[Serusers] BYE message not recognized
Dawid Mielnik
D.Mielnik at elka.pw.edu.pl
Fri Feb 27 19:06:13 CET 2004
Hi,
I have a problem with SER rocognizing BYE messages sent from Asterisk -
which in turn leaves open connections in my accounting database.
My setup is as follows:
UA--NAT--(Internet)--SER--(Internet)--Asterisk--(PSTN)--POTS
sss.sss.ss.sss aaa.aaa.aa.aaa
My problem is when I place a call from the SIP UA to a PSTN phone through
Asterisk and the PSTN phone releases the connection. SER fails to recognize
the BYE message sent by Asterisk and does not put a 'stop time' entry into
my radius database. This does not happen when I call other SIP UA (with the
other UA also behind NAT).
Below I've attached:
1. ser.cfg for the call
2. asterisk sip debug log
3. sip trace from the UA machine
Any help appreciated here, without this I have no billing !
Thanks,
Dave
1. ser.cfg for the call:
----------------------------------------------------------------------------
-------------------------------------------------
#
# $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
#debug=3 # debug level (cmd line: -dddddddddd)
#fork=yes
#log_stderror=no # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
debug=9
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
#port=5060
#children=4
fifo="/tmp/ser_fifo"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
# accounting
loadmodule "/usr/local/lib/ser/modules/acc.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
# Nathelper
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
#modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# -- nathelper params --
modparam("registrar", "nat_flag", 6)
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
# -- acc params --
modparam("acc", "radius_config",
"/usr/local/etc/radiusclient/radiusclient.conf")
modparam("acc", "log_level", 1)
modparam("acc", "radius_flag", 1)
modparam("acc", "report_ack", 0)
# ------------------------- request routing logic -------------------
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
# sprawdzamy czy odzywa sie ktos z za natu....
if (nat_uac_test("3")) {
if (method == "REGISTER" || !search("^Record-Route:")) {
log("LOG: Kolejny NATowiec...\n");
fix_nated_contact();
if (method == "INVITE") {
fix_nated_sdp("1");
};
force_rport(); # dodaj do Via - topmost
setflag(6); # odznacz jako nated
};
};
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
record_route();
# loose-route processing
if (loose_route()) {
route(1); #t_relay();
break;
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
# if (uri=="aaa.aaa.aa.aaa") {
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
if (!www_authorize("sss.sss.ss.sss", "subscriber")) {
www_challenge("sss.sss.ss.sss", "0");
break;
};
save("location");
break;
};
setflag(1);
# native SIP destinations are handled using our USRLOC DB
# going to our sip users ?
if (uri=~"sip:326794*" || uri=~"sip:58279*") {
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
route(1);
# going to pstn
} else {
# };
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
# coming from fax ?
if (search("(f|From): .*3267940@*")) { # fax numbers
# forward to fax gw
rewritehostport("192.168.0.250:5060");
} else {
# forward to voice gw
rewritehostport("aaa.aaa.aa.aaa:5060");
};
if (!t_relay()) {
sl_reply_error();
};
};
# sprawdzamy czy wysylamy do natowanych abonentow
#setflag(1);
#route(1);
# if (!t_relay()) {
# sl_reply_error();
#};
}
route[1]
{
# if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" &&
!search("^Route:")){
# sl_send_reply("479", "We don't forward to private IP addresses");
# break;
# };
# jezeli za natem to uzywamy rtp relay
if (isflagset(6)){
force_rtp_proxy();
};
# natowe przetwarzanie odpowiedzi, ma sie do wszystkich transakcji
t_on_reply("1");
# nat zalatwiony wysylamy, stateful relaying
if (!t_relay()){
sl_reply_error();
};
}
onreply_route[1]
{
# nated ?
if (isflagset(6) && status =~ "(183)|2[0-9][0-9]"){
fix_nated_contact();
force_rtp_proxy();
# lub, jezeli transakcja jest natowana ale nie wiedzielismy o tym
# przetwarzajac zapytanie...
} else if (nat_uac_test("1")) {
fix_nated_contact();
};
}
2. asterisk sip debug log:
----------------------------------------------------------------------------
-------------------------------------------------
Found audio format UNKN
Found description format pcmu
Found description format pcma
Found description format speex
Found description format telephone-event
Capabilities: us - 270, them - 524/0, combined - 12
Non-codec capabilities: us - 1, them - 1, combined - 1
Looking for 0225827915 in default
list_route: hop: <sip:0225827915 at sss.sss.ss.sss;ftag=896605854;lr=on>
list_route: hop: <sip:3267915 at 80.55.21.254:1184>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP sss.sss.ss.sss;branch=z9hG4bKd56d.7b60dd2.0
Via: SIP/2.0/UDP
192.168.2.32:5060;received=80.55.21.254;rport=1184;branch=z9hG4bK3AB182BF553
74A818403FAF5511D2C7B
From: Dawid Mielnik <sip:3267915 at sss.sss.ss.sss>;tag=896605854
To: <sip:0225827915 at sss.sss.ss.sss>;tag=as37250f4f
Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473 at 192.168.2.32
CSeq: 58806 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:0225827915 at aaa.aaa.aa.aaa>
Content-Length: 0
to sss.sss.ss.sss:5060
-- Executing SetCallerID("SIP/-08161e80", "223267915") in new stack
-- Executing Dial("SIP/-08161e80", "Zap/g1/0225827915") in new stack
-- Called g1/0225827915
We're at aaa.aaa.aa.aaa port 10548
Answering with preferred capability 4
Answering with preferred capability 8
Answering with non-codec capability 1
Transmitting (no NAT):
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP sss.sss.ss.sss;branch=z9hG4bKd56d.7b60dd2.0
Via: SIP/2.0/UDP
192.168.2.32:5060;received=80.55.21.254;rport=1184;branch=z9hG4bK3AB182BF553
74A818403FAF5511D2C7B
From: Dawid Mielnik <sip:3267915 at sss.sss.ss.sss>;tag=896605854
To: <sip:0225827915 at sss.sss.ss.sss>;tag=as37250f4f
Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473 at 192.168.2.32
CSeq: 58806 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:0225827915 at aaa.aaa.aa.aaa>
Content-Type: application/sdp
Content-Length: 217
v=0
o=root 26423 26423 IN IP4 aaa.aaa.aa.aaa
s=session
c=IN IP4 aaa.aaa.aa.aaa
t=0 0
m=audio 10548 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
to sss.sss.ss.sss:5060
Feb 27 18:21:48 NOTICE[1236268096]: chan_zap.c:3587 zt_read: Fax detected,
but no fax extension
-- Zap/1-1 is making progress passing it to SIP/-08161e80
-- Zap/1-1 is ringing
11 headers, 0 lines
10 headers, 0 lines
-- Zap/1-1 answered SIP/-08161e80
We're at aaa.aaa.aa.aaa port 10548
Answering with preferred capability 4
Answering with preferred capability 8
Answering with non-codec capability 1
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP sss.sss.ss.sss;branch=z9hG4bKd56d.7b60dd2.0
Via: SIP/2.0/UDP
192.168.2.32:5060;received=80.55.21.254;rport=1184;branch=z9hG4bK3AB182BF553
74A818403FAF5511D2C7B
Record-Route: <sip:0225827915 at sss.sss.ss.sss;ftag=896605854;lr=on>
From: Dawid Mielnik <sip:3267915 at sss.sss.ss.sss>;tag=896605854
To: <sip:0225827915 at sss.sss.ss.sss>;tag=as37250f4f
Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473 at 192.168.2.32
CSeq: 58806 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:0225827915 at aaa.aaa.aa.aaa>
Content-Type: application/sdp
Content-Length: 217
v=0
o=root 26423 26424 IN IP4 aaa.aaa.aa.aaa
s=session
c=IN IP4 aaa.aaa.aa.aaa
t=0 0
m=audio 10548 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
to sss.sss.ss.sss:5060
asterisk*CLI>
Sip read:
ACK sip:0225827915 at aaa.aaa.aa.aaa:5060 SIP/2.0
Record-Route: <sip:0225827915 at sss.sss.ss.sss;ftag=896605854;lr=on>
Via: SIP/2.0/UDP sss.sss.ss.sss;branch=0
Via: SIP/2.0/UDP
192.168.2.32:5060;received=80.55.21.254;rport=1184;branch=z9hG4bK33D0B46E959
A49B2914EB72B18029B74
From: Dawid Mielnik <sip:3267915 at sss.sss.ss.sss>;tag=896605854
To: <sip:0225827915 at sss.sss.ss.sss>;tag=as37250f4f
Contact: <sip:3267915 at 80.55.21.254:1184>
Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473 at 192.168.2.32
CSeq: 58806 ACK
Max-Forwards: 69
Content-Length: 0
11 headers, 0 lines
-- Channel 1, span 1 got hangup
-- Hungup 'Zap/1-1'
== Spawn extension (default, 0225827915, 2) exited non-zero on
'SIP/-08161e80'
set_destination: Parsing
<sip:0225827915 at sss.sss.ss.sss;ftag=896605854;lr=on> for address/port to
send to
set_destination: set destination to sss.sss.ss.sss, port 5060
Reliably Transmitting:
BYE sip:3267915 at 80.55.21.254:1184 SIP/2.0
Via: SIP/2.0/UDP aaa.aaa.aa.aaa:5060;branch=z9hG4bK49c1001d
Route: <sip:3267915 at 80.55.21.254:1184>
From: <sip:0225827915 at sss.sss.ss.sss>;tag=as37250f4f
To: Dawid Mielnik <sip:3267915 at sss.sss.ss.sss>;tag=896605854
Contact: <sip:0225827915 at aaa.aaa.aa.aaa>
Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473 at 192.168.2.32
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to sss.sss.ss.sss:5060
asterisk*CLI>
Sip read:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP aaa.aaa.aa.aaa:5060;branch=z9hG4bK49c1001d
From: <sip:0225827915 at sss.sss.ss.sss>;tag=as37250f4f
To: Dawid Mielnik <sip:3267915 at sss.sss.ss.sss>;tag=896605854
Contact: <sip:3267915 at 192.168.2.32:5060>
Record-Route: <sip:0225827915 at sss.sss.ss.sss;ftag=896605854;lr=on>
Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473 at 192.168.2.32
CSeq: 102 BYE
Server: X-Lite build 1088
Content-Length: 0
10 headers, 0 lines
Message is BYE
3. sip trace from the UA machine:
----------------------------------------------------------------------------
-------------------------------------------------
Sip Scenario Trace
File: radius_calledstat_ends4
Generated: Fri Feb 27 18:32:43 2004
Traced on: Fri Feb 27 18:23:34 2004
Created
by:\\Mielonka\Techniczny\sip_project\test_tools\sip_scenario\sip_scenario.ex
e version=1.2.0
192.168.2.32:5060 sss.sss.ss.sss:5060
| | <Call><PFrame><Time>
| |
|>F1 INVITE (sdp)-------------------------------->| 1 PF:224 18:23:51.4621
| |
|<- trying -- your call is important to us 100 F2<| 1 PF:227 18:23:51.5577
| |
|<------------------(sdp) Session Progress 183 F3<| 1 PF:233 18:23:51.5942
| |
|>F4 (sip incomplete) >>>------------------------>| 1 PF:1097 18:24:1.2413
| |
|<--------------------------------(sdp) OK 200 F5<| 1 PF:1407 18:24:4.5217
| |
|>F6 ACK ---------------------------------------->| 1 PF:1410 18:24:4.5381
| |
|<---------------------------------------- BYE F7<| 1 PF:1542 18:24:5.9842
| |
|>F8 200 Ok ------------------------------------->| 1 PF:1543 18:24:5.9935
| |
|<--------------------------<<< (sip fragment) F9<| 2 PF:2500 18:26:9.9414
============================================================================
====
SIP MESSAGE 1 192.168.2.32:5060(1) -> sss.sss.ss.sss:5060(2)
UDP Frame 224 27/Feb/04 18:23:51.4621
TimeFromPreviousSipFrame=17.4501 TimeFromStart=17.4501
INVITE sip:0225827915 at sss.sss.ss.sss SIP/2.0
Via: SIP/2.0/UDP
192.168.2.32:5060;rport;branch=z9hG4bK3AB182BF55374A818403FAF5511D2C7B
From: Dawid Mielnik <sip:3267915 at sss.sss.ss.sss>;tag=896605854
To: <sip:0225827915 at sss.sss.ss.sss>
Contact: <sip:3267915 at 192.168.2.32:5060>
Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473 at 192.168.2.32
CSeq: 58806 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite build 1088
Content-Length: 247
v=0
o=3267915 27731078 27731078 IN IP4 192.168.2.32
s=X-Lite
c=IN IP4 192.168.2.32
t=0 0
m=audio 8000 RTP/AVP 0 8 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
============================================================================
====
SIP MESSAGE 2 sss.sss.ss.sss:5060(2) -> 192.168.2.32:5060(1)
UDP Frame 227 27/Feb/04 18:23:51.5577
TimeFromPreviousSipFrame=0.0956 TimeFromStart=17.5457
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP
192.168.2.32:5060;rport=1184;branch=z9hG4bK3AB182BF55374A818403FAF5511D2C7B;
received=80.55.21.254
From: Dawid Mielnik <sip:3267915 at sss.sss.ss.sss>;tag=896605854
To: <sip:0225827915 at sss.sss.ss.sss>
Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473 at 192.168.2.32
CSeq: 58806 INVITE
Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux))
Content-Length: 0
Warning: 392 sss.sss.ss.sss:5060 "Noisy feedback tells: pid=7250
req_src_ip=80.55.21.254 req_src_port=1184
in_uri=sip:0225827915 at sss.sss.ss.sss
out_uri=sip:0225827915 at aaa.aaa.aa.aaa:5060 via_cnt==1"
============================================================================
====
SIP MESSAGE 3 sss.sss.ss.sss:5060(2) -> 192.168.2.32:5060(1)
UDP Frame 233 27/Feb/04 18:23:51.5942
TimeFromPreviousSipFrame=0.0365 TimeFromStart=17.5822
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
192.168.2.32:5060;received=80.55.21.254;rport=1184;branch=z9hG4bK3AB182BF553
74A818403FAF5511D2C7B
From: Dawid Mielnik <sip:3267915 at sss.sss.ss.sss>;tag=896605854
To: <sip:0225827915 at sss.sss.ss.sss>;tag=as37250f4f
Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473 at 192.168.2.32
CSeq: 58806 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:0225827915 at aaa.aaa.aa.aaa>
Content-Type: application/sdp
Content-Length: 217
v=0
o=root 26423 26423 IN IP4 aaa.aaa.aa.aaa
s=session
c=IN IP4 aaa.aaa.aa.aaa
t=0 0
m=audio 10548 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
============================================================================
====
SIP MESSAGE 4 192.168.2.32:5060(1) -> sss.sss.ss.sss:5060(2)
UDP Frame 1097 27/Feb/04 18:24:1.2413
TimeFromPreviousSipFrame=9.6471 TimeFromStart=27.2293
Extra Information: Packet is not a complete SIP message
============================================================================
====
SIP MESSAGE 5 sss.sss.ss.sss:5060(2) -> 192.168.2.32:5060(1)
UDP Frame 1407 27/Feb/04 18:24:4.5217
TimeFromPreviousSipFrame=3.2805 TimeFromStart=30.5098
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.2.32:5060;received=80.55.21.254;rport=1184;branch=z9hG4bK3AB182BF553
74A818403FAF5511D2C7B
Record-Route: <sip:0225827915 at sss.sss.ss.sss;ftag=896605854;lr=on>
From: Dawid Mielnik <sip:3267915 at sss.sss.ss.sss>;tag=896605854
To: <sip:0225827915 at sss.sss.ss.sss>;tag=as37250f4f
Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473 at 192.168.2.32
CSeq: 58806 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:0225827915 at aaa.aaa.aa.aaa>
Content-Type: application/sdp
Content-Length: 217
v=0
o=root 26423 26424 IN IP4 aaa.aaa.aa.aaa
s=session
c=IN IP4 aaa.aaa.aa.aaa
t=0 0
m=audio 10548 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
============================================================================
====
SIP MESSAGE 6 192.168.2.32:5060(1) -> sss.sss.ss.sss:5060(2)
UDP Frame 1410 27/Feb/04 18:24:4.5381
TimeFromPreviousSipFrame=0.0164 TimeFromStart=30.5261
ACK sip:0225827915 at aaa.aaa.aa.aaa SIP/2.0
Via: SIP/2.0/UDP
192.168.2.32:5060;rport;branch=z9hG4bK33D0B46E959A49B2914EB72B18029B74
From: Dawid Mielnik <sip:3267915 at sss.sss.ss.sss>;tag=896605854
To: <sip:0225827915 at sss.sss.ss.sss>;tag=as37250f4f
Contact: <sip:3267915 at 192.168.2.32:5060>
Route: <sip:0225827915 at sss.sss.ss.sss;ftag=896605854;lr=on>
Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473 at 192.168.2.32
CSeq: 58806 ACK
Max-Forwards: 70
Content-Length: 0
============================================================================
====
SIP MESSAGE 7 sss.sss.ss.sss:5060(2) -> 192.168.2.32:5060(1)
UDP Frame 1542 27/Feb/04 18:24:5.9842
TimeFromPreviousSipFrame=1.4461 TimeFromStart=31.9723
BYE sip:3267915 at 80.55.21.254:1184 SIP/2.0
Max-Forwards: 10
Record-Route: <sip:3267915 at sss.sss.ss.sss;ftag=as37250f4f;lr=on>
Via: SIP/2.0/UDP sss.sss.ss.sss;branch=z9hG4bKe74a.b59c9824.0
Via: SIP/2.0/UDP aaa.aaa.aa.aaa:5060;branch=z9hG4bK49c1001d
From: <sip:0225827915 at sss.sss.ss.sss>;tag=as37250f4f
To: Dawid Mielnik <sip:3267915 at sss.sss.ss.sss>;tag=896605854
Contact: <sip:0225827915 at aaa.aaa.aa.aaa>
Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473 at 192.168.2.32
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
Route: <sip:3267915 at 80.55.21.254:1184>
============================================================================
====
SIP MESSAGE 8 192.168.2.32:5060(1) -> sss.sss.ss.sss:5060(2)
UDP Frame 1543 27/Feb/04 18:24:5.9935
TimeFromPreviousSipFrame=0.0092 TimeFromStart=31.9815
SIP/2.0 200 Ok
Via: SIP/2.0/UDP sss.sss.ss.sss;branch=z9hG4bKe74a.b59c9824.0
Via: SIP/2.0/UDP aaa.aaa.aa.aaa:5060;branch=z9hG4bK49c1001d
From: <sip:0225827915 at sss.sss.ss.sss>;tag=as37250f4f
To: Dawid Mielnik <sip:3267915 at sss.sss.ss.sss>;tag=896605854
Contact: <sip:3267915 at 192.168.2.32:5060>
Record-Route: <sip:0225827915 at sss.sss.ss.sss;ftag=896605854;lr=on>
Call-ID: AB33A916-A194-4C39-AF7C-7BAF0ED22473 at 192.168.2.32
CSeq: 102 BYE
Server: X-Lite build 1088
Content-Length: 0
============================================================================
====
SIP MESSAGE 9 sss.sss.ss.sss:5060(2) -> 192.168.2.32:5060(1)
UDP Frame 2500 27/Feb/04 18:26:9.9414
TimeFromPreviousSipFrame=123.9479 TimeFromStart=155.9294
Extra Information: Packet was continued from Frame=1643
Extra Information: Packet was continued from Frame=2179
Extra Information: Packet was continued from Frame=2279
Extra Information: Packet was continued from Frame=2369
Extra Information: Packet is not a complete SIP message
Extra Information: Packet does NOT contain a SIP Header but was in the same
connection as Frame=1542
============================================================================
====
2 incomplete sip message(s) encountered
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