[Freeswitch-users] Can't see any Sofia messages

[Gavin Henry]

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 at 87.X.X.X:5060
    RUNNING (0)
                 external       profile   sip:mod_sofia at 87.X.X.X:5080
    RUNNING (0)
                      nat       profile   sip:mod_sofia at 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 at 87.X.X.X SIP/2.0
        Via: SIP/2.0/UDP 193.111.200.132:5060;branch=z9hG4bK2344219b;rport
        From: "0XXX" <sip:0XXX at 193.111.200.132>;tag=as6f63bcf8
        To: <sip:0XXX at 87.X.X.X>
        Contact: <sip:0XXX4 at 193.111.200.132>
        Call-ID: 4a5d70d13c3e6f1b5d7c5791318c02cd at 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.