sr-users November 2005

sr-users@lists.kamailio.org

245 participants
570 discussions

by Greger V. Teigre

Rajesh, Again, copy the list. The config you sent is NOT the onsip.org config and as I have said: You call force_rtp_proxy several times. g-) ----- Original Message ----- From: rajesh volluru To: Greger V. Teigre Sent: Thursday, November 10, 2005 10:52 PM Subject: Re: [Serusers] problem with rtpproxy hi I see that ser is appending to new IP address multiple times in the Invite message instead of rewriting the address. I had copied entire ser.cfg iam using. Please look at it and let me know if anything is wrong. sdpData arg - v=0 o=Rajesh 0 0 IN IP4 192.168.1.223 s=- c=IN IP4 68.225.20.25069.229.23.5769.229.23.5769.229.23.57 t=0 0 m=audio 351843518435236 RTP/AVP 0 3 4 5 6 8 15 18 m=video 35238 RTP/AVP 26 34 31 a=recvonly a=direction:active a=oldmediaip:192.168.1.223 a=nortpproxy:yes a=nortpproxy:yes java.lang.NumberFormatException: For input string: "351843518435236" Thanks Rajesh # $Id: nat-rtpproxy.cfg 9 2005-08-19 15:30:55Z /CN=Greger V. Teigre/emailAddress=greger(a)onsip.org $ debug=9 fork=yes log_stderror=yes listen=69.229.23.57 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" 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/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 60) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("69.229.23.57:5060;nat=yes"); if(method == "INVITE") { fix_nated_sdp("3"); force_rtp_proxy(); t_on_reply("1"); }; } else if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); end_media_session(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); #fix_nated_sdp("3"); }; force_rtp_proxy("l"); if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); break; }; if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("adsl-69-229-23-57.dsl.irvnca.pacbell.net","subscriber")) { www_challenge("adsl-69-229-23-57.dsl.irvnca.pacbell.net","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("adsl-69-229-23-57.dsl.irvnca.pacbell.net","subscriber")) { www_challenge("adsl-69-229-23-57.dsl.irvnca.pacbell.net","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("adsl-69-229-23-57.dsl.irvnca.pacbell.net","subscriber")) { proxy_challenge("adsl-69-229-23-57.dsl.irvnca.pacbell.net","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); use_media_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } "Greger V. Teigre" <greger(a)teigre.com> wrote: Please copy the list. If you use a vanilla onsip.org config, it's pretty thoroughly tested by now, so I would be tempted to say that there is something wrong with the client or the setup of the client. g-) ----- Original Message ----- From: rajesh volluru To: Greger V. Teigre Sent: Wednesday, November 09, 2005 2:04 AM Subject: Re: [Serusers] problem with rtpproxy hi Thanks for your reply. I did configure ser.cfg as said at onsip.org for rtp proxy. But, i still get below exception when I am behind NAT. net.java.sip.communicator.media.MediaException: Incorrect SDP data! at net.java.sip.communicator.media.MediaManager.openMediaStreams(MediaManager.java:307) at net.java.sip.communicator.SipCommunicator.callStateChanged(SipCommunicator.java:758) at net.java.sip.communicator.sip.Call.fireCallStatusChangedEvent(Call.java:250) at net.java.sip.communicator.sip.Call.setState(Call.java:156) at net.java.sip.communicator.sip.CallProcessing.processAck(CallProcessing.java:566) at net.java.sip.communicator.sip.SipManager.processRequest(SipManager.java:1532) at gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:204) at gov.nist.javax.sip.EventScanner.run(EventScanner.java:352) at java.lang.Thread.run(Unknown Source) Caused by: javax.sdp.SdpParseException: Could not parse message at javax.sdp.SdpFactory.createSessionDescription(SdpFactory.java:114) at net.java.sip.communicator.media.MediaManager.openMediaStreams(MediaManager.java:302) ... 8 more Please let me know the possible reason for this error. When I start ser, it shows the message ' RTP proxy found. support for RTP enabled'. Thanks Rajesh "Greger V. Teigre" <greger(a)teigre.com> wrote: Are you using the configs from onsip.org? If not, you may find some help there ;-) g-) ----- Original Message ----- From: rajesh volluru To: Sip Users Sent: Tuesday, November 08, 2005 12:37 AM Subject: [Serusers] problem with rtpproxy hi I had installed ser with media and rtp proxy. Below is the output that i get when I start ser. However, iam not able receive any audio transmission when I connect through sip communicator when iam behind NAT. I am able to register with the server. 0(0) fixing /usr/local/lib/ser/modules/maxfwd.so mf_process_maxfwd_header 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so client_nat_test 0(0) fixing /usr/local/lib/ser/modules/textops.so search 0(0) module - fixing ^Record-Route: 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so fix_contact 0(0) fixing /usr/local/lib/ser/modules/nathelper.so fix_nated_sdp 0(0) fixing /usr/local/lib/ser/modules/rr.so record_route 0(0) fixing /usr/local/lib/ser/modules/rr.so loose_route 0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf 0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf 0(0) fixing /usr/local/lib/ser/modules/registrar.so lookup 0(0) get_connection(): Connection not found in the pool 0(0) preload_udomain():! ! Table is empty 0(0) release_connection(): Removing connection from the pool 0(0) fixing /usr/local/lib/ser/modules/auth_db.so www_authorize 0(0) get_connection(): Connection not found in the pool 0(0) release_connection(): Removing connection from the pool 0(0) fixing /usr/local/lib/ser/modules/auth.so www_challenge 0(0) fixing /usr/local/lib/ser/modules/registrar.so save 0(0) get_connection(): Connection not found in the pool 0(0) release_connection(): Removing connection from the pool 0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf 0(0) fixing /usr/local/lib/ser/modules/registrar.so lookup 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf 0(0) fixing /usr/local/lib/ser/modules/textops.so search 0(0) module - fixing ^Route: 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so use_media_proxy 0(0) fixing /usr/local/lib/ser/modules/tm.so t_on_reply 0(0) fixing /usr/local/lib/ser/modules/tm.so t_relay 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_reply_error 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so fix_contact 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so use_media_proxy 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so client_nat_test 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so fix_contact 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) DEBUG: udp_init: trying SO_RCVBUF: 221184 0(0) DEBUG: setting SO_RCVBUF; set=221184,verify=221184 0(0) DEBUG: udp_init: trying SO_RCVBUF: 223232 0(0) DEBUG: setting SO_RCVBUF; set=223232,verify=221184 0(0) DEBUG: setting SO_RCVBUF has no effect 0(0) INFO: udp_init: SO_RCVBUF i! ! s finally 221184 0(0) DBG: open_uac_fifo: opening fifo... 0(0) DEBUG: FIFO created @ /tmp/ser_fifo 0(0) DEBUG: fifo /tmp/ser_fifo opened, mode=432 0(0) init_unixsock_socket: No unix domain socket will be opened 2(0) INFO: fifo process starting: 19946 2(0) DEBUG: init_mod_child (-2): tm 2(0) DEBUG: callid: '57b7f7f7-0(a)69.229.23.57' 2(0) DEBUG: init_mod_child (-2): usrloc 2(0) get_connection(): Connection not found in the pool 0(19941) DEBUG: init_mod_child (1): tm 0(19941) DEBUG: callid: '57b7f7f7-19941(a)69.229.23.57' 0(19941) DEBUG: init_mod_child (1): usrloc 0(19941) get_connection(): Connection not found in the pool 1(0) DEBUG: init_mod_child (-1): tm 1(19945) DEBUG: callid: '57b7f7f7-19945(a)69.229.23.57' 1(19945) DEBUG: init_mod_child (-1): usrloc 1(19945) get_connection(): Connection not found in the pool 0(19941) DEBUG: init_mod_child (1): auth_db 0(19941) get_connection(): Connection not found in the pool 1(19945) DEBUG: init_mod_child (-1): auth_db 1(19945) get_connection(): Connection not found in the pool 0(19941) DEBUG: init_mod_child (1): domain 0(19941) get_connection(): Connection found in the pool 0(19941) DEBUG: init_mod_child (1): nathelper 2(19946) DEBUG: init_mod_child (-2): auth_db 2(19946) get_connection(): Connection not found in the pool 1(19945) DEBUG: init_mod_child (-1): domain 1(19945) DEBUG: init_mod_child (-1): nathelper 0(19941) rtpp_test: RTP proxy found, support for it enabled 1(19945) rtpp_test: RTP proxy found, support for it enabled 2(19946) DEBUG: init_mod_child (-2): domain 2(19946) DEBUG: init_mod_child (-2): nathelper 2(19946) rtpp_test: RTP proxy found, support for it enabled 2(19946) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(19946) DEBUG: register_fifo_cmd: new command (print) registered 2(19946) DEBUG: register_fifo_cmd: new command (uptime) registered 2(19946) DEBUG: register_fifo_cmd: new command (version) registered 2(19946) DEBUG: register_fifo_cmd: new command (pwd) registered 2(19946) DEBUG: register_fifo_cmd: new command (arg) registered 2(19946) DEBUG: register_fifo_cmd: new command (which) registered 2(19946) DEBUG: register_fifo_cmd: new command (ps) registered 2(19946) DEBUG: register_fifo_cmd: new command (kill) registered 2(19946) WARNING: no fifo_db_url given - fifo DB commands disabled! Let me know if you find any configuration error in SER. Please let me if you need any more information. I am trying to configure SER from long time. Please help me. Thanks Rajesh ---------------------------------------------------------------------- Enjoy this Diwali with Y! India Click here ---------------------------------------------------------------------- Enjoy this Diwali with Y! India Click here ---------------------------------------------------------------------- _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers -------------------------------------------------------------------------- Enjoy this Diwali with Y! India Click here ------------------------------------------------------------------------------ Enjoy this Diwali with Y! India Click here

18 years, 7 months

RE: [Serusers] SEMS:announcement plug in

by rajesh.rajguru＠wipro.com

Hi, I get following from SEMS's log, but still announcement is not being heared What could be the problem ---------------------------------------------- (23556) DEBUG: write_to_fifo (SerClient.cpp:252): Write to fifo: completed (23556) DEBUG: getReturnCode (AmRequest.cpp:391): response from Ser: 200 fifo_t reply succeeded (23556) DEBUG: setRAddr (AmRtpStream.cpp:402): RTP remote address set to 10.111 17.16:32900 (23556) DEBUG: negotiate (AmSession.cpp:184): Sending Rtp data to 10.111.17.16/ 2900 (23556) DEBUG: onSessionStart (Announcement.cpp:116): starting sending announce ..(/usr/local/lib/sems/audio/6789.wav) (23556) DEBUG: trying to read WAV file (23556) DEBUG: tag = <RIFF> (23556) DEBUG: file size = <129710> (23556) DEBUG: tag = <WAVE> (23556) DEBUG: tag = <fmt > (23556) DEBUG: chunk_size = <18> (23556) DEBUG: fmt = <01> (23556) DEBUG: channels = <1> (23556) DEBUG: rate = <8000> (23556) DEBUG: bits/sample = <16> (23556) DEBUG: tag = <fact> (23556) DEBUG: chunk size = <4> (23556) DEBUG: tag = <data> (23556) DEBUG: chunk size = <129660> (23556) DEBUG: ilbc with format parameters : '', mode=30. (23556) DEBUG: streamGet (AmAudio.cpp:548): 0-stuffing packet: adding 420 bytes (packet size=480) (23556) DEBUG: convert (AmAudio.cpp:246): end of stream (23556) DEBUG: onSessionStart (Announcement.cpp:118): ... announce reaches its nd. (23556) DEBUG: sendRequest (AmRequest.cpp:138): dummy sendrequest ----------------------------------------------------- Regards, Rajesh -----Original Message----- From: Xavier TRENTIN [mailto:xtrentin@softiphone.net] Sent: Tuesday, November 15, 2005 11:09 PM To: Rajesh Rajguru (WT01 - Voice & Next Generation Networks) Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] SEMS:announcement plug in Try to record your .wav in PCM 8Kbit/s format and configure your phone to use PCMU codec. B.R. xavier rajesh.rajguru(a)wipro.com a écrit : >*********************** >Pas de virus detecté dans l'attachement no filename > >Votre Mail a été vérifié par InterScan. >***********-*********** > > > >Hi, > >We are trying to forward INVITE to SEMS and play announcement. >It doesn't play anything. > >The UA which is sending INVITE is getting time out > >But the same .wav file for announcement if I open from sound recorder >manually ,then it plays. > >How to check that there is problem with codec ? > > >Regards, >Rajesh > >-----Original Message----- >From: Xavier TRENTIN [mailto:xtrentin@softiphone.net] > >Sent: Monday, November 14, 2005 10:00 PM >To: Rajesh Rajguru (WT01 - Voice & Next Generation Networks) >Cc: serusers(a)lists.iptel.org >Subject: Re: [Serusers] SEMS:announcement plug in > >Does it play nothing or does it hang off? >Perhaps the problem could be the Codec you used to record the file. > >B.R. >Xavier. > >rajesh.rajguru(a)wipro.com a écrit : > > > >>*********************** >>Pas de virus detecté dans l'attachement no filename >> >>Votre Mail a été vérifié par InterScan. >>***********-*********** >> >> >> >>Hi, >> >>In our ser.cfg following is mentioned for announcement >> >>---- >>if(!t_write_req("/tmp/am_fifo","announcement")) { >> t_reply("500","error contacting sems"); >>----- >> >>we are using fifo for communicating between ser and sems. >>ser and sems are running on the same server (Fedora) When we start >> >> > > > >>sems, the default announcement file mentioned in sems.conf >> >> > > > >>(default_announce=default_en.wav) is not played. >>Also there is no error in /var/log/messages wrt announcement plug in >> >>What could be the problem ? >> >>How does sems decides which sound recorder to play ? >> >> >>The ll /tmp/am_fifo shows following >> >> >>prw-r--r-- 1 root root 0 Nov 14 09:48 /tmp/am_fifo >> >>Regards, >>Rajesh >> >> >> >> >>Confidentiality Notice >> >> >>The information contained in this electronic message and any >> >> > > > >>attachments to this message are intended for the exclusive use of the >> >> > > > >>addressee(s) and may contain confidential or privileged information. >>If >> >> > > > >>you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments. >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> > > > > > > > > >Confidentiality Notice > > >The information contained in this electronic message and any >attachments to this message are intended for the exclusive use of the >addressee(s) and may contain confidential or privileged information. If >you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments. > > > Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

18 years, 7 months

[Serusers] MYSQL Keeps needing to get restarted

by Goran Donev

I have a SER server works fine when I start everything manually. Yet when the computer reboots I have to go in a restart the Mysql service again because when I try to create a user it says can not connect to Mysql database. Is there a anywhere I can have Linux automatically start Mysql.

18 years, 7 months

Re: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius Accounting

by 'Jan Janak'

Yes, this can happen if the callee sends BYE. In this case the URIs in >From and To header fields are swapped. The only SIP RADIUS related document (which has expired a long time ago) says that Calling-Station-Id is the URI from From header field of any SIP request. Thus -- according this I-D it is correct. But use of RADIUS with SIP is underspecified so there is no real standard for this. I don't think it matters that much, because other attributes, namely Acct-Session-Id are used to match start and stop radius requests. We could determine who sent the BYE from the value of ftag parameter in Route header field and swap the values, the question is whether this makes sense. The billing system could take all info from start requests and use the stop request to calculate the session duration only. Jan. On 16-11-2005 15:04, Lenir wrote: > Ok, that put the right things in place. However, here's a twist: > Either way, wit switching Sip-Translated-Request-URI and Called-Station-ID > around, or not, the CDRs are sent differently depending on who "hands up" > the call. For example, notice the User-Name, Calling-Station-ID and > Called-Station-ID: > > User 1000 calls 15615551212, and user 1000 hangs up: > ===================================================== > rad_recv: Accounting-Request packet from host 216.82.124.29:39228, id=206, > length=306 > Acct-Status-Type = Start > Service-Type = Sip-Session > Acct-Terminate-Cause = 200 > Error-Cause = Invite > User-Name = "1000(a)216.82.124.29" > Calling-Station-Id = "sip:1000@216.82.124.29" > Sip-Translated-Request-URI = > "sip:15617220966@216.82.124.29;user=phone" > Called-Station-Id = > "sip:949449852150#15617220966@216.82.124.30;user=phone" > Acct-Session-Id = "000d2890-d47f01f5-6410e8bd-4baf0435(a)66.232.8.116" > Sip-To-Tag = "33C5A3C-76A" > Sip-From-Tag = "000d2890d47f24f85e6c0b91-0092ab53" > Sip-CSeq = "102" > NAS-IP-Address = 216.82.124.29 > NAS-Port = 5060 > Acct-Delay-Time = 0 > rlm_sql (sql): Reserving sql socket id: 15 > rlm_sql (sql): Released sql socket id: 15 > Sending Accounting-Response of id 206 to 216.82.124.29:39228 > rad_recv: Accounting-Request packet from host 216.82.124.29:39228, id=207, > length=300 > Acct-Status-Type = Stop > Service-Type = Sip-Session > Acct-Terminate-Cause = 200 > Error-Cause = 8 > User-Name = "1000(a)216.82.124.29" > Calling-Station-Id = "sip:1000@216.82.124.29" > Sip-Translated-Request-URI = > "sip:15617220966@216.82.124.29;user=phone" > Called-Station-Id = > "sip:949449852150#15617220966@216.82.124.30:5060" > Acct-Session-Id = "000d2890-d47f01f5-6410e8bd-4baf0435(a)66.232.8.116" > Sip-To-Tag = "33C5A3C-76A" > Sip-From-Tag = "000d2890d47f24f85e6c0b91-0092ab53" > Sip-CSeq = "103" > NAS-IP-Address = 216.82.124.29 > NAS-Port = 5060 > Acct-Delay-Time = 0 > > > User 1000 calls 15615551212, and user 15615551212 hangs up: > =========================================================== > > rad_recv: Accounting-Request packet from host 216.82.124.29:39227, id=200, > length=306 > Acct-Status-Type = Start > Service-Type = Sip-Session > Acct-Terminate-Cause = 200 > Error-Cause = Invite > User-Name = "1000(a)216.82.124.29" > Calling-Station-Id = "sip:1000@216.82.124.29" > Sip-Translated-Request-URI = > "sip:15617220966@216.82.124.29;user=phone" > Called-Station-Id = > "sip:949449852150#15617220966@216.82.124.30;user=phone" > Acct-Session-Id = "000d2890-d47f01f4-372dff4b-793d4acc(a)66.232.8.116" > Sip-To-Tag = "33B37E4-77B" > Sip-From-Tag = "000d2890d47f24f639ccef77-4ef08b02" > Sip-CSeq = "102" > NAS-IP-Address = 216.82.124.29 > NAS-Port = 5060 > Acct-Delay-Time = 0 > rlm_sql (sql): Reserving sql socket id: 1 > rlm_sql (sql): Released sql socket id: 1 > Sending Accounting-Response of id 200 to 216.82.124.29:39227 > rad_recv: Accounting-Request packet from host 216.82.124.29:39227, id=201, > length=286 > Acct-Status-Type = Stop > Service-Type = Sip-Session > Acct-Terminate-Cause = 200 > Error-Cause = 8 > User-Name = "15617220966(a)216.82.124.29" > Calling-Station-Id = "sip:15617220966@216.82.124.29;user=phone" > Sip-Translated-Request-URI = "sip:1000@216.82.124.29" > Called-Station-Id = "sip:1000@66.232.8.116:5061" > Acct-Session-Id = "000d2890-d47f01f4-372dff4b-793d4acc(a)66.232.8.116" > Sip-To-Tag = "000d2890d47f24f639ccef77-4ef08b02" > Sip-From-Tag = "33B37E4-77B" > Sip-CSeq = "101" > NAS-IP-Address = 216.82.124.29 > NAS-Port = 5060 > Acct-Delay-Time = 0 > > > Shouldn't it always be that the User-Name attribute is the user who > initiated the call? > > Lenir > > -----Original Message----- > From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf > Of 'Jan Janak' > Sent: Wednesday, November 16, 2005 1:53 PM > To: lsantiago(a)globalgatewaycom.com > Cc: serdev(a)iptel.org; serusers(a)iptel.org; users(a)openser.org; > devel(a)openser.org > Subject: Re: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius > Accounting > > Either reconfigure the billing application to use > Sip-Translated-Request-URI (recommended) or change the ID of the > attribute in the radiusclient dictionary as suggested by Bogdan. > > In this particular example you should set Called-Station-Id to 107. > > Jan. > > On 16-11-2005 13:09, 'Lenir' wrote: > > Ok, so how do I tell acc.so to send Sip-Translated-Request-URI as the > > Called-Station-ID instead? > > > > Lenir Santiago, Partner > > Global Gateway Communications, Inc. > > 2500 Quantum Lakes Drive, Ste. 203 > > Boynton Beach, FL 33426 > > Office: 561-853-2213 > > Cell: 561-722-0966 > > Fax: 877-875-0060 > > http://www.globalgatewaycom.com > > > > -----Original Message----- > > From: 'Jan Janak' [mailto:jan@iptel.org] > > Sent: Wednesday, November 16, 2005 11:29 AM > > To: Bogdan-Andrei Iancu > > Cc: Lenir; 'Lenir Santiago'; serdev(a)iptel.org; serusers(a)iptel.org; > > devel(a)openser.org; users(a)openser.org > > Subject: Re: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius > > Accounting > > > > Why would you do something like this ? All the info that is needed is in > > Sip-Translated-Ruquest-ID below. You can see it in the RADIUS dump: > > > > Sip-Translated-Request-URI = "sip:949449852150#15615551212@gateway:5060" > > > > Jan. > > > > On 16-11-2005 16:50, Bogdan-Andrei Iancu wrote: > > > Hi Lenir, > > > > > > you can keep the modified To value into an AVP and use the extra > > > accounting feature to log it. See > > > http://www.openser.org/docs/modules/1.1.x/acc.html#AEN115 > > > and > > > http://www.openser.org/docs/modules/1.1.x/acc.html#AEN350 > > > > > > the problem is how to set it as Called-Station-ID attribute in order to > > > avoid overlapping. You can do a trick by playing with the radius > > > dictionary (on proxy side) : > > > changed the Called-Station-ID to a different ID attribute > > > set extra accounting with the original attribute ID of > > Called-Station-ID > > > in this way the acc will set de default TO as something else and you AVP > > > > as Called-Station-ID ;) > > > > > > regards, > > > bogdan > > > > > > > > > Lenir wrote: > > > > > > >Well, if userA dials to:30, 30 gets translated to:15615551212 and SER > > dials > > > >15615551212, how can I make the acc.so module send the modified > > > >to:15615551212 as the Called-Station-ID attribute? > > > > > > > >This is important for billing, specially if someone forwards to a long > > > >distance or international long distance number. > > > > > > > >-----Original Message----- > > > >From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@iptel.org] On > > > >Behalf Of Jan Janak > > > >Sent: Wednesday, November 16, 2005 3:15 AM > > > >To: Lenir Santiago > > > >Cc: serdev(a)iptel.org; serusers(a)iptel.org; users(a)openser.org; > > > >devel(a)openser.org > > > >Subject: [Serusers] Re: [Serdev] Inaccurate Radius Accounting > > > > > > > >Hello, > > > > > > > >Called-Station-Id attribute always contains the To URI (not > > > >Request-URI). You can find the outbound Request-URI in > > > >Sip-Translated-Request-URI attribute. > > > > > > > >Even if you rewrite To header field, it will never be accounted in > > > >Called-Station-Id. > > > > > > > > Jan. > > > > > > > >On 15-11-2005 21:59, Lenir Santiago wrote: > > > > > > > > > > > >>Hey guys, > > > >> > > > >> > > > >> > > > >>I'm using Radius to do pretty much everything regarding my users, i.e. > > > >>registrations, authorization, features and preferences. I implemented > my > > > >> > > > >> > > > >own > > > > > > > > > > > >>way of doing speeddial by using AVPs provided by radius and AVPOPs > > module, > > > >>Here is the snippet: > > > >> > > > >> > > > >> > > > >>AVP-SIP = speedial:30:15615551212 > > > >> > > > >> > > > >> > > > >> avp_subst("s:caller_speedial/i:99", "/(.*):(.*)/\2/"); > > > >>#15615551212 > > > >> > > > >> avp_printf("i:98","$rU:*"); #30 > > > >> > > > >> avp_printf("i:97","sip:$avp($temp2)@$td"); # > > > >>sip:15615551212@xx.xx.xx.xx > > > >> > > > >> avp_printf("i:96","<sip:$avp($temp2)@$td>\r\n"); # > > > >><sip:15615551212@xx.xx.xx.xx>. > > > >> > > > >> xlog("L_N"," SPEEDIALING: $fU is speed-dialing > > > >>$hdr(To)->$avp($temp2)\n"); > > > >> > > > >> if (avp_check("s:caller_speedial", "fm/$temp1/g")) { > > > >> > > > >> > > > >> > > > >> #avp_pushto("$To","i:96"); > > > >> > > > >> #avp_pushto("$ruri","i:97"); > > > >> > > > >> #avp_pushto("$duri","i:97"); > > > >> > > > >> #subst('/^To: (.*)(a)(.*)/To: "$avp($temp2)" > > > >><$avp($temp2)@\2/'); > > > >> > > > >> #append_hf("To: \"5615551212\" > > > >><15615551212(a)xx.xx.xx.xx>;user=phone"); > > > >> > > > >> > > > >>#subst_uri('/^sip:(.*)@(.*)$/sip:$avp($temp2)@\2/i'); > > > >> > > > >> #subst_user('/(.*)/$avp($temp2)/'); > > > >> > > > >> #remove_hf("To"); > > > >> > > > >> rewriteuser("15615551212"); > > > >> > > > >> xlog("L_N"," SPEEDIALING: $fU is dialing > > > >> > > > >> > > > >$hdr(To) > > > > > > > > > > > >>$avp($temp2)\n"); > > > >> > > > >> avp_delete("$temp1/g"); > > > >> > > > >> avp_delete("$temp2/g"); > > > >> > > > >> > > > >> > > > >> }; > > > >> > > > >> > > > >> > > > >>In this scenario, im forcing it instead of getting the value from AVPs > > > >> > > > >> > > > >just > > > > > > > > > > > >>for ease of showing my point. The problem that I'm having is that no > > > >> > > > >> > > > >matter > > > > > > > > > > > >>what I do to change the To header or change the R-URI, radius ALWAYS > > sends > > > >>the accounting records with the original R-URI, in my case > > 30(a)xx.xx.xx.xx, > > > >>how can I get acc.so to send the final resulting URI > > > >>(15615551212(a)xx.xx.xx.xx) in the Called-Station-Id AVP? > > > >> > > > >> > > > >> > > > >> > > > >> > > > >>rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, > id=97, > > > >>length=298 > > > >> > > > >> Acct-Status-Type = Start > > > >> > > > >> Service-Type = Sip-Session > > > >> > > > >> Acct-Terminate-Cause = 200 > > > >> > > > >> Error-Cause = Invite > > > >> > > > >> User-Name = "1000(a)xx.xx.xx.xx" > > > >> > > > >> Calling-Station-Id = "sip:1000@xx.xx.xx.xx" > > > >> > > > >> Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" > > > >> > > > >> Sip-Translated-Request-URI = > > > >>"sip:949449852150#15615551212@gateway;user=phone" > > > >> > > > >> Acct-Session-Id = > > > >> > > > >> > > > >"000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" > > > > > > > > > > > >> Sip-To-Tag = "230383F4-2AF" > > > >> > > > >> Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" > > > >> > > > >> Sip-CSeq = "102" > > > >> > > > >> NAS-IP-Address = xx.xx.xx.xx > > > >> > > > >> NAS-Port = 5060 > > > >> > > > >> Acct-Delay-Time = 0 > > > >> > > > >>rlm_sql (sql): Reserving sql socket id: 12 > > > >> > > > >>rlm_sql (sql): Released sql socket id: 12 > > > >> > > > >>Sending Accounting-Response of id 97 to xx.xx.xx.xx:39216 > > > >> > > > >>rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, > id=98, > > > >>length=292 > > > >> > > > >> Acct-Status-Type = Stop > > > >> > > > >> Service-Type = Sip-Session > > > >> > > > >> Acct-Terminate-Cause = 200 > > > >> > > > >> Error-Cause = 8 > > > >> > > > >> User-Name = "1000(a)xx.xx.xx.xx" > > > >> > > > >> Calling-Station-Id = "sip:1000@xx.xx.xx.xx" > > > >> > > > >> Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" > > > >> > > > >> Sip-Translated-Request-URI = > > > >>"sip:949449852150#15615551212@gateway:5060" > > > >> > > > >> Acct-Session-Id = > > > >> > > > >> > > > >"000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" > > > > > > > > > > > >> Sip-To-Tag = "230383F4-2AF" > > > >> > > > >> Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" > > > >> > > > >> Sip-CSeq = "103" > > > >> > > > >> NAS-IP-Address = xx.xx.xx.xx > > > >> > > > >> NAS-Port = 5060 > > > >> > > > >> Acct-Delay-Time = 0 > > > >> > > > >> > > > >> > > > >> > > > >> > > > >>Thanks in advance! > > > >> > > > >> > > > >> > > > >> > > > >> > > > >>Lenir > > > >>_______________________________________________ > > > >>Users mailing list > > > >>Users(a)openser.org > > > >>http://openser.org/cgi-bin/mailman/listinfo/users > > > >> > > > >> > > > >> > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

18 years, 7 months

Recall: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius Accounting

by Lenir

Lenir would like to recall the message, "[Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius Accounting".

18 years, 7 months

Re: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius Accounting

by 'Jan Janak'

Either reconfigure the billing application to use Sip-Translated-Request-URI (recommended) or change the ID of the attribute in the radiusclient dictionary as suggested by Bogdan. In this particular example you should set Called-Station-Id to 107. Jan. On 16-11-2005 13:09, 'Lenir' wrote: > Ok, so how do I tell acc.so to send Sip-Translated-Request-URI as the > Called-Station-ID instead? > > Lenir Santiago, Partner > Global Gateway Communications, Inc. > 2500 Quantum Lakes Drive, Ste. 203 > Boynton Beach, FL 33426 > Office: 561-853-2213 > Cell: 561-722-0966 > Fax: 877-875-0060 > http://www.globalgatewaycom.com > > -----Original Message----- > From: 'Jan Janak' [mailto:jan@iptel.org] > Sent: Wednesday, November 16, 2005 11:29 AM > To: Bogdan-Andrei Iancu > Cc: Lenir; 'Lenir Santiago'; serdev(a)iptel.org; serusers(a)iptel.org; > devel(a)openser.org; users(a)openser.org > Subject: Re: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius > Accounting > > Why would you do something like this ? All the info that is needed is in > Sip-Translated-Ruquest-ID below. You can see it in the RADIUS dump: > > Sip-Translated-Request-URI = "sip:949449852150#15615551212@gateway:5060" > > Jan. > > On 16-11-2005 16:50, Bogdan-Andrei Iancu wrote: > > Hi Lenir, > > > > you can keep the modified To value into an AVP and use the extra > > accounting feature to log it. See > > http://www.openser.org/docs/modules/1.1.x/acc.html#AEN115 > > and > > http://www.openser.org/docs/modules/1.1.x/acc.html#AEN350 > > > > the problem is how to set it as Called-Station-ID attribute in order to > > avoid overlapping. You can do a trick by playing with the radius > > dictionary (on proxy side) : > > changed the Called-Station-ID to a different ID attribute > > set extra accounting with the original attribute ID of > Called-Station-ID > > in this way the acc will set de default TO as something else and you AVP > > as Called-Station-ID ;) > > > > regards, > > bogdan > > > > > > Lenir wrote: > > > > >Well, if userA dials to:30, 30 gets translated to:15615551212 and SER > dials > > >15615551212, how can I make the acc.so module send the modified > > >to:15615551212 as the Called-Station-ID attribute? > > > > > >This is important for billing, specially if someone forwards to a long > > >distance or international long distance number. > > > > > >-----Original Message----- > > >From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@iptel.org] On > > >Behalf Of Jan Janak > > >Sent: Wednesday, November 16, 2005 3:15 AM > > >To: Lenir Santiago > > >Cc: serdev(a)iptel.org; serusers(a)iptel.org; users(a)openser.org; > > >devel(a)openser.org > > >Subject: [Serusers] Re: [Serdev] Inaccurate Radius Accounting > > > > > >Hello, > > > > > >Called-Station-Id attribute always contains the To URI (not > > >Request-URI). You can find the outbound Request-URI in > > >Sip-Translated-Request-URI attribute. > > > > > >Even if you rewrite To header field, it will never be accounted in > > >Called-Station-Id. > > > > > > Jan. > > > > > >On 15-11-2005 21:59, Lenir Santiago wrote: > > > > > > > > >>Hey guys, > > >> > > >> > > >> > > >>I'm using Radius to do pretty much everything regarding my users, i.e. > > >>registrations, authorization, features and preferences. I implemented my > > >> > > >> > > >own > > > > > > > > >>way of doing speeddial by using AVPs provided by radius and AVPOPs > module, > > >>Here is the snippet: > > >> > > >> > > >> > > >>AVP-SIP = speedial:30:15615551212 > > >> > > >> > > >> > > >> avp_subst("s:caller_speedial/i:99", "/(.*):(.*)/\2/"); > > >>#15615551212 > > >> > > >> avp_printf("i:98","$rU:*"); #30 > > >> > > >> avp_printf("i:97","sip:$avp($temp2)@$td"); # > > >>sip:15615551212@xx.xx.xx.xx > > >> > > >> avp_printf("i:96","<sip:$avp($temp2)@$td>\r\n"); # > > >><sip:15615551212@xx.xx.xx.xx>. > > >> > > >> xlog("L_N"," SPEEDIALING: $fU is speed-dialing > > >>$hdr(To)->$avp($temp2)\n"); > > >> > > >> if (avp_check("s:caller_speedial", "fm/$temp1/g")) { > > >> > > >> > > >> > > >> #avp_pushto("$To","i:96"); > > >> > > >> #avp_pushto("$ruri","i:97"); > > >> > > >> #avp_pushto("$duri","i:97"); > > >> > > >> #subst('/^To: (.*)(a)(.*)/To: "$avp($temp2)" > > >><$avp($temp2)@\2/'); > > >> > > >> #append_hf("To: \"5615551212\" > > >><15615551212(a)xx.xx.xx.xx>;user=phone"); > > >> > > >> > > >>#subst_uri('/^sip:(.*)@(.*)$/sip:$avp($temp2)@\2/i'); > > >> > > >> #subst_user('/(.*)/$avp($temp2)/'); > > >> > > >> #remove_hf("To"); > > >> > > >> rewriteuser("15615551212"); > > >> > > >> xlog("L_N"," SPEEDIALING: $fU is dialing > > >> > > >> > > >$hdr(To) > > > > > > > > >>$avp($temp2)\n"); > > >> > > >> avp_delete("$temp1/g"); > > >> > > >> avp_delete("$temp2/g"); > > >> > > >> > > >> > > >> }; > > >> > > >> > > >> > > >>In this scenario, im forcing it instead of getting the value from AVPs > > >> > > >> > > >just > > > > > > > > >>for ease of showing my point. The problem that I'm having is that no > > >> > > >> > > >matter > > > > > > > > >>what I do to change the To header or change the R-URI, radius ALWAYS > sends > > >>the accounting records with the original R-URI, in my case > 30(a)xx.xx.xx.xx, > > >>how can I get acc.so to send the final resulting URI > > >>(15615551212(a)xx.xx.xx.xx) in the Called-Station-Id AVP? > > >> > > >> > > >> > > >> > > >> > > >>rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, id=97, > > >>length=298 > > >> > > >> Acct-Status-Type = Start > > >> > > >> Service-Type = Sip-Session > > >> > > >> Acct-Terminate-Cause = 200 > > >> > > >> Error-Cause = Invite > > >> > > >> User-Name = "1000(a)xx.xx.xx.xx" > > >> > > >> Calling-Station-Id = "sip:1000@xx.xx.xx.xx" > > >> > > >> Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" > > >> > > >> Sip-Translated-Request-URI = > > >>"sip:949449852150#15615551212@gateway;user=phone" > > >> > > >> Acct-Session-Id = > > >> > > >> > > >"000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" > > > > > > > > >> Sip-To-Tag = "230383F4-2AF" > > >> > > >> Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" > > >> > > >> Sip-CSeq = "102" > > >> > > >> NAS-IP-Address = xx.xx.xx.xx > > >> > > >> NAS-Port = 5060 > > >> > > >> Acct-Delay-Time = 0 > > >> > > >>rlm_sql (sql): Reserving sql socket id: 12 > > >> > > >>rlm_sql (sql): Released sql socket id: 12 > > >> > > >>Sending Accounting-Response of id 97 to xx.xx.xx.xx:39216 > > >> > > >>rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, id=98, > > >>length=292 > > >> > > >> Acct-Status-Type = Stop > > >> > > >> Service-Type = Sip-Session > > >> > > >> Acct-Terminate-Cause = 200 > > >> > > >> Error-Cause = 8 > > >> > > >> User-Name = "1000(a)xx.xx.xx.xx" > > >> > > >> Calling-Station-Id = "sip:1000@xx.xx.xx.xx" > > >> > > >> Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" > > >> > > >> Sip-Translated-Request-URI = > > >>"sip:949449852150#15615551212@gateway:5060" > > >> > > >> Acct-Session-Id = > > >> > > >> > > >"000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" > > > > > > > > >> Sip-To-Tag = "230383F4-2AF" > > >> > > >> Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" > > >> > > >> Sip-CSeq = "103" > > >> > > >> NAS-IP-Address = xx.xx.xx.xx > > >> > > >> NAS-Port = 5060 > > >> > > >> Acct-Delay-Time = 0 > > >> > > >> > > >> > > >> > > >> > > >>Thanks in advance! > > >> > > >> > > >> > > >> > > >> > > >>Lenir > > >>_______________________________________________ > > >>Users mailing list > > >>Users(a)openser.org > > >>http://openser.org/cgi-bin/mailman/listinfo/users > > >> > > >> > > >> >

18 years, 7 months

Re: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius Accounting

by Bogdan-Andrei Iancu

Hi Lenir, you can keep the modified To value into an AVP and use the extra accounting feature to log it. See http://www.openser.org/docs/modules/1.1.x/acc.html#AEN115 and http://www.openser.org/docs/modules/1.1.x/acc.html#AEN350 the problem is how to set it as Called-Station-ID attribute in order to avoid overlapping. You can do a trick by playing with the radius dictionary (on proxy side) : changed the Called-Station-ID to a different ID attribute set extra accounting with the original attribute ID of Called-Station-ID in this way the acc will set de default TO as something else and you AVP as Called-Station-ID ;) regards, bogdan Lenir wrote: >Well, if userA dials to:30, 30 gets translated to:15615551212 and SER dials >15615551212, how can I make the acc.so module send the modified >to:15615551212 as the Called-Station-ID attribute? > >This is important for billing, specially if someone forwards to a long >distance or international long distance number. > >-----Original Message----- >From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@iptel.org] On >Behalf Of Jan Janak >Sent: Wednesday, November 16, 2005 3:15 AM >To: Lenir Santiago >Cc: serdev(a)iptel.org; serusers(a)iptel.org; users(a)openser.org; >devel(a)openser.org >Subject: [Serusers] Re: [Serdev] Inaccurate Radius Accounting > >Hello, > >Called-Station-Id attribute always contains the To URI (not >Request-URI). You can find the outbound Request-URI in >Sip-Translated-Request-URI attribute. > >Even if you rewrite To header field, it will never be accounted in >Called-Station-Id. > > Jan. > >On 15-11-2005 21:59, Lenir Santiago wrote: > > >>Hey guys, >> >> >> >>I'm using Radius to do pretty much everything regarding my users, i.e. >>registrations, authorization, features and preferences. I implemented my >> >> >own > > >>way of doing speeddial by using AVPs provided by radius and AVPOPs module, >>Here is the snippet: >> >> >> >>AVP-SIP = speedial:30:15615551212 >> >> >> >> avp_subst("s:caller_speedial/i:99", "/(.*):(.*)/\2/"); >>#15615551212 >> >> avp_printf("i:98","$rU:*"); #30 >> >> avp_printf("i:97","sip:$avp($temp2)@$td"); # >>sip:15615551212@xx.xx.xx.xx >> >> avp_printf("i:96","<sip:$avp($temp2)@$td>\r\n"); # >><sip:15615551212@xx.xx.xx.xx>. >> >> xlog("L_N"," SPEEDIALING: $fU is speed-dialing >>$hdr(To)->$avp($temp2)\n"); >> >> if (avp_check("s:caller_speedial", "fm/$temp1/g")) { >> >> >> >> #avp_pushto("$To","i:96"); >> >> #avp_pushto("$ruri","i:97"); >> >> #avp_pushto("$duri","i:97"); >> >> #subst('/^To: (.*)(a)(.*)/To: "$avp($temp2)" >><$avp($temp2)@\2/'); >> >> #append_hf("To: \"5615551212\" >><15615551212(a)xx.xx.xx.xx>;user=phone"); >> >> >>#subst_uri('/^sip:(.*)@(.*)$/sip:$avp($temp2)@\2/i'); >> >> #subst_user('/(.*)/$avp($temp2)/'); >> >> #remove_hf("To"); >> >> rewriteuser("15615551212"); >> >> xlog("L_N"," SPEEDIALING: $fU is dialing >> >> >$hdr(To) > > >>$avp($temp2)\n"); >> >> avp_delete("$temp1/g"); >> >> avp_delete("$temp2/g"); >> >> >> >> }; >> >> >> >>In this scenario, im forcing it instead of getting the value from AVPs >> >> >just > > >>for ease of showing my point. The problem that I'm having is that no >> >> >matter > > >>what I do to change the To header or change the R-URI, radius ALWAYS sends >>the accounting records with the original R-URI, in my case 30(a)xx.xx.xx.xx, >>how can I get acc.so to send the final resulting URI >>(15615551212(a)xx.xx.xx.xx) in the Called-Station-Id AVP? >> >> >> >> >> >>rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, id=97, >>length=298 >> >> Acct-Status-Type = Start >> >> Service-Type = Sip-Session >> >> Acct-Terminate-Cause = 200 >> >> Error-Cause = Invite >> >> User-Name = "1000(a)xx.xx.xx.xx" >> >> Calling-Station-Id = "sip:1000@xx.xx.xx.xx" >> >> Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" >> >> Sip-Translated-Request-URI = >>"sip:949449852150#15615551212@gateway;user=phone" >> >> Acct-Session-Id = >> >> >"000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" > > >> Sip-To-Tag = "230383F4-2AF" >> >> Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" >> >> Sip-CSeq = "102" >> >> NAS-IP-Address = xx.xx.xx.xx >> >> NAS-Port = 5060 >> >> Acct-Delay-Time = 0 >> >>rlm_sql (sql): Reserving sql socket id: 12 >> >>rlm_sql (sql): Released sql socket id: 12 >> >>Sending Accounting-Response of id 97 to xx.xx.xx.xx:39216 >> >>rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, id=98, >>length=292 >> >> Acct-Status-Type = Stop >> >> Service-Type = Sip-Session >> >> Acct-Terminate-Cause = 200 >> >> Error-Cause = 8 >> >> User-Name = "1000(a)xx.xx.xx.xx" >> >> Calling-Station-Id = "sip:1000@xx.xx.xx.xx" >> >> Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" >> >> Sip-Translated-Request-URI = >>"sip:949449852150#15615551212@gateway:5060" >> >> Acct-Session-Id = >> >> >"000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" > > >> Sip-To-Tag = "230383F4-2AF" >> >> Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" >> >> Sip-CSeq = "103" >> >> NAS-IP-Address = xx.xx.xx.xx >> >> NAS-Port = 5060 >> >> Acct-Delay-Time = 0 >> >> >> >> >> >>Thanks in advance! >> >> >> >> >> >>Lenir >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> >>

18 years, 7 months

[Serusers] ser-0.9.3 vs ser-0.9.4

by Chia Huey Lim

Hi list, Is there any major difference for www_authorize and www_challenge function in ser 0.9.3 and ser 0.9.4 ? When I am using ser 0.9.3, have no problem with the registration, but when I upgrade to ser 0.9.4 version, Most of the UAs failed to register, and the 401 rate is extremely high. Once I downgrade back to ser 0.9.3, then no more issue. I m using if (!www_authorize("voip","subscriber")) { www_challenge("voip","0"); break; }; The SIP_DOMAIN is set to voip Anyone facing this issue too? Thanks Regards, Chia -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.3/172 - Release Date: 11/15/2005

18 years, 7 months

[Serusers] ser-0.9.3 vs ser-0.9.4

by Chia Huey

18 years, 7 months

[Serusers] ACK before INVITE on re-INVITE causes Cisco AS's to return a 500 error

by Evan Borgstrom

Hey all, This has more to do with Asterisk & Cisco AS's IMO but I thought it would be interesting to get some responses from this group before I begin the battle with Cisco to get a bug report underway. We have an Asterisk box acting as a PBX sending calls to our SER instances. The asterisk box has "reinvite=yes" in the sip.conf file which causes Asterisk to send a re-INVITE for the call once it receives the 200 OK. This re-INVITE has it's SDP changed to that of the IP Phone behind the Asterisk box so that all media doesn't need to traverse it. This works fine except in certain instances where messages get out of order and it causes our Cisco GW's to error the call with a generic 500 message. When the messages are in the correct order the 500 error does not appear. So here's the situation, user behind the asterisk box picks up the phone and dials a PSTN destination. Asterisk fires off the initial invite and everything proceeds as normal and we eventually receive the 200 OK message to which the Asterisk box replies with ACK and a new INVITE that come in this order (X = Asterisk, Y = SER, Z = Cisco): U 2005/11/15 16:51:51.557322 X.X.X.X:5060 -> Y.Y.Y.Y:5060 ACK # U 2005/11/15 16:51:51.557948 X.X.X.X:5060 -> Y.Y.Y.Y:5060 INVITE SER accordingly processes the two messages and replies 100 Trying to the new INVITE message. Here's where the problem occurs, SER then sends these messages out to the PSTN gateway but the order has been reversed: # U 2005/11/15 16:51:51.558937 Y.Y.Y.Y:5060 -> X.X.X.X:5060 SIP/2.0 100 trying # U 2005/11/15 16:51:51.559000 Y.Y.Y.Y:5060 -> Z.Z.Z.Z:5060 INVITE # U 2005/11/15 16:51:51.559657 Y.Y.Y.Y:5060 -> X.X.X.X:5060 ACK When this happens the Cisco GW replies with a 500 error message and the following is sent to the Cisco's logs: # U 2005/11/15 16:51:51.711355 Z.Z.Z.Z:5060 -> Y.Y.Y.Y:5060 SIP/2.0 500 Server Internal Error. Nov 15 21:51:51.613: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: Y.Y.Y.Y:5060 Nov 15 21:51:51.613: *****CCB found in UAS Request table. ccb=0x653BCD0C Nov 15 21:51:51.613: CCSIP-SPI-CONTROL: act_sentsucc_new_message Nov 15 21:51:51.617: CCSIP-SPI-CONTROL: act_sentsucc_new_message_request Nov 15 21:51:51.617: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE Nov 15 21:51:51.617: sip_stats_status_code Nov 15 21:51:51.617: sipSPICheckRequest: CheckRequest fail on method 102 error code: 2 and status: 500 Is there any parameter that I can set to ensure that message leave SER in the order they enter? Should the Cisco be able to handle this condition (I check a number of different sections of the RFC but couldn't find anything specific for either side)? Thanks, Evan

18 years, 7 months

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sr-users November 2005