[Serusers] BYE message not recognized

Klaus Darilion klaus.mailinglists at pernau.at
Fri Feb 27 19:40:20 CET 2004


Maybe it's a loose_route problem. If the BYE message is caught in the 
loose_route block, and sent to route[1], then the BYE message does not 
see a setflag(1).

Klaus

Dawid Mielnik wrote:

> 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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> 
> 




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