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[gavin.henry at gmail.com]

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.

Thanks.

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[mike at jerris.com]

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:

[gavin.henry at gmail.com]

2008/10/16 Michael Jerris <mike@jerris.com>: