sr-users January 2006

sr-users@lists.kamailio.org

134 participants
259 discussions

[Users] OpenSER TLS with Windows Messenger
by Cagri Koksal 16 Jan '06

16 Jan '06

Hi all, I'm trying to test openset TLS feature with windows messenger. I setup openser on Debian Linux and modified the openser.cfg and openssl.cnfas required by the TLS faq and docs at openser.org. When I try to sign in using windows messenger I got the following message from OpenSer: "tls-accept: Error in SSL:" At the same time windows messenger gives the following error messenger; "There was a problem verifying the certificate from the server ....." Here is an excerpt from my openser.cfg - disable_tls = 0 listen = tls:192.168.0.128:5061 tls_verify = 0 tls_require_certificate = 0 tls_method = TLSv1 ( I also tried SSLv23 ) tls_certificate = "/usr/local/etc/openser/tls/sipsrv_rootCA/sipsrv01/cert.pem" tls_private_key = "/usr/local/etc/openser/tls/sipsrv_rootCA/sipsrv01/privkey.pem" tls_ca_list = "/usr/local/etc/openser/tls/sipsrv_rootCA/sipsrv01/calist.pem" and an excerpt from my openssl.cnf dir = "/usr/local/etc/openser/tls/sipsrv_rootCA/demoCA" certs = $dir/certs crl_dir = $dir/crl database = $dir/index.txt new_certs_dir = $dir/newcerts certificate = $dir/cacert.pem serial = $dir/serial crl = $dir/crl.pem I also used the existing certificates that come with openser-tls package. It resulted the same. Any suggestion or guidance is appreciated, thanks. -Cagri Koksal

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[Users] Radius Accounting
by martin＠campus-merseburg.de 16 Jan '06

16 Jan '06

Hi, why does OpenSER sends three Start-Radius-Packets on an Invite und and two on a bye? My config has these entries: [...] modparam("acc", "radius_flag", 3) [...] if (method=="BYE" || method=="CANCEL") { setflag(3); }; [...] } else if (method=="INVITE") { setflag(3); [...]

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[Users] TM functions : merging
by Bogdan-Andrei Iancu 15 Jan '06

15 Jan '06

Hi everybody, I just committed on CVS head some changes into TM script functions - related functions are merged into new ones; the goal is to simplify the usage of these functions and to open way for NAPTR lookup. Following functions were merged: - t_relay_to_udp(), t_relay_to_tcp(), t_relay_to_tls() merged into t_relay(proto:host:port); - t_forward_nonack_uri(), t_forward_nonack_udp(), t_forward_nonack_tcp(), t_forward_nonack_tls() merged into t_forward_nonack(proto:host:port) and t_forward_nonack(); - t_replicate_udp(), t_replicate_tcp(), t_replicate_tls() merged into t_replicate(proto:host:port) Note that the functionality is not affected. To make the transition easier, the replaced functions are not removed yet, but marked as obsolete (via documentation and messages at runtime). The plan is to have them removed soon - a second set of changes into the API is to come. regards, bogdan

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[Users] route on reply reason in failure_route
by Walter Schober 15 Jan '06

15 Jan '06

Can anyone please give me a hint how to access the reply reason in failure_route - or is it really not possible to trigger some event on the response _reason_. 0(704) SIP Reply (status): 0(704) version: <SIP/2.0> 0(704) status: <404> 0(704) reason: <CLIR Inactive> <----- I want this! There is a pseudo variable for that, but no function can search the replied reason! Is it really true, that you can use the reply status only (t_check_status("404"))? Using avpops the pseude variables $rr and $rs are <null> at that moment. Do I miss something? I'm doing that: failure_route[3] { if (t_check_status("404") ) { avp_printf("$reason", "The reply: $rr, $rs"); log("\n----------------------------------------\n"); avp_print(); log("\n----------------------------------------\n"); } } resulting in: 0(740) DEBUG:avpops:print_avp: p=0xb6177838, flags=2 0(740) DEBUG: id=<32> 0(740) DEBUG: val_str=<The reply: <null>, <null>> Any hints welcome. /Walter

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[Users] Problem with IAX and Openser.
by david.castro 13 Jan '06

13 Jan '06

Hello. I am in a strange situation. I have two asterisk. Asterisk "A" makes a call for asterisk "B" by IAX. Asterisk "B" recives the call and delivers it to Openser by SIP. The problem is openser printing this in the screen: ERROR: parse_to : unexpected char ["] in status 5: <<"David" <sip:>> . ERROR:parse_from_header: bad from header ERROR: new_t: no valid From in INVITE ERROR: t_newtran: new_t failed ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) Do you know the reason? Is this a bug? Which is guilty, asterisk or openser? I have this problem only in this scenario. Thank you. -- -------------------------------- David Castro Mosquera Adianta Tel: (34) 902 999 178 Fax: (34) 902 501 437 Email: david.castro(a)adianta.net Web: www.adianta.net --------------------------------

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[Users] New functionality : Direction detection in ACC module
by Bogdan-Andrei Iancu 13 Jan '06

13 Jan '06

Hi everybody, I just committed on CVS head a new functionality that was requested by Lenir <lenirsantiago(a)yahoo.com> some time ago. It's about detecting and compensating in the ACC records of the direction flow for sequential requests. For more details about it and about how to used it, please see the online documentation: http://openser.org/docs/modules/1.1.x/acc.html#AEN625 Any testing/feedback is highly appreciated ;). regards, bogdan

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[Users] delete specific contacts from location
by Mike Williams 13 Jan '06

13 Jan '06

Is it possible to delete a specific contact location for a user, and by using openserctl? I see that adding and deleting a certain contact is on the openser roadmap, and if this means that this functionality has made it into openser but not into openserctl, than I'm willing to try to add it myself. Any advice where to start is appreciated. Mike Williams

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[Users] Functionality
by Andréa Giordanna 13 Jan '06

13 Jan '06

Hi all I need a functionality on openser. The functionality would allow users to save on the server - a message, - a time and date - and a group of uris, and at the given time, the server would send the message to each one of the group of uris. This functionality would be useful for users to send reminders of events to a list of uris. Are there any modules on openser that do something like that? If not, how could I implement it? Thanks Andrea Giordanna --------------------------------- Yahoo! doce lar. Faça do Yahoo! sua homepage.

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[Users] about sems
by unplug 12 Jan '06

12 Jan '06

Since I have no idea about asterisk, and I would like to ask can I use sems instead of asterisk to provide same function. Any openser+sems howto available for fedora core4?

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[Users] Openser as a Outbound Proxy
by david.castro 12 Jan '06

12 Jan '06

This is the sip.conf from asterisk: ########################################################################### [general] bindport=5062 bindaddr=0.0.0.0 ;canreinvite=no disallow=all ;allow=all register => asterisk:asterisk@my_openser_ip ;you can found more info of this in http://www.voip-info.org/wiki/ [ser] allow=ulaw allow=alaw allow=g723.1 allow=g726 allow=g729 allow=gsm allow=ilbc allow=lpc10 allow=speex allow=adpcm type=friend ; We allow incoming and outgoing calls. Use peer if you are only doing MWI context=internal ; This is the context incoming calls land in host=ip_ser ; This is the hostname or IP address of your SER server ;fromdomain=ser.server.rld ; This is your SER_DOMAIN insecure=very ; This allows incoming calls from the phones routing through ser to be passed into asterisk dtmfmode=rfc2833 This is the openser.cfg ###################################################################### # # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/openser_fifo" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls #tls_verify = 1 #tls_require_certificate = 0 #tls_require_certificate = 1 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_certificate = "/usr/local/etc/openser/tls/openser1/cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_private_key = "/usr/local/etc/openser/tls/openser1/privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" #tls_ca_list = "/usr/local/etc/openser/tls/openser1/calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database 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/textops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/uac.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" #loadmodule "/usr/local/lib/openser/modules/avpops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/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) # -- 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) # ------------------------- request routing logic ------------------- # main routing logic route{ xlog("yo soy <$si> y el destino es <$tu> <$td> <$du> <$rm> \n"); # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # ----------------------------------------------------------------- # Handle NOTIFY requests from Sipura's to keep-alive NAT # ----------------------------------------------------------------- if ((is_method("NOTIFY")) && (! uri=~"sip:.*[@]+.*")) { xlog("entre en notify \n"); sl_send_reply("200", "OK keep-alive"); return; } # ----------------------------------------------------------------- # Only allow known sip messages,ack, pq no hace falta responder. # ----------------------------------------------------------------- if (!is_method("INVITE|REGISTER|BYE|ACK|CANCEL")) { xlog("Estoy en metodo no diponible y que es <$rm>\n"); sl_send_reply("501", "Not implemented here"); return; }; # 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(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { xlog("entree en loose route\n"); # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) # if (uri==myself) { if (uri=~"^sip:*.") { if (method=="ACK" && (src_ip!=XXXXXX)){ # Uncomment this if you want to use digest authentication if (!www_authorize("XXXXXXX", "subscriber")) { www_challenge("XXXXXXXX", "0"); exit; }; consume_credentials(); xlog("Voy a mandar respuesta a un ACK, yo soy <$si> y el destino es <$tu> <$td> <$du>\n"); route(1); exit; }; if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("XXXXX", "subscriber")) { www_challenge("XXXXX", "1"); exit; }; xlog("Me llego un register y yo soy <$si> y el destino es <$tu> <$td> <$du>\n"); consume_credentials(); save("location"); exit; }; if (method=="INVITE") { # Uncomment this if you want to use digest authentication if (!www_authorize("XXXXX", "subscriber") && (src_ip!=XXXXXX) ) { www_challenge("XXXXX", "0"); exit; }; # consume_credentials(); xlog("Un tio mando un invite <$si> y el destino es <$tu> <$td> <$du> \n"); sl_send_reply("100", "Trying"); lookup("aliases"); #Si es de asterisk y no encuentra el telefono digo que no esta. if (!lookup("location") && (src_ip==XXXXXXXX) ) { #xlog("no encuentra el destino\n"); sl_send_reply("404", "Not Found"); exit; } else if (lookup("location") && (src_ip==XXXXX)){#if a call come from * and sip phone online i route to ring sl_send_reply("180", "Ringing"); route(2); return; } else {#otherway i forward to *. append_hf("P-hint: usrloc applied\r\n"); #sl_send_reply("180", "Ringing"); route(1); return; }; }; if (method=="CANCEL" && (src_ip==XXXXX)) {#Si viene de asterisk, lo mando con estado al telefo. #xlog("Me llego un cancel de asterisk\n"); route(2); return; } else if (method=="CANCEL" && (src_ip!=XXXXXX)) { #Si no viene de asterisk... # Uncomment this if you want to use digest authentication if (!www_authorize("XXXXXX", "subscriber")) { www_challenge("XXXXXX", "0"); exit; }; consume_credentials(); #xlog("Me llego un cancel de NO asterisk\n"); lookup("aliases"); if (!lookup("location")) {#primero compruebo que este numero existe. #xlog("no encuentra el destino\n"); sl_send_reply("404", "Not Found"); exit; } else { route(1); #Se lo mando a asterisk. return; }; }; if (method=="BYE") { # Uncomment this if you want to use digest authentication if (!www_authorize("XXXXX", "subscriber") && (src_ip!=XXXXXX) ) { www_challenge("XXXXX", "0"); exit; }; consume_credentials(); xlog("Me llego un BYE \n"); if (src_ip!=XXXXXXX){ route(1); } else { route(2); }; sl_send_reply("200", "OK"); exit; }; }; } #I route to asterisk here route[1] { xlog("Estoy en route(1) \n"); forward(XXXXX,5062); return; } #I route from asterisk to sip phones here route[2] { xlog("Estoy en route(2)\n"); if (!t_relay()) { #xlog("Dentro de error del t_relay del 2\n"); sl_reply_error(); }; return; } ################################################### Notes: XXXX=ip_asterisk This configuration works in the same net. Now, I'm trying to work behind a nat. This configuration don't works behind a nat. I don't challenge asterisk, if a call comes from asterisk_ip, I trust. Bye.

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sr-users January 2006