sr-users August 2006

sr-users@lists.kamailio.org

184 participants
361 discussions

[Serusers] forwarded calls got timeout when called part hangup
by ashou han 10 Aug '06

10 Aug '06

Hi there, I use a t_on_failure("1") in route[1] for a unavailable server to another sip server, but always get timeout (code=408) when the called part hangup, no error (code=200) when calling part hangup. Please help!! here is my ser.cfg and log: ============================== # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes #log_stderror=yes # (cmd line: -E) listen=127.0.0.1 /* 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/postgres.so" loadmodule "/usr/local/lib/ser/modules/avpops.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" #loadmodule "/usr/local/lib/ser/modules/xlog.so" loadmodule "/usr/local/lib/ser/modules/acc.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 --------------- modparam("acc", "log_missed_flag", 2) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "log_fmt", "miocfsdrtup") #modparam("acc", "db_flag", 3) #modparam("acc", "db_missed_flag", 4) #modparam("acc", "failed_transactions", 1) #modparam("acc", "db_url", "postgres://gatekeeper:mm1ch0t@localhost/ser") #modparam("acc", "log_extra" ,"ua=hdr/User-Agent") #modparam("acc", "db_extra" ,"uuid=avp/i:888; src_ip=avp/i:999") modparam("avpops", "avp_aliases", "c_uri=i:888; s_ip=i:999" ) # -- 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) # — tm params -- # set time for which ser will be waiting for a final response; # fr_inv_timer sets value for INVITE transactions, fr_timer # for all others modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "fr_timer", 10 ) # ------------------------- 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 (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; if (method=="INVITE" || method=="BYE" || method=="CANCEL"){ ## Accounting for all calls setflag(1); # setflag(2); }; if (method=="INVITE") { record_route(); } # write the source IP as avp "s_ip" for accounting purposes # avp_write( "$src_ip", "$s_ip"); if (loose_route() ) { log("------- LOOSE ROUTE"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); log("------- IF ROUTE"); #route(1); break; } else { # write the current RURI as avp "c_uri" for accounting purposes #avp_write( "$ruri", "$c_uri"); append_hf("P-hint: outbound alias\r\n"); log("------- ELSE ROUTE"); route(1); break; }; append_hf("P-hint: usrloc applied\r\n"); log("------- LAST ROUTE"); #route(1); setflag(4); break; } route[1] { if ((src_ip==192.168.0.199)) { if ((uri=~"^sip:1617.*@127\.0\.0\.1") or (uri=~"^sip:1508.*@127\.0\.0\.1")) { log(1, "----------- Forwarding to OUR GW\n"); rewritehost("192.168.0.201"); log("----------- Host rewrited\n"); setflag(1); # if we do not get a positive reply, continue at reply_route[1] t_on_failure("1"); }else if ((uri=~"^sip:1.*@127\.0\.0\.1")) { log("----------- Forwarding to GW\n"); rewritehost("192.168.0.100"); log("----------- Host rewrited\n"); setflag(1); }; t_relay(); break; } } failure_route[1] { if (t_check_status("487")) { break; }; # forwarding failed -- try again at another destination if (method=="INVITE" && t_check_status("403|408|500|503")) { # revert_uri(); log("----------- Next gateway\n"); rewritehost("192.168.0.100"); log("----------- Host rewrited\n"); append_branch(); setflag(1); t_relay(); } } ========================================= log: Jul 31 15:17:38 localhost /usr/local/sbin/ser[3468]: ACC: transaction answered: method=INVITE, i-uri=sip:16175552222@127.0.0.1:5060, o-uri=sip:16175552222@192.168.0.201:5060, call_id=C3ECE80-200011DB-9498E149-4F3DC2BB(a)199.232.41.199, from=<sip:192.168.0.199>;tag=318DE490-1872, code=200, totag=SDa3kv999-78f0001f-0-2058902615, fromtag=318DE490-1872, to=<sip:16175552222@127.0.0.1>;tag=SDa3kv999-78f0001f-0-2058902615, uid=n/a, userpart=16175552222 Jul 31 15:17:38 localhost /usr/local/sbin/ser[3463]: ACC: request acknowledged: method=ACK, i-uri=sip:127.0.0.1:5060;ftag=318DE490-1872;lr=on, o-uri=sip:16175552222@192.168.0.100:5060;endpoint=127.2.0.43;transport=udp, call_id=C3ECE80-200011DB-9498E149-4F3DC2BB(a)192.168.0.199, from=<sip:192.168.0.199>;tag=318DE490-1872, code=200, totag=SDa3kv999-78f0001f-0-2058902615, fromtag=318DE490-1872, to=<sip:16175552222@127.0.0.1>;tag=SDa3kv999-78f0001f-0-2058902615, uid=n/a, userpart=n/a Jul 31 15:18:13 localhost /usr/local/sbin/ser[3471]: ACC: transaction answered: method=BYE, i-uri=sip:127.0.0.1:5060;ftag=318DE490-1872;lr=on, o-uri=sip:16175552222@192.168.0.100:5060;endpoint=127.2.0.43;transport=udp, call_id=C3ECE80-200011DB-9498E149-4F3DC2BB(a)192.168.0.199, from=<sip:192.168.0.199>;tag=318DE490-1872, code=408, totag=SDa3kv999-78f0001f-0-2058902615, fromtag=318DE490-1872, to=<sip:16175552222@127.0.0.1>;tag=SDa3kv999-78f0001f-0-2058902615, uid=n/a, userpart=n/a

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[Serusers] SER avpops avp_pushto
by Shaun Hofer 10 Aug '06

10 Aug '06

When I try to use avp_pushto() in one section of ser.cfg it works and in another section it doesn't. When I use it to out right forward a call it works fine (callfwd). When it's used in the failure_route[1] it doesn't seem to push a value onto ruri (I have made sure that avp_db_load() has been called/used) . I'm wondering if avp_pushto() one of those methods that can't be called from a failed route ? I get the following error when I use it: ERROR: t_forward_nonack: no branched for forwarding ERROR: w_t_relay (failure mode): forwarding failed ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) I tried taking away voicemail section in failure_route[1] and made sure I was phoning extension with a number instead of 'voicemail' in value column of the database. That made no difference. When I tried adding append_branch() after avp_pushto() it just ends up ringing the original phone number again. Thanks in advance -Shaun route[1] { # ---------------------------------------------------------- # Default Message Handler # ---------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[3] { # ---------------------------------------------------------- # INVITE Message Handler # ---------------------------------------------------------- if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (avp_db_load("$ruri/username", "s:callfwd")) { setflag(22); avp_pushto("$ruri", "s:callfwd"); route(6); break; }; if (!lookup("location")) { if (uri=~"^sip:0[1-9][0-9]{8}") { # Domestic PSTN route(4); route(5); break; }; sl_send_reply("404", "User Not Found"); break; }; if (avp_db_load("$ruri/username", "s:fwdnoanswer")) { if (!avp_check("s:fwdnoanswer", "eq/$ruri/i")) { setflag(27); }; }; log(1,"t on fail \n"); t_on_failure("1"); route(4); route(1); } route[4] { # ---------------------------------------------------------- # NAT Traversal Section # ---------------------------------------------------------- if (isflagset(6) || isflagset(7) || isflagset(11)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } route[6] { # ---------------------------------------------------------- # Call Forwarding Handler # ---------------------------------------------------------- lookup("aliases"); if (uri!=myself) { if (!isflagset(22)) { append_branch(); }; route(4); route(1); break; }; if (!lookup("location")) { if (uri=~"^sip:88[0-9]{8}") { # Offline VoIP Network user sent to * for voicemail route(4); route(5); break; }; sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } failure_route[1] { if (t_check_status("487")) { break; }; if (isflagset(26) && t_check_status("408")) { if (avp_check("s:fwdnoanswer","eq/voicemail/I")) { avp_delete("s:fwdnoanswer"); resetflag(26); revert_uri(); rewritehostport("x.x.x.x:5060");#goto voicemail server append_branch(); t_relay_to_udp("x.x.x.x", "5060"); break; } else { avp_pushto("$ruri", "s:fwdnoanswer"); avp_delete("s:fwdnoanswer"); resetflag(26); route(6); break; }; }; end_media_session(); }

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[Serusers] Problem with fr_inv_timer_avp
by Michel Zenone 10 Aug '06

10 Aug '06

Hi! Does somebody know why 'fr_inv_timer_avp' does nothing when I use it as follow: ... modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("avpops","avp_url","mysql://root@localhost/ser") modparam("avpops","avp_table","avptimer") # 'avptimer' is like 'usr_preferences' ... avp_db_load("$ruri","inv_timeout/avptimer"); Actually I try to set a fr_inv_timer different from an user to another. 'avptimer' timer defines, for each users, their proper fr_inv_timer. If somebody can help... Thanks in advance, Michel

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[Serusers] xmlrpc
by Victoria Cortez 10 Aug '06

10 Aug '06

Hello everybody, Can anybody tell me about xmlrpc module? I've found it in the CVS version. I could not find any docs about it yet. What is this module suppose to do, exactly? Thanks in advance, Victoria

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[Serusers] Looking for Call termination service provider
by John Kim 10 Aug '06

10 Aug '06

Hi, I am looking for a VoIP wholesaler who will allow us to operate our own SIP server infrastructure while we can forward all PSTN destined calls to their gateway. I really didn't think that it would be hard to find one but I have not been successful. I have checked so far with iCallGlobe.com and Deltathree.com. Can any of you recommend us some VoIP wholesaler with quality service and competitive rate who allows this business setup? Below is logical business relation setup: Our subscriber <--->ATA setup to connect sip.ours.com<----->PSTN gateway of wholesaler.com<--->PSTN Basically we are looking for good "wholesaler.com". Any comments will be appreciated. John K.

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Re: [Serusers] SS7 Signaling & Gateways
by Max Clark 09 Aug '06

09 Aug '06

Thanks Victor - unfortunately Cirpack does not support North American networks. On 8/9/06, Victor Stanescu <victor.stanescu(a)gtstelecom.ro> wrote: > I don't know what is reasonable for you, but I really advise you to take > a look at Cirpack (www.cirpack.com). It's the full option gateway/switch > (isdn, r2, ss7, v5, mgcp, sip, h.323, h.248), and if you take time to > study their hardware, you can start from very small and pay as you grow. > > I am using Cirpack's for 3 years and I was never more happier about > other telco equipment :-) > > Max Clark wrote: > > Hi all, > > > > I am looking for a _resonable_ cost SS7 solution to bundle with a TDM > > <-> SIP media gateway. I know the Cisco 5350 supports SS7 when bundled > > with an external solution - is there anything reasonably priced that > > will give me SS7 capability? > > > > Thanks in advance, > > Max > > > -- Max Clark http://www.clarksys.com

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[Serusers] Trying to use a remote mysql....
by Roger Lewau 09 Aug '06

09 Aug '06

I have been running SER quite successfully for a long time now. So tonight I dared to replicate the database to a remote server and changed the ser.cfg as follows. This is the two lines I changed in my ser.cfg filen #fifo_db_url="mysql://ser:heslo@localhost/ser" fifo_db_url="mysql://ser:heslo@192.168.12.140/ser" #modparam("auth_db|permissions|uri_db|usrloc|acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db|permissions|uri_db|usrloc|acc", "db_url", "mysql://ser:heslo@192.168.121.140/ser") Also updated serctl as follows: SQL_HOST=192.168.121.140 I then Stopped Ser, MySQL and restarted SER. It takes a long time then I get these line in the error log.. and ser fails to start. Aug 9 22:36:08 ns2 ser[43517]: new_connection(): Can't connect to local MySQL server through socket '/tmp/mysql.sock' (2) Aug 9 22:36:08 ns2 ser[43517]: db_init(): Could not create a connection Aug 9 22:36:08 ns2 ser[43517]: ERROR: group_db_ver: unable to open database connection Aug 9 22:36:08 ns2 ser[43517]: group:mod_init(): Error while querying table version Aug 9 22:36:08 ns2 ser[43517]: init_mod(): Error while initializing module group Then I start the old local mySQL server again and restart ser and it springs to life. Why is that? I am not pointing to the local mysql server anymore... Have I missed something in the configuration? Even more strange, it seems that all authentication and accounting gets queried in the new server as I see can see the data in the new server alse sees the client connection from SER. Any one who can help with an explanation on this behaviour? Kind regards Roger

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[Users] I can`t add new client by using openserctl add. Please help...
by Ferianto siregar 09 Aug '06

09 Aug '06

Dear all, Now, I try to test my openser configuration by using xlite. I tried to call from client to test the openser configuration that I have built working sucessfully. So, I must create a new user by using "openserctl add" command. But when i tried to do that, I got an error message. The error message said that " no rows effected". I think that this message said that I can not add a new user. For example: #openserctl add aldi aldi aldi(a)pcr.ac.id What can i do for solve this problem? I do hope anybody can give me a sugesstion.Please help me...Please Note: I use mysql for the database, and it can work goodly. ( I can show all the table that there is in openser database) Thanks with cheers, Ferianto --------------------------------- Do you Yahoo!? Next-gen email? Have it all with the all-new Yahoo! Mail Beta.

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[Users] Q:ABOUT:DELAY THE INVITE
by Alejandro Sanchez 09 Aug '06

09 Aug '06

MY scenario is the next. I have 2 video telephones that are conected to the PSTN and they has access trougth a dial up conection to the IP network and the problem is that the 2 video telephones shuld be register before one to other geretate the INVITE then i need make that when the first REGISTER arrive the sip server generate a delay for wait for the REGISTER of the other video telephone and both VT state register for the INVITE message.. Someone knows how can i do this? Thank's Alex. ___________________________________________________________ Do You Yahoo!? La mejor conexión a Internet y <b >2GB</b> extra a tu correo por $100 al mes. http://net.yahoo.com.mx

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[Serusers] Call limiting ?
by Alexandre Snarskii 09 Aug '06

09 Aug '06

Hi! As far as I know, SER does not have functionality of limiting number of calls on per-user basis. It was discussed years ago, and the reason for that was "We don't have ability to know is call still alive" (http://lists.iptel.org/pipermail/serusers/2004-November/013035.html) But, in April 2005, RFC 4028 (Session Timers in SIP) appeared, and it (at least theoretically) can be utilised to achieve knowlege of active dialogs, and, as result - on number of active calls. And this RFC looks like supported on Cisco gateways: U X.X.X.X:59104 -> X.X.X.X:5060 INVITE sip:aaaaaaa@X.X.X.X:5060 SIP/2.0. Supported: 100rel,timer,replaces. ^^^^^ Min-SE: 1800. ^^^^^^^^^^^^^^ User-Agent: Cisco-SIPGateway/IOS-12.x. Question: is this RFC/functionality planned to be supported in SER ? If not - are there any good intro on "How to write SER module by yourself" ? :)

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sr-users August 2006