sr-users September 2004

sr-users@lists.kamailio.org

190 participants
382 discussions

[Serusers] two ser proxies and forward between them

by ser die

hello friends i have the problem while forwarding the call to another ser proxy so my scenario is 215104 -- first ser --- second ser --- 36105 ua ua so in my first ser proxy of ser.cfg i have the used the forward like this if(uri=="sip:36105@XXX.XXX.XXXX.in") { forward(xxx.xxx.xxx.18,5060); break; } else { if !lookup("location")) { sl_send_reply ("404","NotFound"); break; }; } actaully the operation should be like this when any user of first ser proxy dial to the second ser proxy user the invite should be directed to the second ser proxy and call should be established but in the 214105 it says not found 36105 so how to proceed from here what are the changes i need to made to my ser.cfg with regards serdiehard __________________________________ Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. http://promotions.yahoo.com/new_mail

20 years, 5 months

[Serusers] Problems running CVS STABLE

by Markos Vakondios

Hi, I downloaded the stable cvs versions a couple of days ago but I haven't been able to make run yet. I have compiled with both bison and yacc as it mentioned in a previous post but no luck, even in different PCs. I am running SuSE 9, SER compiles with warnings about .d files all along but the build succeeds. Nevertheless, when I try to start it I get the following output: 0(2363) read 852533296 from /dev/random 0(2363) seeding PRNG with 1946404459 0(2363) test random number 166762925 0(2363) shm_mem_init: success 0(2363) DEBUG: register_fifo_cmd: new command (print) registered 0(2363) DEBUG: register_fifo_cmd: new command (uptime) registered 0(2363) DEBUG: register_fifo_cmd: new command (version) registered 0(2363) DEBUG: register_fifo_cmd: new command (pwd) registered 0(2363) DEBUG: register_fifo_cmd: new command (arg) registered 0(2363) DEBUG: register_fifo_cmd: new command (which) registered 0(2363) DEBUG: register_fifo_cmd: new command (ps) registered 0(2363) DEBUG: register_fifo_cmd: new command (kill) registered debug=1 0(2363) parse error (1,3-5): parse error fk= log_stderr= check_via= dns= rev_dns= #pt=5 #children= fifo="/tmp/ser_fifo # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/sl.so loadmodule "/usr/local/lib/ser/modules/rr.so loadmodule "/usr/local/lib/ser/modules/usrloc.so # Uncomment this if you want digest authenticati # mysql.so must be loaded #loadmodule "/usr/local/lib/ser/modules/auth.so # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", # Uncomment this if you want to use SQL database # f persistent stage comment the previous line #modparam("usrloc", "db_mode", 2 # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", # # If you set "calculate_ha1" parameter to (which in this cfig), # uncomment also the following parameter # #modparam("auth_db", "passwd_column", "passwd" # -- 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 # initial sanity checks -- messages with # max_fwards==, excessively lg requests if (mf_process_maxfwd_header("1")) sl_send_reply("483","Too Many Hops") break } if ( msg:len > max_len ) sl_send_reply("513", "Message too big") break } # we recd-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream downstream entities # use different transpt protocol recd_route(); # loose-route processing if (loose_route()) t_relay() break } # if the request is f other domain use UsrLoc # (in case, it does t wk, use the following comm # with proper names addresses in it if (==myself) if (method=="REGISTER") # Uncomment this if you want to use digest authenticati # if (www_authize("iptel.g", "subscriber")) # www_challenge("iptel.g", "") # break # } save("locati") break } # native SIP destinatis are hled using our USRLOC DB if (lookup("locati")) sl_send_reply("44", "Not Found") break } } # fward to current w; use stateful fwarding; that # wks reliably even if we fward from TCP to if (t_relay()) sl_reply_err() } ERROR: bad config file (1 errors) 0(2363) shm_mem_destroy 0(2363) destroying the shared memory lock Why does it output the config file?And why is it corrupted? Please help... Thanks Markos

20 years, 5 months

[Serusers] Monitoring Ser end points

by prajyot

20 years, 5 months

[Serusers] User ser+nathelper+rtppoxy as outbound server

by Jiang Zhou

I want to cofig ser as a outbound server to resolve all nat problem. The follow picture is the system architecture. uac--nat-->outboud1(ser1+nathelper+rtpproxy)------------->ser(main) uac--nat-->outboud2(ser2+nathelper+rtpproxy)-------------| uac--nat-->outboud3(ser3+nathelper+rtpproxy)-------------| Can someone know if ser can rewrite the uac's contact as itself ip address and forward to main ser? Nathelper can rewrite sdp ip/port to it own ip/port. But it only rewrite contac with nat's ip and port.

20 years, 5 months

RE: [Serusers] Radius Accounting Message Format.

by Ricardo Martinez

Hi Jesus. Thanks for you response. And what about a timestamp parameter inserted by SER? is that possible? Thanks in advance. Ricardo.- > -----Mensaje original----- > De: Jesus Rodriguez [SMTP:jesusr@voztele.com] > Enviado el: Miércoles, 01 de Septiembre de 2004 06:20 p.m. > Para: Ricardo Martinez > CC: serusers(a)lists.iptel.org > Asunto: Re: [Serusers] Radius Accounting Message Format. > > On Wed, 1 Sep 2004, Ricardo Martinez wrote: > > Hello, > > > I succesfully install the auth and acc modules for SER. I'm trying > > to keep track for the duration of a "local" (or between two registered > user > > agents in SER) call. I understand that for a PSTN or Gateway call the > > Accounting record is placed by the terminating gateway, but i'm > interesting, > > as i mentioned, in the local calls. I copy two accounting records from > my > > RADIUS server for a local call, as you can see i don't have any > parameter > > indicating the duration of the call or even timstamps. Do i have to set > > anything else in my ser.cfg file?. I'm also attaching my ser.cfg. > > No session time attribute exists because it's not calculated by SER. SER > will just give you one Start and one Stop record per session and you will > have to calculate the session time yourself. > > > > > > Code: Accounting-Request > > Attributes: > > Acct-Status-Type = Start > > Service-Type = Sip-Session > > Sip-Response-Code = 200 > > Sip-Method = 1 > > User-Name = "5552408196(a)xx.xx.xx.246" > > Calling-Station-Id = "sip:5552408196@xx.xx.xx.246" > > Called-Station-Id = "sip:005622408196@xx.xx.xx.246" > > Sip-Translated-Req-ID = "sip:005622408196@xx.xx.xx.50:5060" > > Acct-Session-Id = > > "95e15442-5f12-fae4-805e-0002a400f1e9(a)xx.xx.xx.242" > > Sip-To-Tag = "4a419534a4" > > Sip-From-Tag = "9542fa5ea4" > > Sip-Cseq = "33" > > NAS-IP-Address = 127.0.0.1 > > NAS-Port = 5060 > > Acct-Delay-Time = 0 > > > > Code: Accounting-Request > > Attributes: > > Acct-Status-Type = Stop > > Service-Type = Sip-Session > > Sip-Response-Code = 200 > > Sip-Method = 8 > > User-Name = "5552408196(a)xx.xx.xx.246" > > Calling-Station-Id = "sip:5552408196@xx.xx.xx..246" > > Called-Station-Id = "sip:005622408196@xx.xx.xx.xx.246" > > Sip-Translated-Req-ID = "sip:005622408196@xx.xx.xx.50" > > Acct-Session-Id = > > "95e15442-5f12-fae4-805e-0002a400f1e9(a)xx.xx.xx.xx.242" > > Sip-To-Tag = "4a419534a4" > > Sip-From-Tag = "9542fa5ea4" > > Sip-Cseq = "34" > > NAS-IP-Address = 127.0.0.1 > > NAS-Port = 5060 > > Acct-Delay-Time = 0 > > > > Saludos > JesusR. > > ------------------------------- > Jesus Rodriguez > VozTelecom Sistemas, S.L. > jesusr(a)voztele.com > http://www.voztele.com > Tel. 902360305 > -------------------------------

20 years, 5 months

[Serusers] Compare "Contact" and "Digest username" in REGISTER

by Gerhard Zweimueller

Hi list, we operate a commercial PSTN-replacement service. For my setup we need to restrict the contact-information that the user register with. Only the assigned E.164 number in "Contact" shall be allowed with a specific "Digest username" in REGISTER Our contact and digest_username are very similar and only differ by a prefixed '0' in contact. E.g. Contact: <sip:0123456@cpe.network:5060;transport=UDP> Authorization: Digest username="123456", realm="network...... Is it possible to compare these two and only allow registration if the comparison returns true? Thanks for your help and best regards, Gerhard __________________________________________________________________________________ Dieses Mail wurde vom Infotech SecureMail Service ueberprueft und fuer sicher befunden. Fuer weitere Informationen zu Infotech SecureMail Service waehlen Sie bitte: www.infotech.at/securemail/ This email has been scanned by Infotech SecureMail Service and it has been classified as secure. For more information on Infotech SecureMail direct your web browser to: www.infotech.at/securemail/

20 years, 5 months

[Serusers] Early Media in SEMS/announcement

by Gerhard Zweimueller

Hi list, is it possible to use ser/sems as an announcement server with early media? I.e. the call will NOT be connected with the other SIP-UA with OK/ACK, but after sending 183 the announcement is played in the early-media channel? This would facilitate proper accounting, where announcements should not be billed. Any help is appreciated. Thank you very much, Gerhard __________________________________________________________________________________ Dieses Mail wurde vom Infotech SecureMail Service ueberprueft und fuer sicher befunden. Fuer weitere Informationen zu Infotech SecureMail Service waehlen Sie bitte: www.infotech.at/securemail/ This email has been scanned by Infotech SecureMail Service and it has been classified as secure. For more information on Infotech SecureMail direct your web browser to: www.infotech.at/securemail/

20 years, 5 months

[Serusers] Failover PSTN Gateways

by Gerhard Zweimueller

Hi list, We have a setup with 2 redundant stateful SER SIP-Servers with accounting. For calls to the PSTN we have 2 ISDN-PRI gateways connected to different ISDN-PRI lines. We need to implement a dynamic failover mechanism and load balancing. a) Incoming calls: No problem, the PSTN switch takes care of that b) Outgoing calls: Both GWs are registered at both SERs; calls are usually relayed with "t_relay_to_udp(x.x.x.x, "5060");" But how can I implement a quick failover including load-distribution between those gatways? I.e. if I receive "busy here" or no response after a short period after the "invite", the call should be redirected to the other GW. Thanks in advance for your help! Best regards, Gerhard __________________________________________________________________________________ Dieses Mail wurde vom Infotech SecureMail Service ueberprueft und fuer sicher befunden. Fuer weitere Informationen zu Infotech SecureMail Service waehlen Sie bitte: www.infotech.at/securemail/ This email has been scanned by Infotech SecureMail Service and it has been classified as secure. For more information on Infotech SecureMail direct your web browser to: www.infotech.at/securemail/

20 years, 5 months

[Serusers] Radius Accounting Message Format.

by Ricardo Martinez

Hi. I succesfully install the auth and acc modules for SER. I'm trying to keep track for the duration of a "local" (or between two registered user agents in SER) call. I understand that for a PSTN or Gateway call the Accounting record is placed by the terminating gateway, but i'm interesting, as i mentioned, in the local calls. I copy two accounting records from my RADIUS server for a local call, as you can see i don't have any parameter indicating the duration of the call or even timstamps. Do i have to set anything else in my ser.cfg file?. I'm also attaching my ser.cfg. Code: Accounting-Request Attributes: Acct-Status-Type = Start Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = 1 User-Name = "5552408196(a)xx.xx.xx.246" Calling-Station-Id = "sip:5552408196@xx.xx.xx.246" Called-Station-Id = "sip:005622408196@xx.xx.xx.246" Sip-Translated-Req-ID = "sip:005622408196@xx.xx.xx.50:5060" Acct-Session-Id = "95e15442-5f12-fae4-805e-0002a400f1e9(a)xx.xx.xx.242" Sip-To-Tag = "4a419534a4" Sip-From-Tag = "9542fa5ea4" Sip-Cseq = "33" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 Acct-Delay-Time = 0 Code: Accounting-Request Attributes: Acct-Status-Type = Stop Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = 8 User-Name = "5552408196(a)xx.xx.xx.246" Calling-Station-Id = "sip:5552408196@xx.xx.xx..246" Called-Station-Id = "sip:005622408196@xx.xx.xx.xx.246" Sip-Translated-Req-ID = "sip:005622408196@xx.xx.xx.50" Acct-Session-Id = "95e15442-5f12-fae4-805e-0002a400f1e9(a)xx.xx.xx.xx.242" Sip-To-Tag = "4a419534a4" Sip-From-Tag = "9542fa5ea4" Sip-Cseq = "34" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 Acct-Delay-Time = 0 # # $Id: ser.cfg,v 1.21.2.2 2003/10/13 22:53:06 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=0 # 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/ser_fifo" # ------------------ 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/registrar.so" loadmodule "/usr/local//lib/ser/modules/textops.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" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/group_radius.so" loadmodule "/usr/local/lib/ser/modules/acc.so" # ----------------- setting module-specific parameters --------------- # ----------------- setting module-specific parameters --------------- # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("acc","log_level",1) modparam("auth_radius","radius_config","/usr/local/etc/radiusclient/radiuscl ient.conf") modparam("auth_radius","service_type",15) modparam("acc","radius_config","/usr/local/etc/radiusclient/radiusclient.con f") modparam("acc", "service_type", 15) modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 3) modparam("acc", "failed_transactions", 1) modparam("acc", "report_ack", 0) #group radius modparam("group_radius","radius_config","/usr/local/etc/radiusclient/radiusc lient.conf") modparam("group_radius", "use_domain", 1) # ------------------------- 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")) { sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) { 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=="BYE") setflag(1); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # 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 (method=="REGISTER") { # Challenge/Response if ( !radius_www_authorize("")) { www_challenge("", "1"); break; }; # Si es un Request Autenticado, Update Contact Database ahora (Ricardo Martinez) if (!save("location")) { sl_reply_error(); }; break; }; }; #Proceso el trafico local hacia la PSTN (Ricardo Martinez) #Busco el Username Canonico? (Ricardo Martinez) lookup("aliases"); #Chequeamos los destinos 00562 es enviado al gateway if (uri=~"^sip:00562[0-9]{7}@.*") { route(3); break; }; #Llamadas SIP locales #Off-line o usuarios que no existen (Ricardo Martinez) if (!lookup("location")) { route(4); break; }; #Flag para el modulo de acc para reportar "Missed Calls" al syslog (Ricardo Martinez) setflag(3); #Do it ? (Ricardo Martinez) append_hf("P-hint: USRLOC\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } /* end of initial routing logic */ #--------- Route 1 que estaba de antes ------------------------------- route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } #------- Route 3 : Trafico de Internet hacia la PSTN ------------------- route[3] { # Agregamos record route para todas las llamadas. record_route(); #Autentificamos al usuario #PRONTO if (method=="INVITE") { setflag(1); # Aca en la eventualidad van los chequeos de Saldo y llamadas LD, LDI, Local, etc }; rewritehostport("XX.XX.XX.XX:5060"); append_hf("P-hint: GATEWAY\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } #------ Route 4 : Llamadas a usuarios Off-line ------------------------ route[4] { if (!t_newtran()) { sl_reply_error(); }; if (!t_reply("404", "Not Found")) { sl_reply_error(); }; break; } Any ideas? Thanks in advance. Ricardo Martinez.-

20 years, 5 months

[Serusers] Ser/pa module and publish request

by Ansari, Mohammed

Hi, I send a Publish request to the SER server and it returns a 200 OK without the SIP-ETag: header. Anyone knows why? Also a subsequent SUBSCRIBE request comes back with a "400 Bad request" response. Does the SER pa module support subscribe/notify sip extensions? Some documentation seem to suggest subscribe/notify is supported only in Jabber? Is this correct? I am a bit confused. Thanks for the answers. Ansari

20 years, 5 months

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sr-users September 2004