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:[EMAIL PROTECTED]:5060
RUNNING (0)
external profile sip:[EMAIL PROTECTED]:5080
RUNNING (0)
nat profile sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 193.111.200.132:5060;branch=z9hG4bK2344219b;rport
From: "0XXX" <sip:[EMAIL PROTECTED]>;tag=as6f63bcf8
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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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