sr-users August 2006

sr-users@lists.kamailio.org

184 participants
360 discussions

by Sebastian Gabris

Hi everybody, does somebody know if the iptelorg GmbH is still existent.Visited their homepage www.iptelorg.com and it look pretty unmaintained to me. Is there any other company in Germany who offers support & installation services for SER? Best regards, Sebastian P.S. Congrats to iptel.org for their new homepage, will look closer at it tommorow :-) > -----Ursprüngliche Nachricht----- > Von: Sebastian Gabris <Sebastian.Gabris(a)web.de> > Gesendet: 11.08.06 16:21:36 > An: ratif(a)burraqtel.com.p, serusers(a)iptel.org > Betreff: [Serusers] ser SDP rewrite problem, rtpproxy is not working !!! > Hi, > > got also this problem, did you find a solution? > > Sebastian > > hello there, > > I am working on SER with two network interfaces. > > EP1--------------->(172.16.16.160) SER (192.168.5.1) -------------> EP2 > > what happens, SER receives invite from EP1, it reinitiates invite for > EP2. now what it does is that it writes the external IP i.e. > 172.16.16.160 in the SDP contact info, where as it should be the > internal one i.e. 192.168.5.1 ... > now I want to overwrite the SDP contact info in that invite. I have got > two options to do it one option is to use force_rtp_proxy("ei", > "192.168.5.1") or use sdp_mangle_ip() function of mangler.so . but they > are not working for me. when I play around with force_rtp_proxy function > rtpproxy get crashed and sdp_mangle_ip() is not behaving properly ...... > whats your experience ? > > I have used ser-0.8.14 and now using ser-0.9.0 with cvs rtpproxy. I am > running rtpproxy like > rtpproxy -s udp:192.168.5.1 -l 172.16.16.160 > and I am using this line in ser.cfg > modparam("nathelper", "rtpproxy_sock", "udp:192.168.5.1") > > I am really stucked please help !!!! > > thanking you > > regards, > -- > Atif > ______________________________________________________________ > Verschicken Sie romantische, coole und witzige Bilder per SMS! > Jetzt bei WEB.DE FreeMail: http://f.web.de/?mc=021193 > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers ______________________________________________________________ Verschicken Sie romantische, coole und witzige Bilder per SMS! Jetzt bei WEB.DE FreeMail: http://f.web.de/?mc=021193

18 years, 2 months

[Serusers] About the alias call

by AR.

Dear All I have a problem about the alias call When I use my PC and hardphone (cisco7960) , I can call someone by alias name But, It can't if I use the NB or use other PCs, Could anyone help me? thanks AR.

18 years, 2 months

Re: [Serusers] IMPORANT ANNOUNCEMENT: Launch of new iptel.org site!!

by Sebastian Gabris

Hi all, I cannot register as new user. The reply email from iptel has a different username but my password(from the registration formular) in it. Tried it two times, and got two different usernames as reply. Also if I tried to login with my email-address instead of username but it didnt work. best regards, Sebastian > -----Ursprüngliche Nachricht----- > Von: "Greger V. Teigre" <greger(a)teigre.com> > Gesendet: 16.08.06 14:59:10 > An: sip <sip(a)arcdiv.com> > CC: serdev(a)iptel.org, serusers(a)iptel.org, serusers-bounces(a)lists.iptel.org > Betreff: Re: [Serusers] IMPORANT ANNOUNCEMENT: Launch of new iptel.org site!! sip wrote: > I actually just happened to hit the main site yesterday and noticed the > change. Much glitzier looking, although it's going to take me a little time to > get used to the new layout. > ;-) at least there is no flash... > One thing I didn't notice on there (perhaps because I just missed it) was a > link to the CVS repository to browse. Not a huge deal, it just used to be nice > to browse the READMEs for the modules (since the documentation hasn't all been > in once place). > Good point! Feel free to add it! > Maybe here?! > http://www.iptel.org/doc > > There is a need to do something about the module documentation. The > xml-based based documentation is a bit hard to maintain and I don't > think anybody has really gotton used to it. One suggestion is to move > the authoritative module documention to the new iptel.org site and keep > them maintained there. It will then be easier to maintain, more easily > accessible, and we can keep documentation for different versions easily > accessible. > > I'm looking forward to seeing the future of SER. > Me too! > g-) > > > N. > On Wed, 16 Aug 2006 09:50:09 +0200, Greger V. Teigre wrote > > > Dear all! > We are doing this the soft way: Launching when everybody is on vacation :-D > We have for some time now been working on putting up a new community-based framework for the iptel.org open source projects. Our aim is to step-wise address lack of documentation, roadmap, > release plans, transparent management and so on. > The new http://iptel.org/ site contains quite a lot of content. There is still plenty to migrate from the old iptel.org site, > ONsip.org, as well as better organize documentation around SER development and releases. You can help out with that by adding a small FAQ, how-to, copy something from ONsip.org that you found particularly helpful and so on! > We have created some basic content that we hope you will help comment and develop further. Here are some pointers to content on the new site: > ======================================== > About the new site and how to contribute content > http://www.iptel.org/iptelorgsite/intro > How to get started with SER? > http://www.iptel.org/ser/using > Overview of documentation > http://www.iptel.org/doc > The old ONsip.org Getting Started document > http://www.iptel.org/ser/doc/gettingstarted > Contributed modules and patches > http://www.iptel.org/ser/contributions > A beginning skeleton for a roadmap > http://www.iptel.org/ser/roadmap > A wishlist for future versions > http://www.iptel.org/ser/wishlist > Release policies > http://www.iptel.org/ser/development/release_policies > ====================================================== > Below is the launch announcement. > Best regards, > Greger Teigre > g-) > --------------------------------------------------------- > The new http://iptel.org/, home of SER, SEMS, and SERWeb is now up. If you have an account from ONsip.org, you can log in using your old ONsip.org username/password. However, all content is available without registration. > The new site is supported by the joint efforts of the iptel.org and ONsip.org teams. Our aim is to dramatically improve the project communication, documentation, and development process. The start is already up there, and we have many plans. The best thing of all: The new http://iptel.org site is a great basis for contributions from anyone. > We are working on migrating and organizing the remaining content from the old site, but meanwhile you can access everything from the old site from http://dog.iptel.org/ > We look forward to a revitalised SER open source project! > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > ----------------------------------------------------------------- > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > ______________________________________________________________ Verschicken Sie romantische, coole und witzige Bilder per SMS! Jetzt bei WEB.DE FreeMail: http://f.web.de/?mc=021193

18 years, 2 months

[Serusers] Can ser listen on two or more then two ports?

by Charles Wang

Hi, ALL: Can ser listen on 2 or more than 2 ports on the same IP of one server? -- Best Regards Charles

18 years, 2 months

[Users] SIP general question about via header

by Ricardo Martinez

Hello Is permitted to mix the two via forms (normal form and compact form) in one SIP message? For example i have an INVITE with this VIA's : INVITE sip:005623341876@10.0.0.67:5060 SIP/2.0 Record-Route: <sip:10.0.0.36;ftag=2179110362;lr=on> Record-Route: <sip:10.0.0.35;ftag=2179110362;lr=on> Via: SIP/2.0/UDP 10.0.0.36;branch=z9hG4bK58f4.cdc602d2.0 Via: SIP/2.0/UDP 10.0.0.35;branch=z9hG4bK58f4.83fe5745.0 v: SIP/2.0/UDP 64.76.148.234:5060;branch=z9hG4bK2881566366 .... If the answer is yes where can i found some documents to support this information? Thanks again! Best Regards, Ricardo Martinez.-

18 years, 2 months

[Users] sip:+5555;npdi=yes@domain.com issue with enum_query()

by Greg Fausak

Howdy! How can I get rid of the tel: number crap on the end of a RURI like the one in the header? When I do my enum lookup it is failing, because it is looking up +5555;npdi=yes when I want it to look up +5555 Any advice from the wise would be greatly appreciated! Thanks, ---greg -- Greg Fausak greg(a)thursday.com

18 years, 2 months

[Users] Need help to migrate OpenSer v1.0.x to v1.1.x

by Rafo RGV

Hi I am migrating OpenSER v1.0.9 to v1.1.0 using the config file I send below that works perfect in 1.0.9 and I modified to run in 1.1.x , I´ve just needed to change naming schema in parameter given to exported functions of avpops module to run without errors in this new version but when running can´t see basic logs and can´t register any client even if I install DB schema and create subscriber using its openserctl , ngrep just show me a 'forbidden' message but dont know why, please someone send me some advice. thanks rafael # ----------------- global configuration parameters -------------------- debug=4 fork=yes log_stderror=yes listen=192.168.2.136 # OPENSER IP ADDRESS HERE port=5060 # Hostname matching an alias will satisfy the condition uri==myself". # alias=mydomain.com.pe:5060 # alias=mydomain.com.pe alias=192.168.2.136:5060 alias=192.168.2.136 check_via=yes # (cmd. line: -v) children=4 dns=no rev_dns=no fifo="/tmp/openser_fifo" # fifo="FIFO" fifo_mode=0666 # Fifo permissions can be changes from here. fifo_db_url="mysql://dbuser:dbpass@dbt2.mydomain.com.pe/openser" # ----------------------------------------------------------------------- loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/group.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" ## loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/uac_redirect.so" loadmodule "/usr/local/lib/openser/modules/diversion.so" # ----------------------------------------------------------------------- modparam("auth_db|permissions|group|uri_db|usrloc", "db_url", "mysql://dbuser:dbpass@dbt2.mydomain.com.pe/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("avpops", "avp_url", "mysql://dbuser:dbpass@dbt2.mydomain.com.pe/openser") modparam("avpops", "avp_table", "usr_preferences") # ------------- Mysql Accounting parameters modparam("acc", "log_flag", 1) modparam("acc", "log_level", 2) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 3) modparam("acc", "log_missed_flag", 3) modparam("acc", "db_url", "mysql://dbuser:dbpass@dbt2.mydomain.com.pe/openser") modparam("acc", "report_ack", 0) # 1 reporta dos starts en acc (para INVITE y ACK) modparam("acc", "log_fmt", "miocfsputdr") modparam("acc", "detect_direction", 1) # only in 1.1.x or cvs modparam("acc", "multi_leg_enabled", 1) modparam("acc", "src_leg_avp_id", 110) modparam("acc", "dst_leg_avp_id", 111) # ------------- Domain parameters modparam("usrloc", "use_domain", 1) modparam("auth_db", "use_domain", 1) modparam("registrar", "use_domain", 1) modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") # ------------------------------------------------------------------------- route { # log(1, "-------------------------------------------------------\n"); # log(1, "entering main loop\n"); ## populate the avps (for Multi call leg accounting): if (!method=="REGISTER" && !method=="NOTIFY") { avp_write("$from/username", "$avp(i:110)"); avp_write("$ruri/username", "$avp(i:111)"); }; # set flag for Missed calls for Accounting: if (!method=="OPTIONS") setflag(3); if (method=="INVITE") { log(1, " INVITE MESSAGE RECEIVED - START ACC\n"); }; if (method=="BYE") { log (1, " BYE MESSAGE RECEIVED - STOP ACCOUNTING\n"); }; if (method=="CANCEL") { log (1, " CANCEL MESSAGE RECEIVED - STOP ACCOUNTING\n"); }; if (method=="OPTIONS") { log(1, " OPTIONS MESSAGE RECEIVED\n"); return; #Do Nothing for 'Options' }; # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); return; }; # ----------------------------------------------------------------- # Handle NOTIFY requests from Sipura's to keep-alive NAT # ----------------------------------------------------------------- if ((is_method("NOTIFY")) && (! uri=~"sip:.*[@]+.*")) { sl_send_reply("200", "OK keep-alive"); log(1, " OK keep-alive\n"); return; }; # ----------------------------------------------------------------- # Only allow known sip messages # ----------------------------------------------------------------- if (!is_method("INVITE|REGISTER|BYE|NOTIFY|ACK|CANCEL|REFER|SUBSCRIBE")) { sl_send_reply("501", "Not implemented here"); log(1, " 501 - Not implemented here\n"); return; }; # ----------------------------------------------------------------- # SUBSCRIBE cleanup # ----------------------------------------------------------------- if (is_method("SUBSCRIBE")) { replace(" talk", "talk"); }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && nat_uac_test("19")) { # OPENSER IP ADDRESS OR DOMAIN HERE: record_route_preset("192.168.2.136:5060;nat=yes"); setflag(1); ## Set Flag 1 for Acc: test from this location!! } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="REFER" || method=="CANCEL") { unforce_rtp_proxy(); setflag(1); ### Set Flag 1 for Acounting }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(7); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbidden"); }; return; }; if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { setflag(1); # Set Flag for Acounting route(3); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (!lookup("location") && !method=="ACK") { if (does_uri_exist()) { log(1, " uri_exist but its Temporarily Unavailable 1 \n"); route(7); # To Voicemail return; } else { log(1, " unable to locate user - User Not Exist 1 \n"); sl_send_reply("404", "User Not Exist"); acc_db_request("404 User Not Exist", "missed_calls"); return; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (is_user_in("from", "desactivado")) { sl_send_reply("402", "desactivada temporalmente"); log(1," Cta. desactivada temporalmente \n"); return; }; if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { log(1," Fails to Register \n"); www_challenge("","0"); return; }; if (!check_to()) { log(1," Unauthorized registration attempt \n"); sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); log(1," Registered! \n"); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- log(1," route[3]---> INVITE Message Handler \n"); if ( allow_trusted() ) log(1, " Call from pstn or *pbx, no authentication is required. \n"); if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); log(1, " Sorry, only registered users are allowed \n"); return; }; consume_credentials(); }; append_rpid_hf(); # some digits rules here.... lookup("aliases"); if (!is_uri_host_local()) { route(4); route(1); return; }; if (nat_uac_test("19")) { setflag(7); force_rport(); }; # ----------------------------------------------------------------- # to PSTN Routes!!! # ----------------------------------------------------------------- route(6); # To PSTN & LDI Routes block!!! # ----------------------------------------------------------------- if ( is_uri_host_local() ) { # (Is he in the voicemail group?) -- determine it now and store it in # flag 4, before we rewrite the flag using UsrLoc: if (is_user_in("Request-URI", "voicemail")) { log(1, " requested user is in voicemail group \n"); setflag(4); }; # if user is on-line and is in Voicemail group, enable redirection # Pendiente: # ver como meter voicemail al avp junto a las otras funciones... if (method == "INVITE" && isflagset(4)) { log(1, " invite for voicemail user->initiate failureroute[1]\n"); t_on_failure("1"); }; # flag 22 determines if we need to call the append_branch() for Blind CallFwd. if (avp_db_load("$ruri/username", "$avp(s:callfwd)")) { setflag(22); avp_pushto("$ruri", "$avp(s:callfwd)"); route(6); return; }; # native SIP destinations are handled using our USRLOC DB: if (!lookup("location") ) { if (does_uri_exist()) { log(1," uri_exist but its Temporarily Unavailable 2 \n"); route(7); # To Voicemail return; } else { log(1," Unable to locate xcompany user - User Not Exist 2X \n"); prefix("iu"); ## in asterisk: iu: exten => _iu1780XXXX,1,Playback(invalid,noanswer) acc_db_request("404 User Invalid", "missed_calls"); rewritehostport("192.168.2.131:5070"); t_relay(); return; }; }; # t_on_failure() informs SER that we want to perform special handling when a failure condition occurs. # Failure conditions in this context refer to 4xx and 5xx response codes. # By setting t_on_failure(1) before calling t_relay(), oSER will pass control to the failure_route[1] t_on_failure("1"); route(4); route(1); }; } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- ### First check and resetflags if UAS are behind the same NAT system: ### if (isflagset(6) && isflagset(7)){ log(2, " Both Clients are behind NAT"); # Store the destination domain into an AVP avp_printf("$avp(i:450)", "$dd"); if (avp_check("$avp(i:450)", "eq/$src_ip/g")){ log(3, " Detected Two Clients Behind the Same NAT - Disabling Mediaproxy"); # Do not use media-proxy as the clients seem to be behind the same NAT resetflag(6); resetflag(7); }; }; if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); force_rport(); fix_nated_contact(); force_rtp_proxy(); }; }; } route[5] { # ----------------------------------------------------------------- # ILD - PSTN Handler # ----------------------------------------------------------------- # Forward international calls to Asterisk (a SIP LDI Providers): if(uri=~"^sip:011"){ log(1, " 011N match - Larga Distancia Internacional a USA \n"); if (!is_user_in("from", "ldiusa")) { log(1," No permission for international calls to USA \n"); sl_send_reply("403", "No permission for international calls to USA"); acc_db_request("403 Forbidden", "missed_calls"); return; }; strip(2); }; if(uri=~"^sip:011"){ log(1," 011N match - Larga Distancia Internacional \n"); if (!is_user_in("from", "ldix")) { log(1, " No permission for international calls \n"); sl_send_reply("403", "No permission for international calls"); acc_db_request("403 Forbidden", "missed_calls"); return; }; }; rewritehostport("192.168.2.137:5070"); # Asterisk or PSTN GATEWAY IP ADDRESS t_on_failure("1"); route(4); route(1); } route[6] { # ----------------- Free Calls ------------------- # To Info numbers 102, 104, 123 y 080011236 if(uri=~"^sip:102@" || uri=~"^sip:104@" || uri=~"^sip:123@" || uri=~"^sip:0800123456@"){ log(1," To CallCenter xcompany!! \n"); rewriteuri("sip:123456@mydomain.com.pe"); rewritehostport("callcenter.mydomain.com.pe:5060"); route(4); route(1); return; }; if (is_user_in("from", "restringido")) { sl_send_reply("402", "Restriccion temporal"); log(1," 402 - Restriccion temporal \n"); acc_db_request("402 Restricted", "missed_calls"); return; }; lookup("aliases"); if (!is_uri_host_local()) { if (!isflagset(22)) { append_branch(); }; route(4); route(1); return; }; # -------------------------- Seccion de Rutas a LDI ---------------------------------------- # Forward international calls to Asterisk (To SIP LDI Providers): if (uri=~"^sip:011[0-9]*@" && !uri=~"^sip:01152*@") { route(5); return; }; if( uri=~"^sip:[2-8][0-9]{6}@" || uri=~"^sip:1[2-8][0-9]{6}@" || uri=~"^sip:01[2-8][0-9]{6}@" ) { log(1, " A Fijos Mexico PSTN \n"); if (uri=~"^sip:01[2-8][0-9][0-9][0-9][0-9][0-9][0-9]@") { strip(1); }; if (uri=~"^sip:[2-8][0-9][0-9][0-9][0-9][0-9][0-9]@") { prefix("1"); }; if (!is_user_in("from", "fijomexico")) { sl_send_reply("403", "No permission for local fixed calls"); log(1, " 403 - No permission for local fixed calls \n"); acc_db_request("403 Forbidden", "missed_calls"); return; }; rewritehostport("192.168.2.137:5070"); prefix("52"); route(4); route(1); return; }; # no existe? (debe ir al final siempre) if ( uri=~"^sip:[0-9]" && !does_uri_exist() ) { log(1, " Unable to locate route for this dnid - User Not Exist 3X \n"); prefix("mc"); acc_db_request("404 Number Invalid or Incomplete", "missed_calls"); rewritehostport("192.168.2.131:5070"); t_relay(); return; }; } route[7] { # ---------------------- Handling of Unavailable user ---------------------- # non-Voip -- just send "off-line" if (!(method=="INVITE" || method=="ACK" || method=="CANCEL" || method=="BYE" || method=="OPTIONS")) { sl_send_reply("404", "Not Found"); acc_db_request("404 Not Found", "missed_calls"); log(1, " 404 Not Found & non-Voip -- just send off-line \n"); return; }; # Not voicemail subscriber: if (!isflagset(4) && !method=="OPTIONS" && !method=="ACK" && !method=="BYE" && !method=="CANCEL") { #sl_send_reply("404", "Not Found and no VMail turned on"); acc_db_request("480 Temporarily Unavailable", "missed_calls"); # A locucion (Usuario temporalmente fuera de servicio 'ss-noservice'); log(1, " 404 Not Found and no voicemail \n"); prefix("fu"); ## su: exten => _su1780XXXX,1,Playback(ss-noservice,noanswer) rewritehostport("192.168.2.131:5070"); t_relay(); return; }; # forward to voicemail adding prefix to simplify asterisk "extension.conf" :: if (isflagset(4) && !method=="OPTIONS" && !method=="ACK" && !method=="BYE" && !method=="CANCEL") { prefix("vm"); acc_db_request("480 Temporarily Unavailable -> Vm", "missed_calls"); rewritehostport ("192.168.2.131:5070"); t_relay(); return; }; } # -------------------------------------------------------------------------- onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } failure_route[1] { if (t_check_status("487")) { return; }; if (t_check_status("486")){ prefix("vmb"); acc_db_request("486 Busy -> Vm", "missed_calls"); rewritehostport ("192.168.2.131:5070"); append_branch(); t_relay(); return; }; if (t_check_status("408")){ prefix("vmu"); acc_db_request("408 Timeout -> Vm", "missed_calls"); rewritehostport ("192.168.2.131:5070"); append_branch(); t_relay(); return; }; unforce_rtp_proxy(); } failure_route[2] { # handle 302 redirects if (t_check_status("3[0-9][0-9]")) { get_redirects("*"); t_relay(); }; } ### The End ###

18 years, 2 months

[Serusers] Via RPORT for CANCEL message.

by Ricardo Martinez

Hello. My main configuration is based on the ser.cfg from the issue 5 plus some modifications included by me. Beside that i'm using version: ser 0.9.3 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.197 2004/12/03 19:09:31 andrei Exp $ main.c compiled on 11:03:37 Dec 22 2005 with gcc 3.2 I have a question regarding to sending a CANCEL message for an INVITE from a NAT'd endpoint. As far as i know the force_rport() command adds the received IP port to the top most via header in the SIP message, this enables subsequent SIP message to return to the proper port later on in a SIP transaction. My problem is that for a CANCEL message coming from a NAT'd endpoint this command seems not to be working. This is the scenario. NAT'd endpoint : 200.100.100.248 SER : 200.100.100.246 SER-2 : 200.100.100.36 GW : 200.100.100.69 The NAT'd endpoint send an INVITE to the proxy, but then in the middle of the transaction decide to CANCEL the request. As you can see in the "cancel_debug.txt" file included on this mail, the CANCEL message does not contain the "rport" in the Via header, so it seems to be routed back to the default sip port (5060). Is there a way to force the rport in a CANCEL? for a enpoint unable to put the rport by itself? I want to do this according to the onsip document, in a compatible way.. Anyway i made a little test including a force_rport(); in the CANCEL handler .......... # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("200.100.100.246:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); setflag(1); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbidden"); }; break; }; if (method=="CANCEL") { force_rport(); route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; } else if (method=="ACK") { route(1); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); this solve my problem but it seems not to be so accurate. I hope that someone can help me Thanks! Ricardo Martinez.- <<cancel_debug.txt>>

18 years, 2 months

SV: [Users] Route-header DNS lookup?

by Kerker Staffan

hi is this issue (maddr=) fixed in the 1.1.0 release? do you know if the db_mode=3 is in the 1.1.0 release? best regards, /staffan > -----Ursprungligt meddelande----- > Från: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Skickat: den 31 juli 2006 09:36 > Till: Simon Morvan > Kopia: Kerker Staffan; users(a)openser.org > Ämne: Re: [Users] Route-header DNS lookup? > > Hi Simon! > > Please make a unified diff (cvs diff -u) and post it at the > bugtracker on sourceforge: > http://sourceforge.net/tracker/?group_id=139143&atid=743022 > > This way it can't get lost under thousands of emails. > > regards > klaus > > Simon Morvan wrote: > > modified file : modules/rr/loose.c > > > > 2006/7/28, Simon Morvan <garphy(a)gmail.com>: > >> after a quick check, the answer is : no > >> > >> this is a quick hack i've made as a workaround. feedback > welcome :) > >> > >> 781,783c781,803 > >> < if (set_dst_uri(_m, uri) < 0) { > >> < LOG(L_ERR, "after_loose: Error > while setting > >> dst_uri\n"); > >> < return RR_ERROR; > >> --- > >> > if( puri.maddr.s != NULL ){ > >> > str builturistr; > >> > char builturi[150]; > >> > > >> > memcpy( builturi, "sip:", 4 ); > >> > memcpy( builturi+4, puri.maddr.s+6, > >> puri.maddr.len-6 ); > >> > builturi[4+puri.maddr.len-6] ='\0'; > >> > > >> > LOG( L_ERR, "MaddrURI is %p\n", > builturi ); > >> > > >> > builturistr.s = builturi; > >> > builturistr.len = 4+puri.maddr.len-6; > >> > > >> > if (set_dst_uri(_m, &builturistr) < 0) { > >> > LOG(L_ERR, "after_loose: > Error while > >> setting dst_uri\n"); > >> > return RR_ERROR; > >> > } > >> > > >> > }else{ > >> > if (set_dst_uri(_m, uri) < 0) { > >> > LOG(L_ERR, "after_loose: > Error while > >> setting dst_uri\n"); > >> > return RR_ERROR; > >> > } > >> > >> > >> 2006/7/28, Simon Morvan <garphy(a)gmail.com>: > >> > Is this issue solved in Openser 1.1.0 ? > >> > > >> > 2006/7/7, Simon Morvan <garphy(a)gmail.com>: > >> > > As Daniel said on the ml thread : > >> > > > it makes sense, we will analyze the implications > and include it > >> on the > >> > > > to-do list. > >> > > > > >> > > > Cheers, > >> > > > Daniel > >> > > Has it been included in the developpement branch yet ? > >> > > > >> > > -- > >> > > Simon. > >> > > > >> > > 2006/7/7, Klaus Darilion <klaus.mailinglists(a)pernau.at>: > >> > > > I'm not sure about the maddr parameter. doesn't it > mean that > >> > > > the > >> request > >> > > > should be forwarded to this IP regardsless of the domain? If > >> yes, the > >> > > > DNS lookup can be skipped. > >> > > > > >> > > > regards > >> > > > klaus > >> > > > > >> > > > Kerker Staffan wrote: > >> > > > > hi > >> > > > > i recently bounced into this problem, and i'm not > sure here. > >> > > > > i'm running the openser-devel, with the cacheless > db_mode=3. > >> (works fine btw) > >> > > > > > >> > > > > the record-route header received by the proxy on the other > >> side (SNOM4S), inserts > >> > > > > the domain name (iptel1.ipatl.se) and not the hostname > >> (sip.iptel1.ipatl.se) in the > >> > > > > record-route header, and uses the > maddr=<ip_of_server> with > >> the actual server IP address. > >> > > > > > >> > > > > now, when my client (behind the OpenSER) replies > with an ACK > >> to the incomming OK, > >> > > > > it uses the Route-header recieved in the > RR-header, and sends > >> the ACK to OpenSER. i > >> > > > > then get the following errors in OpenSER. > >> > > > > > >> > > > > --- > >> > > > > /usr/local/sbin/openser[3583]: ERROR: mk_proxy: could not > >> resolve hostname: "iptel1.ipatl.se" > >> > > > > /usr/local/sbin/openser[3583]: ERROR: uri2proxy: bad host > >> > > > > name > >> in URI > >> > <sip:4ffec4ce755c218a72228c6643cb3b6b@iptel1.ipatl.se:5060;maddr=172. > >> 28.248.66;transport=udp;lr> > >> > >> > > > > --- > >> > > > > > >> > > > > the ACK i sent look like this: > >> > > > > > >> > > > > --- > >> > > > > Request-Line: ACK > sip:2307@iptel1.ipatl.se;gruu=6fg9n6dl SIP/2.0 > >> > > > > Via: SIP/2.0/UDP > >> 172.28.248.52:2051;branch=z9hG4bK-d96b1fvapkyn;rport > >> > > > > Route: <sip:172.28.248.10;lr=on;ftag=li9buf1i4p> > >> > > > > Route: > >> > <sip:4ffec4ce755c218a72228c6643cb3b6b@iptel1.ipatl.se:5060;ma > ddr= > >> 172.28.248.66;transport=udp;lr> > >> > > > > From: "Snom 2652" <sip:2652@ipatl.se>;tag=li9buf1i4p > >> > > > > To: <sip:2307@ipatl.se>;tag=hvseiz7kgb > >> > > > > Call-ID: 3c269d83900b-xj3ild14y880@snom360 > >> > > > > CSeq: 1 ACK > >> > > > > Max-Forwards: 70 > >> > > > > Contact: <sip:2652@172.28.248.52:2051;line=cp4a7ljd> > >> > > > > Content-Length: 0 > >> > > > > --- > >> > > > > > >> > > > > as far as i understand, according the rfc 3263, the > >> route-header may contain domain name that > >> > > > > has to be resolved using SRV. > >> > > > > > >> > > > > --- > >> > > > > "6 Constructing SIP URIs > >> > > > > > >> > > > > In many cases, an element needs to construct a > SIP URI for > >> inclusion > >> > > > > in a Contact header in a REGISTER, or in a Record-Route > >> header in an > >> > > > > INVITE. According to RFC 3261 [1], these URIs have to > >> > > > > have > >> the > >> > > > > property that they resolve to the specific element that > >> inserted > >> > > > > them. However, if they are constructed with just an IP > >> address, for > >> > > > > example: > >> > > > > > >> > > > > sip:1.2.3.4 > >> > > > > > >> > > > > then should the element fail, there is no way > to route the > >> request or > >> > > > > response through a backup. > >> > > > > > >> > > > > SRV provides a way to fix this. Instead of using an IP > >> address, a > >> > > > > domain name that resolves to an SRV record can be used: > >> > > > > > >> > > > > sip:server23.provider.com" > >> > > > > --- > >> > > > > > >> > > > > now, OpenSER only asks DNS for an A record of the name > >> recieved in the route header, > >> > > > > and since that's a domain name, it's > unresolvable, and so the > >> ACK is never sent. > >> > > > > > >> > > > > any hints or clues? > >> > > > > > >> > > > > best regards, > >> > > > > /Staffan Kerker > >> > > > > > >> > > > > > >> > > > > -- > >> > > > > Staffan Kerker > >> > > > > Saab Communications, Växjö > >> > > > > p. +46 470 42185 > >> > > > > c. +46 705 391365 > >> > > > > m. staffan.kerker(a)saabgroup.com > >> > > > > > >> > > > > > >> > > > > _______________________________________________ > >> > > > > 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, 2 months

[Serusers] IMPORANT ANNOUNCEMENT: Launch of new iptel.org site!!

by Greger V. Teigre

Dear all! We are doing this the soft way: Launching when everybody is on vacation :-D We have for some time now been working on putting up a new community-based framework for the iptel.org open source projects. Our aim is to step-wise address lack of documentation, roadmap, release plans, transparent management and so on. The new http://iptel.org/ site contains quite a lot of content. There is still plenty to migrate from the old iptel.org site, ONsip.org, as well as better organize documentation around SER development and releases. You can help out with that by adding a small FAQ, how-to, copy something from ONsip.org that you found particularly helpful and so on! We have created some basic content that we hope you will help comment and develop further. Here are some pointers to content on the new site: ======================================== About the new site and how to contribute content http://www.iptel.org/iptelorgsite/intro How to get started with SER? http://www.iptel.org/ser/using Overview of documentation http://www.iptel.org/doc The old ONsip.org Getting Started document http://www.iptel.org/ser/doc/gettingstarted Contributed modules and patches http://www.iptel.org/ser/contributions A beginning skeleton for a roadmap http://www.iptel.org/ser/roadmap A wishlist for future versions http://www.iptel.org/ser/wishlist Release policies http://www.iptel.org/ser/development/release_policies ====================================================== Below is the launch announcement. Best regards, Greger Teigre g-) --------------------------------------------------------- The new http://iptel.org/, home of SER, SEMS, and SERWeb is now up. If you have an account from ONsip.org, you can log in using your old ONsip.org username/password. However, all content is available without registration. The new site is supported by the joint efforts of the iptel.org and ONsip.org teams. Our aim is to dramatically improve the project communication, documentation, and development process. The start is already up there, and we have many plans. The best thing of all: The new http://iptel.org site is a great basis for contributions from anyone. We are working on migrating and organizing the remaining content from the old site, but meanwhile you can access everything from the old site from http://dog.iptel.org/ We look forward to a revitalised SER open source project!

18 years, 2 months

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sr-users August 2006