[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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