sr-users December 2003

sr-users@lists.kamailio.org

108 participants
257 discussions

by Kannaiyan Natesan

Thanks Jan. I did as it appears in, http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/etc/nathelper.cfg?r ev=HEAD When i introduce the code as it specified in the nathelper.cfg modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind -- set_mod_param_regex: parameter <ping_nated_only> not found in module <nathelper> parse error (100,43-44): Can't set module parameter if (nat_uac_test("3")) { -- unknown command, missing hodule -rwxr-xr-x 1 827 827 120363 Nov 21 12:10 nathelper.so -- I hope this is the latest module in nathelper.so Am I doing in right way? Kannaiyan ----- Original Message ----- > From: "Jan Janak" <jan(a)iptel.org> > To: "Kannaiyan Natesan" <nkans(a)lycos.co.uk> > Cc: <serusers(a)lists.iptel.org> > Sent: Monday, December 08, 2003 6:28 PM > Subject: Re: [Serusers] Nathelper cfg ? > > > > See sip_router/etc/nathelper.cfg. The config file below is incomplete > > (you do not rewrite contacts, for example). > > > > Jan. > > > > On 08-12 13:23, Kannaiyan Natesan wrote: > > > My script has following information regarding NAT, > > > > > > I have installed RTPProxy and I cannot hear any communication between > the > > > phones. > > > > > > Here is my configuration, > > > > > > # global parameters > > > > > > mhomed=yes > > > > > > # Nathelper > > > loadmodule "/usr/local/lib/ser/modules/nathelper.so" > > > > > > > > > # Parameter section > > > > > > modparam("nathelper","natping_interval", 10) > > > > > > > > > # Routing section > > > route { > > > > > > ..... > > > if( method=="INVITE") { > > > record_route(); > > > force_rtp_proxy(); > > > /* set up reply processing */ > > > t_on_reply("1"); > > > }; > > > .... > > > } > > > > > > > > > #route leg > > > # all incoming replied for t_onreplied transactions enter here > > > onreply_route[1] { > > > if( status =~"[12][0-9][0-9]" ){ > > > force_rtp_proxy(); > > > }; > > > } > > > > > > Can anyone please help me what should I need to do further to call > between > > > my Cisco ATA 186 and X-lite. > > > > > > Best Regards, > > > Kannaiyan > > > > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > >

20 years, 9 months

[Serusers] Problem with SIMPLE2Jabber gateway

by Alan Yoshioka

Hi, I trying to doing a proof of concept using Windows Messenger and SER (SIMPLE2Jabber Gateway). I have SER (0.8.11-0) installed and running on Redhat-9 (Linux 2.4.20-8). I'm using Windows Messenger (Version 4.7) as my client. It seems like the Windows Messenger clients are able to register with the SER. But when I try to send an IM between the clients nothing happens and I see the following error in the log file: Dec 8 16:45:42 yosh-cm1 ser: 0(2234) XJA:xj_address_translation:2249: - wrong Jabber destination yoshioka(a)yosh-cm1.cisco.com!!! Dec 8 16:45:42 yosh-cm1 ser: 0(2234) XJAB:xj_worker:2249: ERROR SENDING as Jabber message ... Any help would be appreciated. Thanks, Alana

20 years, 9 months

[Serusers] rport OK in 0.8.10 ?

by Lasse Jansson

Hi, http://lists.iptel.org/pipermail/serdev/2003-April/000019.html (text copied below) says that the rport functionality in ser-cvs-version at that time was not correct. I have a simple question related to that: - Is the rport functionality correct in ser-0.8.10, or is the same problem present there ? (The "ISSUES" file does not mention any problems related to rport for any ser release, but I want to be sure) /Lasse ---------------------------------------------------------------------------------------------------------- <http://lists.iptel.org/pipermail/serdev/2003-April/000019.html> follows: [Serdev] Re: received-rport in SER ? Jiri Kuthan jiri(a)iptel.org Fri, 04 Apr 2003 18:57:34 +0200 Previous message: [Serdev] Re: Accounting module mods with MySQL and SERWEB Next message: [Serdev] RE: [Serusers] Accounting Script Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] Hopefuly, it is now fixed on CVS. We haven't tested too much yet, though. -Jiri At 12:37 AM 4/2/2003, Maxim Sobolev wrote: >On Tue, Apr 01, 2003 at 06:28:30PM +0200, Sergio Miguel Garcia wrote: >> Hello, >> >> I have been looking for symmetric-reponse ( receiver-rport pair ) in the mailing lists of SER, and I found some interesting mails about the same issue from you >> I've been trying with the CVS version and got these results: ... when I append an empty "rport" in the Via header, the server fills the Via header with the correct values of "rport" and "received", but the packet is sent back to the "received" IP BUT to the port scpecified in the Via header instead of the "rport" port. >> This means that the response of the request never reaches the original UA using a symmetric NAT ( althought the return IP address is correct), because SER sents the UDP packet to a internal port. >> >> I've tried with the Vovida SIP server v1.4 and works fine. Have you tried to fix it or know someone that have tried to solve this issue ? >> >> Thanks in advance for your help and best regards. >> >> Sergio. -- Jiri Kuthan http://iptel.org/~jiri/

20 years, 9 months

[Serusers] problem getting calls to my gateway

by Rick Gocher

Hi everyone, I have been having lots of problems delivering calls to my companies gateway. I have tried several different configs found online however nothing seems to work. I am including the ser.cfg and a dump from ngrep in hopes a kind person will see what the problem is. I do not know very much about sip or ser so any help is greatly needed. Currently our company uses a cisco voip solution and I am setting up Ser as a test. Unfortunately our admin does not seem very helpful, I'm not sure if he has things setup correctly for my calls on the gateway or if it's my ser.cfg file. I was hoping from the information I'm sending someone can tell me where the problem looks like it's coming from, wether it's his gateway or my config file. I'm thinking it's me as I don't see any attempt of passing the call to the gateway in the ngrep output. Also, I have had to alter my ip's listed in this email. I have been warned under penalty of pain not to broadcast their ip addresses... :p I hope this does not cause a problem. ATA 64.189.165.206 Ser Box 64.189.165.205 Cisco GW 65.189.155.101 Thank you, # ----------- 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 #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" # # $Id pstn.cfg,v 1.2 2003/06/03 031812 jiri Exp $ # # # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/acc.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/group.so" modparam("auth_db", "db_url","sql//secret@localhost/ser") modparam("usrloc", "db_url", "sql//secret@localhost/ser") # ----------------- setting module-specific parameters --------------- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") # -- acc params -- modparam("acc", "log_level", 1) # that is the flag for which we will account -- don't forget to # set the same one -) modparam("acc", "log_flag", 1 ) # ------------------------- request routing logic ------------------- # main routing logic route{ /* ********* ROUTINE CHECKS ********************************** */ # filter too old messages if (!mf_process_maxfwd_header("10")) { log("LOG Too many hops\n"); sl_send_reply("483","Too Many Hops"); break; }; if (msglen >= max_len ) { sl_send_reply("513", "Message too big"); break; }; /* ********* RR ********************************** */ /* grant Route routing if route headers present */ if (loose_route()) { t_relay(); break; }; /* record-route INVITEs -- all subsequent requests must visit us */ if (method=="INVITE") { record_route(); }; # now check if it really is a PSTN destination which should be handled # by our gateway; if not, and the request is an invitation, drop it -- # we cannot terminate it in PSTN; relay non-INVITE requests -- it may # be for example BYEs sent by gateway to call originator if (!uri=~"sip\+?[0-9]+@.*") { if (method=="INVITE") { sl_send_reply("403", "Call cannot be served here"); } else { forward(urihost, uriport); }; break; }; # account completed transactions via syslog setflag(1); # free call destinations ... no authentication needed if ( is_user_in("Request-URI", "free-pstn") /* free destinations */ | uri=~"sip[7][0-9][0-9][0-9]@.*" /* local PBX */ | uri=~"sip98[0-9][0-9][0-9][0-9]") { log("free call"); } else if (src_ip==65.189.155.101) { # our gateway doesn't support digest authentication; # verify that a request is coming from it by source # address log("gateway-originated request"); } else { # in all other cases, we need to check the request against # access control lists; first of all, verify request # originator's identity if (!proxy_authorize( "gateway" /* realm */, "subscriber" /* table name */)) { proxy_challenge( "gateway" /* realm */, "0" /* no qop */ ); break; }; # authorize only for INVITEs -- RR/Contact may result in weird # things showing up in d-uri that would break our logic; our # major concern is INVITE which causes PSTN costs if (method=="INVITE") { # does the authenticated user have a permission for local # calls (destinations beginning with a single zero)? # (i.e., is he in the "local" group?) if (uri=~"sip0[1-9][0-9]+@.*") { if (!is_user_in("credentials", "local")) { sl_send_reply("403", "No permission for local calls"); break; }; # the same for long-distance (destinations begin with two zeros") } else if (uri=~"sip00[1-9][0-9]+@.*") { if (!is_user_in("credentials", "ld")) { sl_send_reply("403", " no permission for LD "); break; }; # the same for international calls (three zeros) } else if (uri=~"sip000[1-9][0-9]+@.*") { if (!is_user_in("credentials", "int")) { sl_send_reply("403", "International permissions needed"); break; }; # everything else (e.g., interplanetary calls) is denied } else { sl_send_reply("403", "Forbidden"); break; }; }; # INVITE to authorized PSTN }; # authorized PSTN # if you have passed through all the checks, let your call go to GW! rewritehostport("65.189.155.1015060"); # forward the request now if (!t_relay()) { sl_reply_error(); break; }; } ################ ngrep output####################### # U 64.189.165.2065060 -> 64.189.165.2055060 INVITE sip776044445556(a)64.189.165.205;user=phone SIP/2.0..Via SIP/2.0/UDP 64.189.165.2065060..From <sip6 044848235(a)64.189.165.205;user=phone>;tag=409936633..To <sip776044445556(a)64.189.165.205;user=phone>..Call-ID 2945885252@64.189.165.206..CSeq 1 INVITE..Contact <sip6044445555(a)64.189.165.2065060;user=phone;transpor t=udp>..User-Agent Cisco ATA 186 v2.16.2 ata18x (030909a)..Expires 300..Content-Length 257..Content-Typ e application/sdp....v=0..o=6044445555 62848 62848 IN IP4 64.189.165.206..s=ATA186 Call..c=IN IP4 64.189.165.206..t=0 0..m=audio 16384 RTP/AVP 18 8 0 101..a=rtpmap18 G729/8000/1..a=rtpmap8 PCMA/8000/1..a=rtpmap0PCMU/8000/1..a=rtpmap101 telephone-event/8000..a=fmtp101 0-15.. # U 64.189.165.2055060 -> 64.189.165.2065060 SIP/2.0 407 Proxy Authentication Required..Via SIP/2.0/UDP 64.189.165.2065060..From <sip6044445555(a)64.189.165.205;user=phone>;tag=409936633..To <sip776044445556(a)64.189.165.205;user=phone>;tag=b27e1a1d33761e85846fc98f5f3a7e58.0ed0..Call-ID 2945885252@64.189.165.206..CSeq 1 INVITE..Proxy-Authenticate Digest realm="gateway", nonce="3fcf790810cb0daaf030be719aa79e574b96b535"..Server Sip EXpress router (0.8.12 (i386/linux)). .Content-Length 0..Warning 392 64.189.165.2055060 "Noisy feedback tells pid=32407 req_src_ip=64.189.165.206 req_src_port=5060 in_uri=sip776044445556(a)64.189.165.205;user=phone out_uri=sip776044445556(a)64.189.165.205;user=phone via_cnt==1".... # U 64.189.165.2065060 -> 64.189.165.2055060 ACK sip776044445556(a)64.189.165.205;user=phone SIP/2.0..Via SIP/2.0/UDP 64.189.165.2065060..From <sip6044 445555(a)64.189.165.205;user=phone>;tag=409936633..To <sip776044445556(a)64.189.165.205;user=phone>;tag=b27e1a1 d33761e85846fc98f5f3a7e58.0ed0..Call-ID 2945885252@64.189.165.206..CSeq 1 ACK..User-Agent Cisco ATA 186 v2.16.2 ata18x (030909a)..Content-Length 0....

20 years, 9 months

[Serusers] help for PA can not work with MS messager 5.0

by hong cheng

hi, In the PA document, it said "This module currently cannot work as a B2BUA, i.e. it cannot subsribe to presence of users using SIP. We are currently working on that feature." So for 0.8.12, dose SER provide Presence Agent feature for SIP UA (like MS Messager) ? Thanks, hong

20 years, 9 months

[Serusers] Nathelper cfg ?

by Kannaiyan Natesan

My script has following information regarding NAT, I have installed RTPProxy and I cannot hear any communication between the phones. Here is my configuration, # global parameters mhomed=yes # Nathelper loadmodule "/usr/local/lib/ser/modules/nathelper.so" # Parameter section modparam("nathelper","natping_interval", 10) # Routing section route { ..... if( method=="INVITE") { record_route(); force_rtp_proxy(); /* set up reply processing */ t_on_reply("1"); }; .... } #route leg # all incoming replied for t_onreplied transactions enter here onreply_route[1] { if( status =~"[12][0-9][0-9]" ){ force_rtp_proxy(); }; } Can anyone please help me what should I need to do further to call between my Cisco ATA 186 and X-lite. Best Regards, Kannaiyan

20 years, 9 months

[Serusers] A user with sip contact and pstn contact

by Chen Youcheng

hello, i try to give two contacts for a callee. One is sip contact and one is pstn contact. When the callee is off line,the call should be lead to pstn contact. I have done: serctl ul add yongcheng sip:017923435@gateway.de. sip contact is yongcheng(a)mydomain.de. I cannot finish the mapping from yongcheng(a)mydomain.de to 017923435(a)gateway.de. how can I do? best regards Yongcheng Chen __________________________________ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/

20 years, 9 months

[Serusers] simply moving rewritehostport location we get 481 on disconnect

by Glenn Dalgliesh

Below are 2 example sections of ser.cfg and the only difference it moving the rewritehostport from route[0] default to route[1]. This seems to cause issues with the BYE messages. When I have it in route[1] the call doesn't hang-up correctly and I see the 481 when looking at ngrep. From what I can make out from the docs both should be valid methods of sending call to the PSTN. Please help me understand the fault in my logic. The reason I am trying to track this issue down it that in my more complex configs with all of the possibilities it would be nice to be able to include certain function in route sub routines. Thanks See Capture1 below for ngrep port 5060 capture of config in example 1 See Capture2 below for ngrep port 5060 capture of config in example 2 Example 1: if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (uri=~"^sip:[0-9]{10}@") { rewritehostport("209.163.109.7:7021"); route(1); break; }; if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } Example 2: if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (uri=~"^sip:[0-9]{10}@") { route(1); break; }; if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP rewritehostport("209.163.109.7:7021"); if (!t_relay()) { sl_reply_error(); }; } Capture 1: # U 162.33.165.203:5060 -> 162.33.165.197:5060 INVITE sip:4102959745@sipdemo.routerboy.com SIP/2.0.. Via: SIP/2.0/UDP 162.33.165.203.. From: "ToadNet Testbed 5" <sip:7003@sipdemo.routerboy.com>;tag=b45ee895-d03c-de8e-caa4-a149eb26e6e8.. To: <sip:4102959745@sipdemo.routerboy.com>.. Contact: <sip:7003@162.33.165.203>.. Call-ID: 10e079f5-5d15-7807-f095-7021e89597c2(a)162.33.165.203.. CSeq: 15616 INVITE.. User-Agent: Grandstream SIP UA 1.0.3.81.. Max-Forwards: 70.. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE.. Content-Type: application/sdp.. Content-Length: 234.. .. v=0.. o=7003 0 0 IN IP4 162.33.165.203.. s=-.. c=IN IP4 162.33.165.203.. t=0 0.. m=audio 5004 RTP/AVP 0 8 4 18 15.. a=ptime:20.. a=rtpmap:0 PCMU/8000.. a=rtpmap:8 PCMA/8000.. a=rtpmap:4 G723/8000.. a=rtpmap:18 G729/8000.. a=rtpmap:15 G728/8000.. # U 162.33.165.197:5060 -> 162.33.165.203:5060 SIP/2.0 100 trying -- your call is important to us.. Via: SIP/2.0/UDP 162.33.165.203.. From: "ToadNet Testbed 5" <sip:7003@sipdemo.routerboy.com>;tag=b45ee895-d03c-de8e-caa4-a149eb26e6e8.. To: <sip:4102959745@sipdemo.routerboy.com>.. Call-ID: 10e079f5-5d15-7807-f095-7021e89597c2(a)162.33.165.203.. CSeq: 15616 INVITE.. Server: Sip EXpress router (0.8.12-dev-23-merged (i386/linux)).. Content-Length: 0.. Warning: 392 162.33.165.197:5060 "Noisy feedback tells: pid=8102 req_src_ip=162.33.165.203 req_src_port=5060 in_ uri=sip:4102959745@sipdemo.routerboy.com out_uri=sip:4102959745@209.163.109.7:7021 via_cnt==1".. .. # U 162.33.165.197:5060 -> 209.163.109.7:7021 INVITE sip:4102959745@209.163.109.7:7021 SIP/2.0.. Record-Route: <sip:4102959745@162.33.165.197;ftag=b45ee895-d03c-de8e-caa4-a149eb26e6e8;lr= on>.. Via: SIP/2.0/UDP 162.33.165.197;branch=z9hG4bK36e6.3c2402d6.0.. Via: SIP/2.0/UDP 162.33.165.203.. From: "ToadNet Testbed 5" <sip:7003@sipdemo.routerboy.com>;tag=b45ee895-d03c-de8e-caa4-a149eb26e6e8.. To: <sip:4102959745@sipdemo.routerboy.com>.. Contact: <sip:7003@162.33.165.203>.. Call-ID: 10e079f5-5d15-7807-f095-7021e89597c2(a)162.33.165.203.. CSeq: 15616 INVITE.. User-Agent: Grandstream SIP UA 1.0.3.81.. Max-Forwards: 69.. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE .. Content-Type: application/sdp.. Content-Length: 234.. .. v=0.. o=7003 0 0 IN IP4 162.33.165.203.. s=-.. c=IN IP4 162.33.165.203.. t=0 0.. m=audio 5004 RTP/AVP 0 8 4 18 15.. a=ptime:20.. a=rtpmap:0 PCMU/8000.. a=rtpmap:8 PCMA/8000.. a=rtpmap:4 G723/8000.. a=rtpmap:18 G729/8000.. a=rtpmap:15 G728/8000.. # U 209.163.109.7:7021 -> 162.33.165.197:5060 SIP/2.0 100 Trying.. Via: SIP/2.0/UDP 162.33.165.197:5060;branch=z9hG4bK36e6.3c2402d6.0.. Via: SIP/2.0/UDP 162.33.165.203:5060.. To: <sip:4102959745@sipdemo.routerboy.com>.. From: "ToadNet Testbed 5"<sip:7003@sipdemo.routerboy.com>;tag=b45ee895-d03c-de8e-caa4-a149eb26e6e8. . Call-ID: 10e079f5-5d15-7807-f095-7021e89597c2(a)162.33.165.203.. CSeq: 15616 INVITE.. Content-Length: 0.. .. # U 209.163.109.7:7021 -> 162.33.165.197:5060 SIP/2.0 183 Session Progress.. Via: SIP/2.0/UDP 162.33.165.197:5060;branch=z9hG4bK36e6.3c2402d6.0.. Via: SIP/2.0/UDP 162.33.165.203:5060.. To: <sip:4102959745@sipdemo.routerboy.com>;tag=DDEA1874-1898.. From: "ToadNet Testbed 5"<sip:7003@sipdemo.routerboy.com>;tag=b45ee895-d03c-de8e-caa4-a149eb26e6e8. . Date: Fri, 05 Dec 2003 18:01:23 GMT.. Call-ID: 10e079f5-5d15-7807-f095-7021e89597c2(a)162.33.165.203.. CSeq: 15616 INVITE.. Server: Cisco-SIPGateway/IOS-12.x.. Content-Disposition: session;handling=required.. Content-Type: application/sdp.. Content-Length: 219.. Allow-Events: telephone-event.. .. v=0.. o=CiscoSystemsSIP-GW-UserAgent 5092 3331 IN I P4 209.163.109.9.. s=SIP Call.. c=IN IP4 209.163.109.9.. t=0 0.. m=audio 17708 RTP/AVP 0 100.. a=rtpmap:0 PCMU/8000.. a=rtpmap:100 X-NSE/8000.. a=fmtp:100 192-194.. a=ptime:20.. # U 162.33.165.197:5060 -> 162.33.165.203:5060 SIP/2.0 183 Session Progress.. Via: SIP/2.0/UDP 162.33.165.203:5060.. To: <sip:4102959745@sipdemo.routerboy.com>;tag=DDEA1874-1898.. From: "ToadNet Testbed 5"<sip:7003@sipdemo.routerboy.com>;tag=b45ee895-d03c-de8e-caa4-a149eb26e6e8. . Date: Fri, 05 Dec 2003 18:01:23 GMT.. Call-ID: 10e079f5-5d15-7807-f095-7021e89597c2(a)162.33.165.203.. CSeq: 15616 INVITE.. Server: Cisco-SIPGateway/IOS-12.x.. Content-Disposition: session;handling=required.. Content-Type: application/sdp.. Content-Length: 219.. Allow-Events: telephone-event.. .. v=0.. o=CiscoSystemsSIP-GW-UserAgent 5092 3331 IN IP4 209.163.109.9.. s=SIP Call.. c=IN IP4 209.163.109.9.. t=0 0.. m=audio 17708 RTP/AVP 0 100.. a=rtpmap:0 PCMU/8000.. a=rtpmap:100 X-NSE/8000.. a=fmtp:100 192-194.. a=ptime:20.. # U 209.163.109.7:7021 -> 162.33.165.197:5060 SIP/2.0 200 OK.. Via: SIP/2.0/UDP 162.33.165.197:5060;branch=z9hG4bK36e6.3c2402d6.0.. Via: SIP/2.0/UDP 162.33.165.203:5060.. To: <sip:4102959745@sipdemo.routerboy.com>;tag=DDEA1874-1898.. From: "ToadNet Testbed 5"<sip:7003@sipdemo.routerboy.com>;tag=b45ee895-d03c-de8e-caa4-a149eb26e6e8. . Date: Fri, 05 Dec 2003 18:01:23 GMT.. Call-ID: 10e079f5-5d15-7807-f095-7021e89597c2(a)162.33.165.203.. CSeq: 15616 INVITE.. Server: Cisco-SIPGateway/IOS-12.x.. Record-Route: <sip:238014102959745@209.163.109.7:5067;maddr=209.163.109.7>,<sip:4102959745 @209.163.109.7:7021;maddr=209.163.109.7>,<sip:4102959745@162.33.165.197;ftag =b45ee895-d03c-de8e-caa4-a149eb26e6e8;lr=on>.. Contact: <sip:238014102959745@209.163.109.9:5060;user=phone>.. Content-Type: application/sdp.. Content-Length: 219.. Allow-Events: telephone-event.. .. v=0.. o=CiscoSystemsSIP-GW-UserAgent 5092 3331 IN IP4 209.163.109.9.. s=SIP Call.. c=IN IP4 209.163.109.9.. t=0 0.. m=audio 17708 RTP/AVP 0 100.. a=rtpmap:0 PCMU/8000.. a=rtpmap:100 X-NSE/8000.. a=fmtp:100 192-194.. a=ptime:20.. # U 162.33.165.197:5060 -> 162.33.165.203:5060 SIP/2.0 200 OK.. Via: SIP/2.0/UDP 162.33.165.203:5060.. To: <sip:4102959745@sipdemo.routerboy.com>;tag=DDEA1874-1898.. From: "ToadNet Testbed 5"<sip:7003@sipdemo.routerboy.com>;tag=b45ee895-d03c-de8e-caa4-a149eb26e6e8. . Date: Fri, 05 Dec 2003 18:01:23 GMT.. C all-ID: 10e079f5-5d15-7807-f095-7021e89597c2(a)162.33.165.203.. CSeq: 15616 INVITE.. Server: Cisco-SIPGateway/IOS-12.x.. Record-Route: <sip:238014102959745@209.163.109.7:5067;maddr=209.163.109.7>,<sip:4102959745 @209.163.109.7:7021;maddr=209.163.109.7>,<sip:4102959745@162.33.165.197;ftag =b45ee895-d03c-de8e-caa4-a149eb26e6e8;lr=on>.. Contact: <sip:238014102959745@209.163.109.9:5060;user=phone>.. Content-Type: application/sdp.. Content-Length: 219.. Allow-Events: telephone-event.. .. v=0.. o=CiscoSystemsSIP-GW-UserAgent 5092 3331 IN IP4 209.163.109.9.. s=SIP Call.. c=IN IP4 209.163.109.9.. t=0 0.. m=audio 17708 RTP/AVP 0 100.. a=rtpmap:0 PCMU/8000 .. a=rtpmap:100 X-NSE/8000.. a=fmtp:100 192-194.. a=ptime:20.. # U 162.33.165.203:5060 -> 162.33.165.197:5060 ACK sip:238014102959745@209.163.109.9:5060 SIP/2.0.. Via: SIP/2.0/UDP 162.33.165.203.. Route: <sip:4102959745@162.33.165.197;ftag=b45ee895-d03c-de8e-caa4-a149eb26e6e8;lr= on>.. Route: <sip:4102959745@209.163.109.7:7021;maddr=209.163.109.7>.. Route: <sip:238014102959745@209.163.109.7:5067;maddr=209.163.109.7>.. From: "ToadNet Testbed 5" <sip:7003@sipdemo.routerboy.com>;tag=b45ee895-d03c-de8e-caa4-a149eb26e6e8.. To: <sip:4102959745@sipdemo.routerboy.com>;tag=DDEA1874-1898.. Contact: <sip:7003@162.33.165.203>.. Call-ID: 10e079f5-5d15-7807-f095-7021e89597c2(a)162.33.165.203.. CSeq: 15616 ACK.. User-Agent: Grandstream SIP UA 1.0.3.81.. Max-Forwards: 70.. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE.. Content-Length: 0.. .. # U 162.33.165.197:5060 -> 209.163.109.7:7021 ACK sip:4102959745@209.163.109.7:7021;maddr=209.163.109.7 SIP/2.0.. Record-Route: <sip:238014102959745@162.33.165.197;ftag=b45ee895-d03c-de8e-caa4-a149eb26e6e 8;lr=on>.. Via: SIP/2.0/UDP 162.33.165.197;branch=0.. Via: SIP/2.0/UDP 162.33.165.203.. Route: <sip:238014102959745@209.163.109.7:5067;maddr=209.163.109.7>.. From: "ToadNet Testbed 5" <sip:7003@sipdemo.routerboy.com>;tag=b45ee895-d03c-de8e-caa4-a149eb26e6e8.. To: <sip:4102959745@sipdemo.routerboy.com>;tag=DDEA1874-1898.. Contact: <sip:7003@162.33.165.203> .. Call-ID: 10e079f5-5d15-7807-f095-7021e89597c2(a)162.33.165.203.. CSeq: 15616 ACK.. User-Agent: Grandstream SIP UA 1.0.3.81.. Max-Forwards: 69.. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE.. Content-Length: 0.. Route: <sip:238014102959745@209.163.109.9:5060>.. P-hint: rr-enforced.. .. # U 209.163.109.7:7021 -> 162.33.165.197:5060 BYE sip:4102959745@162.33.165.197:5060 SIP/2.0.. Via: SIP/2.0/UDP 209.163.109.7:7021;branch=z9hG4bK774aea783c1494ea3869b17500fb9d20.4.. Via: SIP/2.0/UDP 209.163.109.7:5067;branch=z9hG4bK860578b098c2e73149eab3e067750477.2.. Via: SIP/2.0/UDP 209.163.109.9:5060.. To: "ToadNet Testbed 5"<sip:7003@sipdemo.routerboy.com>;tag=b45ee895-d03c-de8e-caa4-a149eb26e6e8. . From: <sip:4102959745@sipdemo.routerboy.com>;tag=DDEA1874-1898.. Date: Fri, 05 Dec 2003 18:01:31 GMT.. Call-ID: 10e079f5-5d15-7807-f095-7021e89597c2(a)162.33.165.203.. CSeq: 101 BYE.. Max-Forwards: 5.. Route: <sip:7003@162.33.165.203:5060>.. Timestamp: 1070647369.. User-Agent: Cisco-SIPGateway/IOS-12.x.. Content-Length: 0.. .. # U 162.33.165.197:5060 -> 162.33.165.203:5060 BYE sip:7003@162.33.165.203:5060 SIP/2.0.. Record-Route: <sip:4102959745@162.33.165.197;ftag=DDEA1874-1898;lr=on>.. Via: SIP/2.0/ UDP 162.33.165.197;branch=z9hG4bK2906.7cd4e111.0.. Via: SIP/2.0/UDP 209.163.109.7:7021;branch=z9hG4bK774aea783c1494ea3869b17500f b9d20.4.. Via: SIP/2.0/UDP 209.163.109.7:5067;branch=z9hG4bK860578b098c2e73149eab3e067750477.2.. Via: SIP/2.0/UDP 209.163.109.9:5060.. To: "ToadNet Testbed 5"<sip:7003@sipdemo.routerboy.com>;tag=b45ee895-d03c-de8e-caa4-a149eb26e6e8. . From: <sip:4102959745@sipdemo.routerboy.com>;tag=DDEA1874-1898.. Date: Fri, 05 Dec 2003 18:01:31 GMT.. Call-ID: 10e079f5-5d15-7807-f095-7021e89597c2(a)162.33.165.203.. CSeq: 101 BYE.. Max-Forwards: 4.. Timestamp: 1070647369.. User-Agent: Cisco-SIPGateway/IOS-12.x.. Content-Length: 0.. P-hint: rr-enforced.. .. # U 162.33.165.203:5060 -> 162.33.165.197:5060 SIP/2.0 200 OK.. Via: SIP/2.0/UDP 162.33.165.197;branch=z9hG4bK2906.7cd4e111.0.. Via: SIP/2.0/UDP 209.163.109.7:7021;branch=z9hG4bK774aea783c1494ea3869b17500fb9d20.4.. Via: SIP/2.0/UDP 209.163.109.7:5067;branch=z9hG4bK860578b098c2e73149eab3e067750477.2.. Via: SIP/2.0/UDP 209.163.109.9:5060.. Record-Route: <sip:4102959745@162.33.165.197;ftag=DDEA1874-1898;lr=on>.. From: <sip:4102959745@sipdemo.routerboy.com>;tag=DDEA1874-1898.. To: "ToadNet Testbed 5"<sip:7003@sipdemo.routerboy.com>;tag=b45ee895-d03c-de8e-caa4-a149eb26e6e8. . Call-ID: 10e079f5-5d15-7807-f095-7021e89597c2(a)162.33.165.203.. CSeq: 101 BYE.. User-Agent: Grandstream SIP UA 1.0.3.81.. Contact: <sip:7003@162.33.165.203>.. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE.. Content-Length: 0.. .. # U 162.33.165.197:5060 -> 209.163.109.7:7021 SIP/2.0 200 OK.. Via: SIP/2.0/UDP 209.163.109.7:7021;branch=z9hG4bK774aea783c1494ea3869b17500fb9d20.4.. Via: SIP/2.0/UDP 209.163.109.7:5067;branch=z9hG4bK860578b098c2e73149eab3e067750477.2.. Via: SIP/2.0/UDP 209.163.109.9:5060.. Record-Route: <sip:4102959745@162.33.165.197;ftag=DDEA1874-1898;lr=on>.. From: <sip:4102959745@sipdemo.routerboy.com>;tag=DDEA1874-1898.. To: "ToadNet Testbed 5"<sip:7003@sipdemo.routerboy.com>;tag=b45ee895-d03c-de8e-caa4-a149eb26e6e8. . Call-ID: 10e079f5-5d15-7807-f095-7021e89597c2(a)162.33.165.203.. CSeq: 101 BYE.. User-Agent: Grandstream SIP UA 1.0.3.81.. Contact: <sip:7003@162.33.165.203>.. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE.. Content-Length: 0.. .. # U 162.33.165.196:29479 -> 162.33.165.197:5060 .. Capture 2: [root@serv1 ser]# ngrep port 5060 interface: eth0 (162.33.165.192/255.255.255.240) filter: ip and ( port 5060 ) # U 162.33.165.203:5060 -> 162.33.165.197:5060 INVITE sip:4102959745@sipdemo.routerboy.com SIP/2.0.. Via: SIP/2.0/UDP 162.33.165.203.. From: "ToadNet Testbed 5" <sip:7003@sipdemo.routerboy.com>;tag=2b00806c-495b-b6c0-cceb-682930719843.. To: <sip:4102959745@sipdemo.routerboy.com>.. Contact: <sip:7003@162.33.165.203>.. Call-ID: 5aa02b3f-1d95-0f2f-dd55-aca6806cbcd7(a)162.33.165.203.. CSeq: 63249 INVITE.. User-Agent: Grandstream SIP UA 1.0.3.81.. Max-Forwards: 70.. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE.. Content-Type: application/sdp.. Content-Length: 234.. .. v=0.. o=7003 0 0 IN IP4 162.33.165.203.. s=-.. c=IN IP4 162.33.165.203.. t=0 0.. m=audio 5004 RTP/AVP 08 4 18 15.. a=ptime:20.. a=rtpmap:0 PCMU/8000.. a=rtpmap:8 PCMA/8000.. a=rtpmap:4 G723/8000.. a=rtpmap:18 G729/8000.. a=rtpmap:15 G72 8/8000.. # U 162.33.165.197:5060 -> 162.33.165.203:5060 SIP/2.0 100 trying -- your call is important to us.. Via: SIP/2.0/UDP 162.33.165.203.. From: "ToadNet Testbed 5" <sip:7003@sipdemo.routerboy.com>;tag=2b00806c-495b-b6c0-cceb-682930719843.. To: <sip:4102959745@sipdemo.routerboy.com>.. Call-ID: 5aa02b3f-1d95-0 f2f-dd55-aca6806cbcd7(a)162.33.165.203.. CSeq: 63249 INVITE.. Server: Sip EXpress router (0.8.12-dev-23-merged (i386/linux)).. Content-Length: 0.. Warning: 392 162.33.165.197:5060 "Noisy feedback tells: pid=8060 req_src_ip=162.33.165.203 req_src_port=5060 in_ uri=sip:4102959745@sipdemo.routerboy.com out_uri=sip:4102959745@209.163.109.7:7021 via_cnt==1".. .. # U 162.33.165.197:5060 -> 209.163.109.7:7021 INVITE sip:4102959745@209.163.109.7:7021 SIP/2.0.. Record-Route: <sip:4102959745@162.33.165.197;ftag=2b00806c-495b-b6c0-cceb-682930719843;lr= on>.. Via: SIP/2.0/UDP 162.33.165.197;branch=z9hG4bK4cc4.32dfa471.0.. Via: SIP/2.0/UDP 162.33.165.203.. From: "ToadNet Testbed 5" <sip:7003@sipdemo.routerboy.com>;tag=2b00806c-495b-b6c0-cceb-682930719843.. To: <sip:4102959745@sipdemo.routerboy.com>.. Contact: <sip:7003@162.33.165.203>.. Call-ID: 5aa02b3f-1d95-0f2f-dd55-aca6806cbcd7(a)162.33.165.203.. CSeq: 63249 INVITE.. User-Agent: Grandstream SIP UA 1.0.3.81.. Max-Forwards: 69.. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE.. Content-Type: application/sdp.. Content-Length: 234.. .. v=0.. o=7003 0 0 IN IP4 162.33.165.203.. s=-.. c=IN IP4 162.33.165.203.. t=0 0.. m=audio 5004 RTP/AVP 0 8 4 18 15.. a=ptime:20.. a=rtpmap:0 PCMU/8000.. a=rtpmap:8 PCMA/8000.. a=rtpmap:4 G723/8000.. a=rtpmap:1 8 G729/8000.. a=rtpmap:15 G728/8000.. # U 209.163.109.7:7021 -> 162.33.165.197:5060 SIP/2.0 100 Trying.. Via: SIP/2.0/UDP 162.33.165.197:5060;branch=z9hG4bK4cc4.32dfa471.0.. Via: SIP/2.0/UDP 162.33.165.203:5060.. To: <sip:4102959745@sipdemo.routerboy.com>.. From: "ToadNet Testbed 5"<sip:7003@sipdemo.routerboy.com>;tag=2b00806c-495b-b6c0-cceb-682930719843. . Call-ID: 5aa02b3f-1d95-0f2f-dd55-aca6806cbcd7(a)162.33.165.203.. CSeq: 63249 INVITE.. Content-Length: 0.. .. # U 209.163.109.7:7021 -> 162.33.165.197:5060 SIP/2.0 183 Session Progress.. Via: SIP/2.0/UDP 162.33.165.197:5060;branch=z9hG4bK4cc4.32dfa471.0.. Via: SIP/2.0/UDP 162.33.165.203:5060.. To: <sip:4102959745@sipdemo.routerboy.com>;tag=DDDC4048-2260.. From: "ToadNet Testbed 5"<sip:7003@sipdemo.routerboy.com>;tag=2b00806c-495b-b6c0-cceb-682930719843. . Date: Fri, 05 Dec 2003 17:46:15 GMT.. Call-ID: 5aa02b3f-1d95-0f2f-dd55-aca6806cbcd7(a)162.33.165.203.. CSeq: 63249 INVITE.. Server: Cisco-SIPGateway/IOS-12.x.. Content-Disposition: session;handling=required.. Content-Type: application/sdp.. Content-Length: 219.. Allow-Events: telephone-event.. .. v=0.. o=CiscoSystemsSIP-GW-UserAgent 1002 2938 IN IP4 209.163.109.9.. s=SIP Call.. c=IN IP4 209.163.109.9.. t=0 0.. m=audio 17976 RTP/AVP 0 100.. a=rtpmap:0 PCMU/8000.. a=rtpmap:100 X-NSE/8000.. a=fmtp:100 192-194.. a=ptime:20.. # U 162.33.165.197:5060 -> 162.33.165.203:5060 SIP/2.0 183 Session Progress.. Via: SIP/2.0/UDP 162.33.165.203:5060.. To: <sip:4102959745@sipdemo.routerboy.com>;tag=DDDC4048-2260.. From: "ToadNet Testbed 5"<sip:7003@sipdemo.routerboy.com>;tag=2b00806c-495b-b6c0-cceb-682930719843. . Date: Fri, 05 Dec 2003 17:46:15 GMT.. Call-ID: 5aa02b3f-1d95-0f2f-dd55-aca6806cbcd7(a)162.33.165.203.. CSeq: 63249 INVITE.. Server: Cisco-SIPGateway/IOS-12.x.. Content-Disposition: session;handling=required.. Content-Type: application/sdp.. Content-Length: 219.. Allow-Events: telephone-event.. .. v=0.. o=CiscoSystemsSIP-GW-UserAgent 1002 2938 IN IP4 209.163.109.9.. s=SIP Call.. c=IN IP4 209.163.109.9.. t=0 0.. m=audio 17976 RTP/AVP 0 100.. a=rtpmap:0 PCMU/8000.. a=rtpmap:100 X-NSE/8000.. a=fmtp:100 192-194.. a=ptime:20.. # U 209.163.109.7:7021 -> 162.33.165.197:5060 SIP/2.0 200 OK.. Via: SIP/2.0/UDP 162.33.165.197:5060;branch=z9hG4bK4cc4.32dfa471.0.. Via: SIP/2.0/UDP 162.33.165.203:5060.. To: <sip:4102959745@sipdemo.routerboy.com>;tag=DDDC4048-2260.. From: "ToadNet Testbed 5"<sip:7003@sipdemo.routerboy.com>;tag=2b00806c-495b-b6c0-cceb-682930719843. . Date: Fri, 05 Dec 2003 17:46:15 GMT.. Call-ID: 5aa02b3f-1d95-0f2f-dd55-aca6806cbcd7(a)162.33.165.203.. CSeq: 63249 INVITE.. Server: Cisco-SIPGateway/IOS-12.x.. Record-Route: <sip:238014102959745@209.163.109.7:5067;maddr=209.163.10 9.7>,<sip:4102959745@209.163.109.7:7021;maddr=209.163.109.7>,<sip:4102959745 @162.33.165.197;ftag=2b00806c-495b-b6c0-cceb-682930 719843;lr=on>.. Contact: <sip:238014102959745@209.163.109.9:5060;user=phone>.. Content-Type: application/sdp.. Content-Length: 219.. Allow-Events: telephone-event.. .. v=0.. o=CiscoSystemsSIP-GW-UserAgent 1002 2938 IN IP4 209.163.109.9.. s=SIP Call.. c=IN IP4 209.163.109.9.. t=0 0.. m=audio 17976 RTP/AVP 0 100.. a=rtpmap:0 PCMU/8000.. a=rtpmap:100 X-NSE/8000.. a=fmtp:100 192-194.. a=ptime:20.. # U 162.33.165.197:5060 -> 162.33.165.203:5060 SIP/2.0 200 OK.. Via: SIP/2.0/UDP 162.33.165.203:5060.. To: <sip:4102959745@sipdemo.routerboy.com>;tag=DDDC4048-2260.. From: "ToadNet Testbed 5"<sip:7003@sipdemo.routerboy.com>;tag=2b00806c-495b-b6c0-cceb-682930719843. . Date: Fri, 05 Dec 2003 17:46:15 GMT.. Call-ID: 5aa02b3f-1d95-0f2f-dd55-aca6806cbcd7(a)162.33.165.203.. CSeq: 63249 INVITE.. Server: Cisco-SIPGateway/IOS-12.x.. Record-Route: <sip:238014102959745@209.163.109.7:5067;maddr=209.163.109.7>,<sip:4102959745 @209.163.109.7:7021;maddr=209.163.109.7>,<sip:41 02959745(a)162.33.165.197;ftag=2b00806c-495b-b6c0-cceb-682930719843;lr=on>.. Contact: <sip:238014102959745@209.163.109.9:5060;user=phone>.. Content-Type: application/sdp.. Content-Length: 219.. Allow-Events: telephone-event.. .. v=0.. o=CiscoSystemsSIP-GW-UserAgent 1002 2938 IN IP4 209.163.109.9.. s=SIP Call.. c=IN IP4 209.163.109.9.. t=0 0.. m=audio 17976 RTP/AVP 0 100.. a=rtpmap:0 PCMU/8000.. a=rtpmap:100 X-NSE/8000.. a=fmtp:100 192-194.. a=ptime:20.. # U 162.33.165.203:5060 -> 162.33.165.197:5060 ACK sip:238014102959745@209.163.109.9:5060 SIP/2.0.. Via: SIP/2.0/UDP 162.33.165.203.. Route: <sip:4102959745@162.33.165.197;ftag=2b00806c-495b-b6c0-cceb-682930719843;lr= on>.. Route: <sip:4102959745@209.163.109.7:7021;maddr=209.163.109.7>.. Route: <sip:238014102959745@209.163.109.7:5067;maddr=209.163.109.7>.. From: "ToadNet Testbed 5" <sip:7003@sipdemo.routerboy.com>;tag=2b00806c-495b-b6c0-cceb-682930719843.. To: <sip:4102959745@sipdemo.routerboy.com>;tag=DDDC4048-2260.. Contact: <sip:7003@162.33.165.203>.. Call-ID: 5aa02b3f-1d95-0f2f-dd55-aca6806cbcd7(a)162.33.165.203.. CSeq: 63249 ACK.. User-Agent: Grandstream SIP UA 1.0.3.81.. Max-Forwards: 70.. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE.. Content-Length: 0.. .. # U 162.33.165.197:5060 -> 209.163.109.7:7021 ACK sip:4102959745@209.163.109.7:7021;maddr=209.163.109.7 SIP/2.0.. Record-Route: <sip:238014102959745@162.33.165.197;ftag=2b00806c-495b-b6c0-cceb-68293071984 3;lr=on>.. Via: SIP/2.0/UDP 162.33.165.197;branch=0.. Via: SIP/2.0/UDP 162.33.165.203.. Route: <sip:238014102959745@209.163.109.7:5067;maddr=209.163.109.7>.. From: "ToadNet Testbed 5" <sip:7003@sipdemo.routerboy.com>;tag=2b00806c-495b-b6c0-cceb-682930719843.. To: <sip:4102959745@sipdemo.routerboy.com>;tag=DDDC4048-2260.. Contact: <sip:7003@162.33.165.203> .. Call-ID: 5aa02b3f-1d95-0f2f-dd55-aca6806cbcd7(a)162.33.165.203.. CSeq: 63249 ACK.. User-Agent: Grandstream SIP UA 1.0.3.81.. Max-Forwards: 69.. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE.. Content-Length: 0.. Route: <sip:238014102959745@209.163.109.9:5060>.. P-hint: rr-enforced.. .. # U 162.33.165.196:29479 -> 162.33.165.197:5060 .. # U 162.33.165.196:29479 -> 162.33.165.197:5060 REGISTER sip:sipdemo.routerboy.com SIP/2.0.. Via: SIP/2.0/UDP 192.168.0.2:5060;rport;branch=z9hG4bKC78EBEF51E6D4C34BC4E11711EE4502D.. From: ser <sip:7000@sipdemo.routerboy.com>.. To: ser <sip:7000@sipdemo.routerboy.com>.. Contact: "ser" <sip:7000@192.168.0.2:5060>.. Call-ID: 6FD5B609AF1C44AB8A3BF67AE2147DA7(a)sipdemo.routerboy.com.. CSeq: 2471 REGISTER.. Expires: 1800.. Max-Forwards: 70.. User-Agent: X-Lite build 1082.. Content-Length: 0.. .. # U 162.33.165.197:5060 -> 162.33.165.196:29479 SIP/2.0 200 OK.. Via: SIP/2.0/UDP 192.168.0.2:5060;rport=29479;branch=z9hG4bKC78EBEF51E6D4C34BC4E11711EE4502D; received=162.33.16 5.196.. From: ser <sip:7000@sipdemo.routerboy.com>.. To: ser <sip:7000@sipdemo.routerboy.com>;tag=b27e1a1d33761e85846fc98f5f3a7e58.5f3a.. Call-ID: 6FD5B609AF1C44AB8A3BF67AE2147DA7(a)sipdemo.routerboy.com.. CSeq: 2471 REGISTER.. Contact: <sip:7000@192.168.0.2:5060>;q=0.00;expires=1800.. Server: Sip EXpress router (0.8.12-dev-23-merged (i386/linux)).. Content-Length: 0.. Warning: 392 162.33.165.197:5060 "Noisy feedback tells: pid=8063 req_src_ip=162.33.165.196 req_src_port=29479 in_uri=sip:sipdemo.routerboy.com ou t_uri=sip:sipdemo.routerboy.com via_cnt==1".. .. # U 209.163.109.7:7021 -> 162.33.165.197:5060 BYE sip:4102959745@162.33.165.197:5060 SIP/2.0.. Via: SIP/2.0/UDP 209.163.109.7:7021;branch=z9hG4bK4408d945c5dec2a7eb659cd67e038a06.4.. Via: SIP/2.0/UDP 209.163.109.7:5067;branch=z9hG4bKe55a1f5fae81e5bff090c078ade18373.2.. Via: SIP/2.0/UDP 209.163.109.9:5060.. To: "ToadNet Testbed 5"<sip:7003@sipdemo.routerboy.com>;tag=2b00806c-495b-b6c0-cceb-682930719843. . From: <sip:4102959745@sipdemo.routerboy.com>;tag=DDDC4048-2260.. Date: Fri, 05 Dec 2003 17:46:23 GMT.. Call-ID: 5aa02b3f-1d95-0f2f-dd55-aca6806cbcd7(a)162.33.165.203.. CSeq: 101 BYE.. Max-Forwards: 5.. Route: <sip:7003@162.33.165.203:5060>.. Timestamp: 1070646430.. User-Agent: Cisco-SIPGateway/IOS-12.x.. Content-Length: 0.. .. # U 162.33.165.197:5060 -> 209.163.109.7:7021 BYE sip:7003@209.163.109.7:7021 SIP/2.0.. Record-Route: <sip:4102959745@162.33.165.197;ftag=DDDC4048-2260;lr=on>.. Via: SIP/2.0/UDP 162.33.165.197;branch=z9hG4bKf207.24777037.0.. Via: SIP/2.0/UDP 209.163.109.7:7021;branch=z9hG4bK4408d945c5dec2a7eb659cd67e038a06.4.. Via: SIP/2.0/UDP 209.163.109.7:5067;branch=z9hG4bKe55a1f5fae81e5bff090c078ade18373.2.. Via: SIP/2.0/UDP 209.163.109.9:5060.. To: "ToadNet Testbed 5"<sip:7003@sipdemo.routerboy.com>;tag=2b00806c-495b-b6c0-cceb-682930719843. . From: <sip:4102959745@sipdemo.routerboy.com>;tag=DDDC4048-2260.. Date: Fri, 05 Dec 2003 17:46:23 GMT.. Call-ID: 5aa02b3f-1d95-0f2f-dd55-aca6806cbcd7(a)162.33.165.203.. CSeq: 101 BYE.. Max-Forwards: 4.. Timestamp: 1070646430.. User-Agent: Cisco-SIPGateway/IOS-12.x.. Content-Length: 0.. P-hint: rr-enforced.. .. # U 209.163.109.7:7021 -> 162.33.165.197:5060 SIP/2.0 482 Loop Detected.. Via: SIP/2.0/UDP 162.33.165.197:5060;branch=z9hG4bKf207.24777037.0.. Via: SIP/2.0/UDP 209.163.109.7:7021;branch=z9hG4bK4408d945c5dec2a7eb659cd67e038a06.4.. Via: SIP/2.0/UDP 209.163.109.7:5067;branch=z9hG4bKe55a1f5fae81e5bff090c078ade18373.2.. Via: SIP/2.0/UDP 209.163.109.9:5060.. To: "ToadNet Testbed 5"<sip:7003@sipdemo.routerboy.com>;tag=2b00806c-495b-b6c0-cceb-682930719843. . From: <sip:4102959745@sipdemo.routerboy.com>;tag=DDDC4048-2260.. Call-ID: 5aa02b3f-1d95-0f2f-dd55-aca6806cbcd7(a)162.33.165.203.. CSeq: 101 BYE.. Content-Length: 0.. .. # U 162.33.165.197:5060 -> 209.163.109.7:7021 SIP/2.0 482 Loop Detected.. Via: SIP/2.0/UDP 209.163.109.7:7021;branch=z9hG4bK4408d945c5dec2a7eb659cd67e038a06.4.. Via: SIP/2.0/UDP 209.163.109.7:5067;branch=z9hG4bKe55a1f5fae81e5bff090c078ade18373.2.. Via: SIP/2.0/UDP 209.163.109.9:5060.. To: "ToadNet Testbed 5"<sip:7003@sipdemo.routerboy.com>;tag=2b00806c-495b-b6c0-cceb-682930719843. . From: <sip:4102959745@sipdemo.routerboy.com>;tag=DDDC4048-2260.. Call-ID: 5aa02b3f-1d95-0f2f-dd55-aca6806cbcd7(a)162.33.165.203.. CSeq: 101 BYE.. Content-Length: 0.. .. # U 162.33.165.203:5060 -> 162.33.165.197:5060 BYE sip:238014102959745@209.163.109.9:5060 SIP/2.0.. Via: SIP/2.0/UDP 162.33.165.203.. Route: <sip:4102959745@162.33.165.197;ftag=2b00806c-495b-b6c0-cceb-682930719843;lr= on>.. Route: <sip:4102959745@209.163.109.7:7021;maddr=209.163.109.7>.. Route: <sip:238014102959745@209.163.109.7:5067;maddr=209.163.109.7>.. From: "ToadNet Testbed 5" <sip:7003@sipdemo.routerboy.com>;tag=2b00806c-495b-b6c0-cceb-682930719843.. To: <sip:4102959745@sipdemo.routerboy.com>;tag=DDDC4048-2260.. Contact: <sip:7003@162.33.165.203>.. Call-ID: 5aa02b3f-1d95-0f2f-dd55-aca6806cbcd7(a)162.33.165.203.. CSeq: 63250 BYE.. User-Agent: Grandstream SIP UA 1.0.3.81.. Max-Forwards: 70.. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE.. Content-Length: 0.. .. # U 162.33.165.197:5060 -> 209.163.109.7:7021 BYE sip:4102959745@209.163.109.7:7021;maddr=209.163.109.7 SIP/2.0.. Record-Route: <sip:238014102959745@162.33.165.197;ftag=2b00806c-495b-b6c0-cceb-68293071984 3;lr=on>.. Via: SIP/2.0/UDP 162.33.165.197;branch=z9hG4bKdbc4.259c2b9.0.. Via: SIP/2.0/UDP 162.33.165.203.. Route: <sip:238014102959745@209.163.109.7:5067;maddr=209.163.109.7>.. From: "ToadNet Testbed 5" <sip:7003@sipdemo.routerboy.com>;tag=2b00806c-495b-b6c0-cceb-682930719843.. To: <sip:4102959745@sipdemo.routerboy.com>;tag=DDDC4048-2260.. Contact: <sip:7003@162.33.165.203>.. Call-ID: 5aa02b3f-1d95-0f2f-dd55-aca6806cbcd7(a)162.33.165.203.. CSeq: 63250 BYE.. User-Agent: Grandstream SIP UA 1.0.3.81.. Max-Forwards: 69.. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE.. Content-Length: 0.. Route: <sip:238014102959745@209.163.109.9:5060>.. P-hint: rr-enforced.. .. # U 162.33.165.197:5060 -> 209.163.109.7:7021 BYE sip:4102959745@209.163.109.7:7021;maddr=209.163.109.7 SIP/2.0.. Record-Route: <sip:238014102959745@162.33.165.197;ftag=2b00806c-495b-b6c0-cceb-68293071984 3;lr=on>.. Via: SIP/2.0/UDP 162.33.165.197;branch=z9hG4bKdbc4.259c2b9.0.. Via: SIP/2.0/UDP 162.33.165.203.. Route: <sip:238014102959745@209.163.109.7:5067;maddr=209.163.109.7>.. From: "ToadNet Testbed 5" <sip:7003@sipdemo.routerboy.com>;tag=2b00806c-495b-b6c0-cceb-682930719843.. To: <sip:4102959745@sipdemo.routerboy.com>;tag=DDDC4048-2260.. Contact: <sip:7003@162.33.165.203>.. Call-ID: 5aa02b3f-1d95-0f2f-dd55-aca6806cbcd7(a)162.33.165.203.. CSeq: 63250 BYE.. User-Agent: Grandstream SIP UA 1.0.3.81.. Max-Forwards: 69.. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE.. Content-Length: 0.. Route: <sip:238014102959745@209.163.109.9:5060>.. P-hint: rr-enforced.. .. # U 209.163.109.7:7021 -> 162.33.165.197:5060 SIP/2.0 481 Call Leg/Transaction Does Not Exist.. Via: SIP/2.0/UDP 162.33.165.197:5060;branch=z9hG4bKdbc4.259c2b9.0.. Via: SIP/2.0/UDP 162.33.165.203:5060.. To: <sip:4102959745@sipdemo.routerboy.com>;tag=DDDC4048-2260.. From: "ToadNet Testbed 5"<sip:7003@sipdemo.routerboy.com>;tag=2b00806c-495b-b6c0-cceb-682930719843. . Call-ID: 5aa02b3f-1d95-0f2f-dd55-aca6806cbcd7(a)162.33.165.203.. CSeq: 63250 BYE.. Content-Length: 0.. .. # U 162.33.165.197:5060 -> 162.33.165.203:5060 SIP/2.0 481 Call Leg/Transaction Does Not Exist.. Via: SIP/2.0/UDP 162.33.165.203:5060.. To: <sip:4102959745@sipdemo.routerboy.com>;tag=DDDC4048-2260.. From: "ToadNet Testbed 5"<sip:7003@sipdemo.routerboy.com>;tag=2b00806c-495b-b6c0-cceb-682930719843. . Call-ID: 5aa02b3f-1d95-0f2f-dd55-aca6806cbcd7(a)162.33.165.203.. CSeq: 63250 BYE.. Content-Length: 0.. .. # U 162.33.165.200:5060 -> 162.33.165.197:5060 .. exit

20 years, 9 months

[Serusers] kphone registration problem

by Jorge Merlino

Hello: I'm trying to register kphone to my ser proxy using authentication with user and password but I can't make it work. In the identity configuration of kphone I have set "jorge" in "user part of SIP URL" and "authentication username" and I have set "asterix" in "host part of SIP URL" and "outbound proxy" The same configuration with user and password works fine with windows messenger 4.7 --------------------------------------------------------------------------------------------------------- The error messages I get from kphone are these: SipClient: Sending: 10:09:30.743 -------------------------------- REGISTER sip:asterix SIP/2.0 Via: SIP/2.0/UDP 192.168.2.175:5062 CSeq: 928 REGISTER To: "jorge" <sip:jorge@asterix> Expires: 900 From: "jorge" <sip:jorge@asterix> Call-ID: 1234450217(a)192.168.2.175 Content-Length: 0 User-Agent: KPhone/3.12 Event: registration Allow-Events: presence Contact: "jorge" <sip:jorge@192.168.2.175:5062;transport=udp>;q=1.0;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK" SipClient: Receiving message... SipClient: Received: 10:09:30.755 --------------------------------- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.175:5062 CSeq: 928 REGISTER To: "jorge" <sip:jorge@asterix>;tag=b27e1a1d33761e85846fc98f5f3a7e58.fbe3 From: "jorge" <sip:jorge@asterix> Call-ID: 1234450217(a)192.168.2.175 WWW-Authenticate: Digest realm="asterix", nonce="3fcb14af87f1bd1110526a8d29ad258492f165bd" Server: Sip EXpress router (0.8.11 (i386/linux)) Content-Length: 0 --------------------------------------------------------------------------------------------------------- And my ser.cfg file is this: alias="asterix" alias=192.168.2.176 sip_warning=no check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/ser_fifo" # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; record_route(); if (loose_route()) { t_relay(); break; }; if (uri==myself) { if (uri =~ "^sip:9[0-9]*@asterix") { log ("enviando al asterisk\n"); t_relay_to_udp("192.168.2.175", "5060"); break; }; if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("asterix", "subscriber")) { www_challenge("asterix", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; if (!t_relay()) { sl_reply_error(); }; }

20 years, 9 months

RE: [Serusers] Two RTP Proxies | Traffic not forwarded

by dhiraj.2.bhuyan＠bt.com

Hi Jiri, Yes one RTP-Proxy should be good enough. However I am trying to do a hop-by-hop IPSec (without IKE) thing. In a small setup, I can ofcourse do an IPSec between RTP-Proxy 1 and UA2 (that is implementing what you suggested - just one RTP Proxy on the path). However, in a real world this will not be possible - since dynamically setting up this IPSec session will be virtually impossible (UA2 does not know who SER/RTP Proxy 1 is). It is however possible to set up an IPSec (no IKE, symmetric keys) session between UA2 and SER/RTP Proxy 2 (since that's his service provider). Also, there can be an agreement between service providers on how to set up IPSec between them (SER1 and SER2). So what I am trying to do is something like - [UA1]<----IPSec---->[SER/RTP proxy1]<----IPSec--->[SER/RTY Proxy 2]<---IPSec---->[UA2] Its very difficult to explain why I am trying to do something like this (maybe someone having a good knowledge of 3GPP will understand me better) - but please excuse me for the moment, I need to do this :) Dhiraj -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]On Behalf Of Jiri Kuthan Sent: 08 December 2003 10:56 To: Bhuyan,D,Dhiraj,XVR3 R; serusers(a)lists.iptel.org Subject: Re: [Serusers] Two RTP Proxies | Traffic not forwarded The easiest way you can do is label INVITEs which are rtp-proxied with a special header (append_hf('P-Nat-foo: bar\r\n')) and don't rtp-proxy an INVITE if such a header is already present (is_present_hf("P-Nat-foo")). The rationale is one RTP proxy in the path is good enough. There is lot of other heuristics you could do, but that would be no longer a short email. -jiri At 11:43 AM 12/8/2003, dhiraj.2.bhuyan(a)bt.com wrote: >Hello List, > >I have the following settings - > >[ UA1 ] <-------> [ SER/RTP Proxy 1] <--------> [ SER/RTP Proxy 2] <-------> [ UA2 ] > >Both SERs are configured to force RTP Proxy whether or not the UA is behind NAT (I know this is not required, but is necessary for what I am trying to do - basically some IPSec and 3GPP things). UA1 is registered to SER1 and UA2 is registered to SER2. I tried setting up a call between UA1 and UA2 - but could not hear anything on either ends. I monitored the network traffic (SIP and SDP messages appear to be fine - ethereal dump attached) and found that the RTP Proxy 1 does not forward the traffic to RTP Proxy 2 and vice versa. RTP traffic from UA1 goes to RTP Proxy 1 and does not go any further. Similary, RTP traffic from UA2 just hangs at RTP Proxy 2. I am aware of the fact that the RTP proxy waits for one UDP packet from both UAs before forwarding the RTP traffic only if both the UAs have SIP URI that belong to the same domain - otherwise they should forward the traffic immediately. In my test case, the RTP proxies should have forwarded the RTP traffic without waiting (since UA1 and UA2 belong to two different domain) - and this appears to be failing. Can anyone throw some light on this? I am attaching my SER config file (both SERs have exactly the same config file - other than the alias domain part). I am using SER from the CVS (1st December). > >I am attaching the ethereal dump of the SIP messaging (just to confirm that SIP/SDP exchanges went fine). Note that - >UA1 - 132.146.196.18 >UA2 - 132.146.196.49 >SER1 - 132.146.196.75 (bump1.alien.bt.co.uk) >SER2 - 132.146.196.49 (eniac.alien.bt.co.uk) > >Thanks, > >Dhiraj Bhuyan >Network Security Specialist, >BT Exact Business Assurance Solutions > >Tel: +44 1473 643932 >Mob: +44 7962 012145 >Email: dhiraj.2.bhuyan(a)bt.com > > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/ _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

20 years, 9 months

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sr-users December 2003