sr-users March 2005

sr-users@lists.kamailio.org

250 participants
627 discussions

[Serusers] getting "error: 400; check if you use aliases in SER" when adding a new user with serctl

by Alberto Cruz

Hi list maybe this is an stupid question but I can't find any clue what I'm doing wrong. When I try to add a new user using serctl add I'm getting the following error: error: 400; check if you use aliases in SER What does "error: 400" mean? I have looked it at messages and I'm no receiving any log information. I'm using two aliases at my ser.cfg I'm attaching my ser.cfg How should I handle this error? Regards Alberto Cruz # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # # # ----------- 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) listen=65.208.39.215 port=5060 alias=65.208.39.215 alias=sip.telereunion.com.mx children=4 fifo_mode=0666 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) modparam("domain", "db_mode", 1) modparam("auth_db|usrloc|domain|group", "db_url", "mysql://ser:heslo@localhost/ser") modparam("group", "use_domain", 0) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("mediaproxy", "natping_interval", 60) modparam("registrar", "nat_flag", 2) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { if (method!="ACK") { sl_send_reply("483","Too Many Hops"); }; break; }; if (msg:len >= 2048 ) { if (method!="ACK") { sl_send_reply("513", "Message too big"); }; break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); if (method=="REGISTER") { if (uri==myself || is_from_local()) { # Mark as NAT'ed if (client_nat_test("3")) { setflag(2); force_rport(); fix_contact(); }; if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; } else if (!check_to()) { sl_send_reply("403", "Username!=To not allowed"); break; }; if (!save("location")) { sl_reply_error(); }; } else { append_hf("P-hint: outbound alias\r\n"); sl_send_reply("403", "This domain is not served here"); }; break; }; if (method=="INVITE") { if (!(is_from_local() || uri==myself || is_uri_host_local())) { sl_send_reply("403", "Relaying is forbidden"); break; }; t_on_failure("1"); } else if (method == "BYE" || method == "CANCEL") { end_media_session(); }; # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); # The following lines are added due media proxy if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; # end media session for BYE and CANCEL is done above # before entering the loose route. no need to call it here t_relay(); break; }; if (client_nat_test("3") && !search("^Record-Route:")) { # Mark as NAT'ed force_rport(); fix_contact(); }; ### Begin PSTN evaluation if (method=="INVITE") { if (uri=~"sip:01[1-9][0-9]+@.*") { if (!is_user_in("From", "ld")) { sl_send_reply("403", "LD permissions needed"); break; }; rewritehostport("65.208.39.219:5060"); } else if (uri=~"sip:00[1-9][0-9]+@.*") { if (!is_user_in("From", "int")) { sl_send_reply("403", "International permissions needed"); break; }; rewritehost("65.208.39.219"); }; t_on_reply("1"); }; ### End PSTN evaluation if (is_uri_host_local() || uri==myself) { # join with next if? # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "User not found"); break; }; }; if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } failure_route[1] { end_media_session(); } onreply_route[1] { if (status=~"(180)|(183)|(2[0-9][0-9])") { if (client_nat_test("1")) { fix_contact(); }; use_media_proxy(); }; }

19 years, 8 months

[Serusers] Why Does SER Keep Retransmitting "487 Request Terminated" Messages?

by Java Rockx

Hi All. I don't know if this problem is related to the re-INVITE problem I posted yesterday http://lists.iptel.org/pipermail/serusers/2005-March/016961.html But it seems that when our Sonus PSTN gateway is invovled in a call where the SIP UA sends a CANCEL before the other party answers, that ser restransmits 487 messages to the SIP UA. When dialing SIP->SIP everything works fine, it's just when the Sonus GW is being accessed that the problem surfaces. It appears that everything has been ACKed, so I'm stumped as to what is happening here. The attached call log shows the complete dialog where my Grandstream UA dialed a PSTN number and cancelled the call when the other side was ringing. Can anyone give me a pointer as to why this might be? Regards, Paul

19 years, 8 months

[Serusers] developing custom module

by Martin van den Berg

Hello, I want to develop a custom module for SER. Where do I find something like an example, introduction or howto-guide? Thanks Martin

19 years, 8 months

Re: [Serusers] Mediaproxy and parallel forking

by Enthu

On Mon, 21 Mar 2005 11:17:04 -0300, Ezequiel Colombo <ecolombo(a)arcotel.net> wrote: > Hi, are you using end_media_session() on receive of non-sucessfull reply ? Yes, I am doing ... but there wasnt any non-successfull reply in the scenario mentioned. Also, mediaproxy does terminate the session based on Call-ID (correct me if I am wrong) and it terminates the session completely, ignoring branches.

19 years, 8 months

[Serusers] Registrar server helpppppppppppppp

by karim Oumalou

Hi everybody I having a weird behavior with the registrar module i have my ser.cfg ok, and i want to add the location and registration server. Please give me some comments for this problem. Excuse my english thanks

19 years, 8 months

[Serusers] To and From Headers - Angle Brackets Or Not

by Java Rockx

Hi All. I'm using ser-0.9.1. Does anyone know if angle brackets are required around the TO: and FROM: headers? Our Sonus gateway sometimes omits these brackets and I just want to make sure that SER correctly identifies SIP transactions. Here is a PSTN->SIP INVITE that shows the brackets being omitted. U 2005/03/21 10:19:08.966284 216.229.127.60:5060 -> 10.3.0.221:5060 INVITE sip:3212514231@10.3.0.221:5060 SIP/2.0. Via: SIP/2.0/UDP 216.229.127.60:5060;branch=z9hG4bK619440ef48f-34104637. Via: SIP/2.0/UDP 66.243.109.99:5060;branch=4ad748fe5b9d60cbf6a53c0614a3da05. To: 3212514231 <sip:3212514231@66.243.109.99:5060>. From: sip:4073192279@66.236.245.98;tag=67DD67C-D31. Call-ID: 6997024-3320407223-278012(a)66.243.109.99. CSeq: 1 INVITE. Max-Forwards: 4. Contact: sip:4073192279@66.243.109.99:5060. Record-Route: <sip:216.229.127.60:5060;lr>. Content-Type: application/sdp. Content-Length: 313. . v=0. o=NexTone-MSW 1234 3665 IN IP4 66.236.245.98. s=sip call. c=IN IP4 66.236.245.98. t=0 0. m=audio 17694 RTP/AVP 18 0 4 8 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=fmtp:4 annexa=yes. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. Regards, Paul

19 years, 8 months

RE: [Serusers] Getting Started - Issue 02 covers MySQL Support

by Didier Hung Wan Luk

Hi Simon, Is there a plan to include how to make Windows Messenger work with ser? How to make WM use peer-to-peer presence with ser? Kind Regards, DH. ________________________________ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Simon Miles Sent: Monday, March 21, 2005 3:55 PM To: serusers(a)lists.iptel.org Subject: [Serusers] Getting Started - Issue 02 covers MySQL Support Dear All, Paul, Greger and myself would like you to know that the next issue of 'Getting Started' is now available on the www.ONsip.org web site. This new version builds upon the first issue by adding MySQL support. We are already finalising the next issue covering NAT support - a subject that always fills up this mailing list with questions. We are doing this for the community and would welcome any feedback on the document to make it more readable etc. Comments are welcome on the www.ONsip.org <http://www.onsip.org/> web site, goto Forums and select the Getting Started Feedback area. To download the document, go to the download section. We hope this is of benefit to the Community. Thanks Paul, Simon, Greger This mail has been scanned for known virus like activity on behalf of GALANA DISTRIBUTION PETROLIERE S.A. This E-mail is privileged, confidential and intended solely for the addressee(s).Any unauthorised dissemination or copying of this E-mail, and any use or disclosure of any information contained in it, is strictly prohibited and may be illegal. If you have received this E-mail in error please notify the sender immediately and delete it from your system. GALANA DISTRIBUTION PETROLIERE S.A. does not accept any liability for any errors, omissions, interceptions, corrupted mail, lost communications or late delivery which may arise as a result of this E-mail or for any virus that may be contained in it. Any views or opinions expressed in this E-mail are solely those of the author and do not necessarily reflect those of GALANA DISTRIBUTION PETROLIERE S.A. Ce courriel a fait l'objet, par GALANA DISTRIBUTION PETROLIERE S.A. de verifications le protegeant des virus connus. Ce courriel est personnel, confidentiel et adresse exclusivement a son ou ses destinataire(s). Toute diffusion ou copie non autorisee ainsi que l'usage partiel ou total de son contenu est strictement interdit et peut-etre illegal. Si vous avez recu ce courriel par erreur, merci d'en informer son emetteur et de l'effacer de votre ordinateur. GALANA DISTRIBUTION PETROLIERE S.A. refuse toutes responsabilites liees aux erreurs, omissions, interceptions, deformations, communications perdues, ou retards de transmissions dont ce courriel pourrait etre l'objet ou a l'origine ainsi que des virus qu'il pourrait contenir. Les avis ou opinions exprimes dans ce courriel refletent exclusivement ceux de son auteur et non ceux de GALANA DISTRIBUTION PETROLIERE S.A.

19 years, 8 months

[Serusers] SER 0.8.14 + MEDIAPROXY: 2 UA BEHIND the same NAT : no sound

by Nicolas Ruiz

HI, I have installed SER 0.8.14 + MEDIAPROXY : when I make a call with 2 UA on different NAT, it's good. But, when the UAs are behind the same NAT, there are no sound. The RTP packets go to the MEDIAPROXY but NO SOUND. Does someone know how to solve this problem??? Best Regards NICOLAS RUIZ Thats my conf : if (method=="REGISTER") { # Mark as NAT'ed if (client_nat_test("3")) { setflag(2); force_rport(); fix_contact(); }; if (!www_authorize("sip.vivaction.net", "subscriber")) { www_challenge("sip.vivaction.net", "0"); break; } else if (!check_to()) { sl_send_reply("403", "Username!=To not allowed"); break; }; if (!save("location")) { sl_reply_error(); }; break; }; if (method == "BYE" || method == "CANCEL") { end_media_session(); }; if (loose_route()) { if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; # end media session for BYE and CANCEL is done above # before entering the loose route. no need to call it here t_relay(); break; }; # Force subsequent messages to pass trough this proxy if (method == "INVITE") { record_route(); }; if (client_nat_test("3") && !search("^Record-Route:")) { # Mark as NAT'ed force_rport(); fix_contact(); }; if (method=="INVITE") { t_on_reply("1"); }; if (is_uri_host_local()) { if (!lookup("location")) { sl_send_reply("404", "User not found"); break; }; }; if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } failure_route[1] { end_media_session(); } onreply_route[1] { if (status=~"(183)|(2[0-9][0-9])") { if (client_nat_test("1")) { fix_contact(); }; use_media_proxy(); }; } Vos Solutions Voix-Data ! Nicolas Ruiz Service Technique Ligne directe : + 33 (0) 1 56 38 39 71 Fax :+ 33 (0) 1 47 24 74 77 nruiz(a)vivaction.com Immeuble Plein Ouest 177 av. Georges Clemenceau 92024 Nanterre - France Tel : 0 811 02 6000 www.vivaction.com ____________________________________________________________________________ ____________________________________________ This e-mail and the information it contains are confidential and legally protected by law. Only access by the intended recipient is authorized. Review, distribution,reproduction, publication or other use of this e-mail is prohibited. Cet e-mail et les informations qu'il contient sont confidentiels et protégés par la loi. L'accès à ce message n'est autorisé qu'au destinataire de celui-ci. Toute modification,distribution, reproduction, publication, ou autre utilisation de cet e-mail est formellement interdite.

19 years, 8 months

[Serusers] Troubles calling from a Nated UA to a PSTN gateway on a Public address space using SER 0.9.1 and Mediaproxy Release 1.2.1

by Alberto Cruz

Hi everybody I having a weird behavior with Mediaproxy I'm using SER 0.9.1 and Mediaproxy Release 1.2.1 The Gateway is a Cisco AS5300 using release 12.2(15)T I have a UA using behind a NAT, the PSTN gateway and the SER server both are using public IP addresses. The behaviors are the following: 1. When I tried to place a call from a NATed UA to the PSTN gateway I'm not receiving the call progress tone or ring back tone at the NATed UA. 2. If I decide to wait until the call is completed (doesn't matter if a heard death air as progress tone) I start hearing the calling party as soon the call is completed and we can talk each other without any troubles. 3. If I cancel the call and hang up the phone at the NATed UA before the call is completed (during the call progress stage) the SER/Mediaproxy don't cancel the call it still in progress until it is completed or cancelled by the PSTN. I'm attaching my ser configuration and the logging I'm getting when I cancel the call before it's completed. Please Help me to know how to fix this If I'm making some mistake with mediaproxy configuration. Thanks in advanced. Regards Alberto Cruz Mar 20 00:10:02 matrix proxydispatcher[1379]: command request call-80BF3413-4297-D911-0217-21(a)172.31.254.240 172.31.254.240:10286:audio 200.67.33.247 172.31.254.240 remote 65.208.39.219 remote Quintum/1.0.0 info=from:8412@172.31.254.240,to:018183324166@65.208.39.215,fromtag:ac1ffef0-20,totag: Mar 20 00:10:02 matrix proxydispatcher[1379]: will use default mediaproxy for this call. Mar 20 00:10:02 matrix mediaproxy[1376]: command request call-80BF3413-4297-D911-0217-21(a)172.31.254.240 172.31.254.240:10286:audio 200.67.33.247 172.31.254.240 remote 65.208.39.219 remote Quintum/1.0.0 info=from:8412@172.31.254.240,to:018183324166@65.208.39.215,fromtag:ac1ffef0-20,totag:,dispatcher Mar 20 00:10:02 matrix mediaproxy[1376]: session call-80BF3413-4297-D911-0217-21(a)172.31.254.240: started. listening on 65.208.39.215:35150 Mar 20 00:10:02 matrix mediaproxy[1376]: command execution time: 9.23 ms Mar 20 00:10:02 matrix proxydispatcher[1379]: forwarding to mediaproxy on /var/run/mediaproxy.sock: got: '65.208.39.215 35150' Mar 20 00:10:02 matrix proxydispatcher[1379]: command execution time: 13.13 ms Mar 20 00:10:06 matrix proxydispatcher[1379]: command lookup call-80BF3413-4297-D911-0217-21(a)172.31.254.240 65.208.39.219:18342:audio 65.208.39.219 172.31.254.240 remote 65.208.39.215 unknown Cisco-SIPGateway/IOS-12.x info=from:8412@172.31.254.240,to:018183324166@65.208.39.215,fromtag:ac1ffef0-20,totag:711504-73 Mar 20 00:10:06 matrix mediaproxy[1376]: command lookup call-80BF3413-4297-D911-0217-21(a)172.31.254.240 65.208.39.219:18342:audio 65.208.39.219 172.31.254.240 remote 65.208.39.215 unknown Cisco-SIPGateway/IOS-12.x info=from:8412@172.31.254.240,to:018183324166@65.208.39.215,fromtag:ac1ffef0-20,totag:711504-73,dispatcher Mar 20 00:10:06 matrix mediaproxy[1376]: command execution time: 1.63 ms Mar 20 00:10:06 matrix proxydispatcher[1379]: forwarding to mediaproxy on /var/run/mediaproxy.sock: got: '65.208.39.215 35150' Mar 20 00:10:06 matrix proxydispatcher[1379]: command execution time: 4.88 ms Mar 20 00:10:06 matrix mediaproxy[1376]: session call-80BF3413-4297-D911-0217-21(a)172.31.254.240: caller signed in from 200.67.33.247:49125 (RTP) (will return to 200.67.33.247:49125) Mar 20 00:10:06 matrix mediaproxy[1376]: session call-80BF3413-4297-D911-0217-21(a)172.31.254.240: called signed in from 65.208.39.219:18342 (RTP) (will return to 65.208.39.219:18342) Mar 20 00:10:08 matrix mediaproxy[1376]: session call-80BF3413-4297-D911-0217-21(a)172.31.254.240: caller signed in from 200.67.33.247:49126 (RTCP) (will return to 200.67.33.247:49126) Mar 20 00:10:11 matrix mediaproxy[1376]: session call-80BF3413-4297-D911-0217-21(a)172.31.254.240: called signed in from 65.208.39.219:18343 (RTCP) (will return to 65.208.39.219:18343) Mar 20 00:10:13 matrix proxydispatcher[1379]: command delete call-80BF3413-4297-D911-0217-21(a)172.31.254.240 info= Mar 20 00:10:13 matrix mediaproxy[1376]: command delete call-80BF3413-4297-D911-0217-21(a)172.31.254.240 info=dispatcher Mar 20 00:10:13 matrix mediaproxy[1376]: session call-80BF3413-4297-D911-0217-21(a)172.31.254.240: 29/44/73 packets, 1380/5320/6700 bytes (caller/called/relayed) Mar 20 00:10:13 matrix mediaproxy[1376]: session call-80BF3413-4297-D911-0217-21(a)172.31.254.240: ended. Mar 20 00:10:13 matrix mediaproxy[1376]: command execution time: 1.43 ms Mar 20 00:10:13 matrix proxydispatcher[1379]: forwarding to mediaproxy on /var/run/mediaproxy.sock: got: '' Mar 20 00:10:13 matrix proxydispatcher[1379]: command execution time: 4.29 ms Mar 20 00:10:45 matrix proxydispatcher[1379]: command lookup call-80BF3413-4297-D911-0217-21(a)172.31.254.240 65.208.39.219:18342:audio 65.208.39.219 172.31.254.240 remote 65.208.39.215 unknown Cisco-SIPGateway/IOS-12.x info=from:8412@172.31.254.240,to:018183324166@65.208.39.215,fromtag:ac1ffef0-20,totag:711504-73 Mar 20 00:10:45 matrix proxydispatcher[1379]: warning: trying to lookup session with non-existent id: 'call-80BF3413-4297-D911-0217-21(a)172.31.254.240' Mar 20 00:10:45 matrix proxydispatcher[1379]: command execution time: 0.91 ms Mar 20 00:10:45 matrix /usr/local/sbin/ser[11723]: error: use_media_proxy(): empty response from mediaproxy Mar 20 00:10:45 matrix /usr/local/sbin/ser[11723]: ERROR: on_reply processing failed Mar 20 00:10:46 matrix proxydispatcher[1379]: command lookup call-80BF3413-4297-D911-0217-21(a)172.31.254.240 65.208.39.219:18342:audio 65.208.39.219 172.31.254.240 remote 65.208.39.215 unknown Cisco-SIPGateway/IOS-12.x info=from:8412@172.31.254.240,to:018183324166@65.208.39.215,fromtag:ac1ffef0-20,totag:711504-73 Mar 20 00:10:46 matrix proxydispatcher[1379]: warning: trying to lookup session with non-existent id: 'call-80BF3413-4297-D911-0217-21(a)172.31.254.240' Mar 20 00:10:46 matrix proxydispatcher[1379]: command execution time: 1.08 ms Mar 20 00:10:46 matrix /usr/local/sbin/ser[11725]: error: use_media_proxy(): empty response from mediaproxy Mar 20 00:10:46 matrix /usr/local/sbin/ser[11725]: ERROR: on_reply processing failed Mar 20 00:10:47 matrix proxydispatcher[1379]: command lookup call-80BF3413-4297-D911-0217-21(a)172.31.254.240 65.208.39.219:18342:audio 65.208.39.219 172.31.254.240 remote 65.208.39.215 unknown Cisco-SIPGateway/IOS-12.x info=from:8412@172.31.254.240,to:018183324166@65.208.39.215,fromtag:ac1ffef0-20,totag:711504-73 Mar 20 00:10:47 matrix proxydispatcher[1379]: warning: trying to lookup session with non-existent id: 'call-80BF3413-4297-D911-0217-21(a)172.31.254.240' Mar 20 00:10:47 matrix proxydispatcher[1379]: command execution time: 1.09 ms Mar 20 00:10:47 matrix /usr/local/sbin/ser[11723]: error: use_media_proxy(): empty response from mediaproxy Mar 20 00:10:47 matrix /usr/local/sbin/ser[11723]: ERROR: on_reply processing failed # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # # # ----------- 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) listen=65.208.39.215 port=5060 alias=65.208.39.215 alias=sip.telereunion.com.mx children=4 fifo_mode=0666 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) modparam("domain", "db_mode", 1) modparam("auth_db|usrloc|domain|group", "db_url", "mysql://ser:heslo@localhost/ser") modparam("group", "use_domain", 0) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("mediaproxy", "natping_interval", 60) modparam("registrar", "nat_flag", 2) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { if (method!="ACK") { sl_send_reply("483","Too Many Hops"); }; break; }; if (msg:len >= 2048 ) { if (method!="ACK") { sl_send_reply("513", "Message too big"); }; break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); if (method=="REGISTER") { if (uri==myself || is_from_local()) { # Mark as NAT'ed if (client_nat_test("3")) { setflag(2); force_rport(); fix_contact(); }; if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; } else if (!check_to()) { sl_send_reply("403", "Username!=To not allowed"); break; }; if (!save("location")) { sl_reply_error(); }; } else { append_hf("P-hint: outbound alias\r\n"); sl_send_reply("403", "This domain is not served here"); }; break; }; if (method=="INVITE") { if (!(is_from_local() || uri==myself || is_uri_host_local())) { sl_send_reply("403", "Relaying is forbidden"); break; }; t_on_failure("1"); } else if (method == "BYE" || method == "CANCEL") { end_media_session(); }; # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); # The following lines are added due media proxy if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; # end media session for BYE and CANCEL is done above # before entering the loose route. no need to call it here t_relay(); break; }; if (client_nat_test("3") && !search("^Record-Route:")) { # Mark as NAT'ed force_rport(); fix_contact(); }; ### Begin PSTN evaluation if (method=="INVITE") { if (uri=~"sip:01[1-9][0-9]+@.*") { if (!is_user_in("From", "ld")) { sl_send_reply("403", "LD permissions needed"); break; }; rewritehostport("65.208.39.219:5060"); } else if (uri=~"sip:00[1-9][0-9]+@.*") { if (!is_user_in("From", "int")) { sl_send_reply("403", "International permissions needed"); break; }; rewritehost("65.208.39.219"); }; t_on_reply("1"); }; ### End PSTN evaluation if (is_uri_host_local() || uri==myself) { # join with next if? # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "User not found"); break; }; }; if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } failure_route[1] { end_media_session(); } onreply_route[1] { if (status=~"(180)|(183)|(2[0-9][0-9])") { if (client_nat_test("1")) { fix_contact(); }; use_media_proxy(); }; }

19 years, 8 months

[Serusers] SER Radius Accounting - Two Start Messages for Every Call

by matt morris

Hello List, I got a problem with ser radius accounting. Seems ser sends two start messages to my freeradius server (ser 0.10.99-dev5, radiusclient-ng 0.5.0 and freeradius snapshot 20050315 in the same computer) for every calls I made, and also for missed calls (two same failed message). Does it have to do with my dictionary file(s)? For my ser.cfg, I only set the radius_flag to be 1 and radius_missed_flag to 2. I only called setflag(1) once with this if condition before record_route(): if (method=="INVITE" || method=="BYE") { setflag(1); }; Did not have this problem when I used db_flag There are another two users with this same or similar problem, but there is no answers to their thread. So wondering if they have solved it or anyone else know why is it behaving like this? Please help me out, thank you. My radacct log: Sun Mar 20 20:40:31 2005 Acct-Status-Type = Start Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = 1 User-Name = "xxxxxx(a)xxxxxx.xxxxxx.xxx" Calling-Station-Id = "sip:xxxxxx@xxxxxx.xxxxxx.xxx:xxxx" Called-Station-Id = "sip:yyyyyy@xxxxxx.xxxxxx.xxx" Sip-Translated-Request-URI = "sip:yyyyyy@xxxxxx.xxxxxx.xxx:xxxxx" Acct-Session-Id = "58FCBA73-03FD-4AB4-A4CC-6C0A890414F3(a)xxx.xxx.xxx.xxx" Sip-To-Tag = "3607001048" Sip-From-Tag = "577868341" Sip-Cseq = "18570" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 Acct-Delay-Time = 0 Client-IP-Address = 127.0.0.1 Acct-Unique-Session-Id = "2e516ab9838c2cbd" Stripped-User-Name = "xxxxxx" Realm = "DEFAULT" Timestamp = 1111369231 Sun Mar 20 20:40:31 2005 Acct-Status-Type = Start Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = 4 User-Name = "xxxxxx(a)xxxxxx.xxxxxx.xxx" Calling-Station-Id = "sip:xxxxxx@xxxxxx.xxxxxx.xxx:xxxx" Called-Station-Id = "sip:yyyyyy@xxxxxx.xxxxxx.xxx" Sip-Translated-Request-URI = "sip:yyyyyy@xxxxxx.xxxxxx.xxx:xxxxx" Acct-Session-Id = "58FCBA73-03FD-4AB4-A4CC-6C0A890414F3(a)xxx.xxx.xxx.xxx" Sip-To-Tag = "3607001048" Sip-From-Tag = "577868341" Sip-Cseq = "18570" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 Acct-Delay-Time = 0 Client-IP-Address = 127.0.0.1 Acct-Unique-Session-Id = "2e516ab9838c2cbd" Stripped-User-Name = "xxxxxx" Realm = "DEFAULT" Timestamp = 1111369231 Sun Mar 20 20:41:05 2005 Acct-Status-Type = Stop Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = 8 User-Name = "xxxxxx(a)xxxxxx.xxxxxx.xxx" Calling-Station-Id = "sip:yyyyyy@xxxxxx.xxxxxx.xxx:xxxx" Called-Station-Id = "sip:xxxxxx@xxxxxx.xxxxxx.xxx" Sip-Translated-Request-URI = "sip:xxxxxx@xxxxxx.xxxxxx.xxx:xxxxx" Acct-Session-Id = "58FCBA73-03FD-4AB4-A4CC-6C0A890414F3(a)xxx.xxx.xxx.xxx" Sip-To-Tag = "577868341" Sip-From-Tag = "3607001048" Sip-Cseq = "2254" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 Acct-Delay-Time = 0 Client-IP-Address = 127.0.0.1 Acct-Unique-Session-Id = "7b53eafcefa8c57e" Stripped-User-Name = "yyyyyy" Realm = "DEFAULT" Timestamp = 1111369265 The same for missed calls too: Sun Mar 20 17:51:53 2005 Acct-Status-Type = Failed Service-Type = Sip-Session Sip-Response-Code = 408 Sip-Method = 1 User-Name = "xxxxxx(a)xxxxxx.xxxxxx.xxx" Calling-Station-Id = "sip:xxxxxx@xxxxxx.xxxxxx.xxx:xxxx" Called-Station-Id = "sip:yyyyyy@xxxxxx.xxxxxx.xxx" Sip-Translated-Request-URI = "sip:yyyyyy@xxxxxx.xxxxxx.xxx:xxxxx" Acct-Session-Id = "60A406F8-6CA2-4251-8CD6-796A69F82A3A(a)xxx.xxx.xxx.xxx" Sip-To-Tag = "n/a" Sip-From-Tag = "3544681223" Sip-Cseq = "4658" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 Acct-Delay-Time = 0 Client-IP-Address = 127.0.0.1 Acct-Unique-Session-Id = "5059d3ad9e266ea1" Stripped-User-Name = "xxxxxx" Realm = "DEFAULT" Timestamp = 1111359113 Sun Mar 20 17:51:53 2005 Acct-Status-Type = Failed Service-Type = Sip-Session Sip-Response-Code = 408 Sip-Method = 1 User-Name = "xxxxxx(a)xxxxxx.xxxxxx.xxx" Calling-Station-Id = "sip:xxxxxx@xxxxxx.xxxxxx.xxx:xxxx" Called-Station-Id = "sip:yyyyyy@xxxxxx.xxxxxx.xxx" Sip-Translated-Request-URI = "sip:yyyyyy@xxxxxx.xxxxxx.xxx:xxxxx" Acct-Session-Id = "60A406F8-6CA2-4251-8CD6-796A69F82A3A(a)xxx.xxx.xxx.xxx" Sip-To-Tag = "n/a" Sip-From-Tag = "3544681223" Sip-Cseq = "4658" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 Acct-Delay-Time = 0 Client-IP-Address = 127.0.0.1 Acct-Unique-Session-Id = "5059d3ad9e266ea1" Stripped-User-Name = "xxxxxx" Realm = "DEFAULT" Timestamp = 1111359113 _________________________________________________________________ Scan and help eliminate destructive viruses from your inbound and outbound e-mail and attachments. http://join.msn.com/?pgmarket=en-ca&page=byoa/prem&xAPID=1994&DI=1034&SU=ht… Start enjoying all the benefits of MSN® Premium right now and get the first two months FREE*.

19 years, 8 months

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sr-users March 2005