If you are seeing nothing at all on the console with all that set, then the packets are never getting to FreeSWITCH. My first guess would be either firewall or bound to the wrong ip/port. Mike On Oct 16, 2008, at 9:27 AM, Gavin Henry wrote:
Hi All,
I'm trying to get a SIP forwarded call to do something with FS, i.e. go into a conference.
I can't even see anything getting rejected:
sofia status API CALL [sofia(status)] output: Name Type Data State = = = = = = = = = = = = = = = = = = = = = = = = = = = ====================================================================== internal profile sip:mod_sofia@87.X.X.X:5060 RUNNING (0) external profile sip:mod_sofia@87.X.X.X:5080 RUNNING (0) nat profile sip:mod_sofia@87.X.X.X:5070 RUNNING (0) default alias internal ALIASED pbx.XXXXX alias internal ALIASED outbound alias external ALIASED = = = = = = = = = = = = = = = = = = = = = = = = = = = ====================================================================== 3 profiles 3 aliases
The sip request is coming fine, no firewall issues.
pbx.XXXXX :/usr/local/freeswitch/conf# tcpdump -i eth0 -n -s0 -v udp port 5060 tcpdump: listening on eth0, link-type EN10MB (Ethernet), capture size 65535 bytes 14:08:34.464662 IP (tos 0x0, ttl 58, id 37907, offset 0, flags [DF], proto: UDP (17), length: 890) 193.111.200.132.5060 > 87.X.X.X.5060: SIP, length: 862 INVITE sip:0XXXXXX@87.X.X.X SIP/2.0 Via: SIP/2.0/UDP 193.111.200.132:5060;branch=z9hG4bK2344219b;rport From: "0XXX" <sip:0XXX@193.111.200.132>;tag=as6f63bcf8 To: <sip:0XXX@87.X.X.X> Contact: <sip:0XXX4@193.111.200.132> Call-ID: 4a5d70d13c3e6f1b5d7c5791318c02cd@193.111.200.132 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 16 Oct 2008 13:14:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 317
v=0 o=root 20381 20381 IN IP4 193.111.200.132 s=session c=IN IP4 193.111.200.132 t=0 0 m=audio 14998 RTP/AVP 8 0 18 4 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - -
I've switched on all debugging with:
cat set_debug.sh #!/bin/bash export SOFIA_DEBUG=9 export NUA_DEBUG=9 export SOA_DEBUG=9 export NEA_DEBUG=9 export IPTSEC_DEBUG=9 export NTA_DEBUG=9 export TPORT_DEBUG=9 export TPORT_LOG=9 export TPORT_DUMP=/tmp/tport_sip.log export SU_DEBUG=9
Set:
sofia loglevel 9 console loglevel 9
Calls will only ever come in via 193.111.201.114 and I have an ACL for it:
2008-10-16 13:35:28 [NOTICE] switch_core.c:886 switch_load_network_lists() Adding 193.111.201.114/255.255.255.0 (allow) to list strict
I've come from Asterisk and I'm used to seeing a lot more info. What am I missing?
I haven't added anything for the conference to the dialplan or anything, but I just want to see *something* showing the inbound call.