isdn channels for outgoing/incoming calls. furthermore, i want to use additional voip trunks configured at feepbx for outgoing calls.
lancom offers a tutorial (unfortuneately in german, but the asterisk syntax shall be understandable) http://www2.lancom.de/kb.nsf/b8f10fe5665f950dc125726c00589d94/e1759294d6f0c6e0c12573e60050acb4?OpenDocument
now my questions:
-
freepbx uses mysql instead of the config files. where can i setup all the settings, e.g. extensions.conf?
-
i have created a user to connect freepbx with lancom router. the outbound route of freepbx doesn’t work, i always get a line is busy signal. a trace of the call is attached.
----- snip
SIP-Packet] 2009/11/27 01:18:33,010 [PACKET] :
Receiving datagram with length 824 from 192.168.1.66:5060 to 192.168.1.200:5060
INVITE sip:[email protected];user=phone SIP/2.0\r\n
Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bK435e1a38;rport\r\n
From: “44” sip:44@intern;tag=as0cceadf6\r\n
To: sip:[email protected];user=phone\r\n
Contact: sip:[email protected]\r\n
Call-ID: 3bb851e874950911018f6c26082a5453@intern\r\n
CSeq: 102 INVITE\r\n
User-Agent: Asterisk PBX\r\n
Max-Forwards: 70\r\n
Date: Fri, 27 Nov 2009 07:13:54 GMT\r\n
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY\r\n
Supported: replaces\r\n
Content-Type: application/sdp\r\n
Content-Length: 285\r\n
\r\n
v=0\r\n
o=root 3149 3149 IN IP4 192.168.1.66\r\n
s=session\r\n
c=IN IP4 192.168.1.66\r\n
t=0 0\r\n
m=audio 14478 RTP/AVP 0 8 3 101\r\n
a=rtpmap:0 PCMU/8000\r\n
a=rtpmap:8 PCMA/8000\r\n
a=rtpmap:3 GSM/8000\r\n
a=rtpmap:101 telephone-event/8000\r\n
a=fmtp:101 0-16\r\n
a=silenceSupp:off - - - -\r\n
a=ptime:20\r\n
a=sendrecv\r\n
[SIP-Packet] 2009/11/27 01:18:33,020 [PACKET] :
Sending datagram with length 434 from 192.168.1.200:5060 to 192.168.1.66:5060
SIP/2.0 404 Not Found\r\n
Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bK435e1a38;rport\r\n
From: "44"sip:44@intern;tag=as0cceadf6\r\n
To: sip:[email protected];user=phone\r\n
Call-ID: 3bb851e874950911018f6c26082a5453@intern\r\n
CSeq: 102 INVITE\r\n
Max-Forwards: 70\r\n
User-Agent: LANCOM 1823 VoIP (Annex B) / 7.70.0100 / 18.08.2009\r\n
Server: lancom\r\n
Allow: REGISTER, INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, OPTIONS\r\n
Content-Length: 0\r\n
\r\n
[SIP-Packet] 2009/11/27 01:18:33,020 [PACKET] :
Receiving datagram with length 374 from 192.168.1.66:5060 to 192.168.1.200:5060
ACK sip:[email protected];user=phone SIP/2.0\r\n
Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bK435e1a38;rport\r\n
From: “44” sip:44@intern;tag=as0cceadf6\r\n
To: sip:[email protected];user=phone\r\n
Contact: sip:[email protected]\r\n
Call-ID: 3bb851e874950911018f6c26082a5453@intern\r\n
CSeq: 102 ACK\r\n
User-Agent: Asterisk PBX\r\n
Max-Forwards: 70\r\n
Content-Length: 0\r\n
\r\n[TraceStopped] 2009/11/27 09:10:53,000
-----snip
if i put the user 44 credentials in a sip phone, it works -> config on lancom side seems to work.
Any suggestions why i cannot make an outgoing call from a phone logged in at the freepbx using the lancom outgoing route?
christian