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

[Michael Jerris]

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 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.
>