sr-users February 2007

sr-users@lists.kamailio.org

158 participants
285 discussions

Re: [Users] TLS and dead tcp child
by Paulo Angonese 01 Mar '07

01 Mar '07

The core, binaries and the full log are in ftp://ftp.procergs.com.br/pub/procergs/openser This is the backtrace: .... #0 0x00c63b87 in strchr () from /lib/libc.so.6 #1 0x0033dc53 in krb5_kt_resolve () from /usr/lib/libkrb5.so.3 #2 0x003cf5a0 in kssl_keytab_is_available () from /lib/libssl.so.5 #3 0x003bbc4c in ssl3_choose_cipher () from /lib/libssl.so.5 #4 0x003b73c0 in ssl3_accept () from /lib/libssl.so.5 #5 0x003c4eba in SSL_accept () from /lib/libssl.so.5 #6 0x080e1dc9 in tls_accept (c=0xb616b490) at tls/tls_server.c:236 #7 0x080e35cc in tls_fix_read_conn (c=0xb616b490) at tls/tls_server.c:872 #8 0x0809c428 in tcp_read_req (con=0xb616b490, bytes_read=0xbfad8448) at tcp_read.c:411 #9 0x0809c7a7 in handle_io (fm=0x814e1b0, idx=-1) at tcp_read.c:772 #10 0x0809dbf3 in io_wait_loop_epoll (h=0x8113020, t=Variable "t" is not available. ) at io_wait.h:722 #11 0x0809df67 in tcp_receive_loop (unix_sock=17) at tcp_read.c:893 #12 0x08093969 in tcp_init_children (chd_rank=0x810dcd4) at tcp_main.c:1745 #13 0x0806ce1e in main_loop () at main.c:1009 #14 0x0806e8cb in main (argc=1, argv=0xbfad86f4) at main.c:1477 My version of krb5-libs is krb5-libs-1.4.1-5 Please, any help? Thanks Paulo Angonese >>> >>> Paulo Angonese wrote: >>>> Ok. >>>> >>>> The core (two files was generated) and the full log (debug=9) are in >>>> ftp://ftp.procergs.com.br/pub/procergs/openser >>>> >>>> >>>> >>>> Bogdan-Andrei Iancu escreveu: >>>>> Hi Paulo, >>>>> >>>>> some suggestion :) - could you provide full log (debug) and >>>>> core+binaries (for download) ? Or backtrace? >>>>> >>>>> regards, >>>>> bogdan >>>>> >>>>> Paulo Angonese wrote: >>>>>> >>>>>> Hi, >>>>>> >>>>>> I have the same problem. >>>>>> When the SNOM ( 360 SIP 6.5.2 ) sends the REGISTER using TLS >>>>>> OpenSER crashes. >>>>>> I´m using >>>>>> OpenSer (1.1.1-tls (i386/linux)), >>>>>> openssl-0.9.7f-7.10 >>>>>> openssl-devel-0.9.7f-7.10 >>>>>> Fedora 2.6.11-1.1369_FC4 >>>>>> >>>>>> Any help? >>>>>> >>>>>> Thanks >>>>>> >>>>>> Paulo Angonese >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> [Users] CVS 31.03, TLS and dead tcp child >>>>>> Bogdan-Andrei Iancu bogdan at voice-system.ro >>>>>> Tue Oct 31 17:13:43 CET 2006 >>>>>> >>>>>> * Previous message: [Users] CVS 31.03, TLS and dead tcp child >>>>>> * Next message: [Users] Q:ABOUTE:SET UP MESSAGE FLOW >>>>>> * Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] >>>>>> >>>>>> Hi Frank, >>>>>> >>>>>> can you pack the cores + binaries (lib + exe) + sources+logs(in >>>>>> debug >>>>>> mode) and put it somewhere for download? >>>>>> I will try to take a look. >>>>>> >>>>>> regards, >>>>>> Bogdan >>>>>> >>>>>> Frank DInnocenzo wrote: >>>>>> >>>>>> > Bogdan, >>>>>> > >>>>>> > I am using openser-1.1.0-tls_src. I actually get two core files >>>>>> after >>>>>> > the crash. Can I post them somehow? >>>>>> > >>>>>> > >>>>>> > # openser -V >>>>>> > version: openser 1.1.0-tls (i386/linux) >>>>>> > flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, >>>>>> > USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, >>>>>> > FAST_LOCK-ADAPTIVE_WAIT >>>>>> > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, >>>>>> MAX_LISTEN 16, >>>>>> > MAX_URI_SIZE 1024, BUF_SIZE 65535 >>>>>> > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >>>>>> > @(#) $Id: main.c,v 1.20 2006/07/04 17:25:54 bogdan_iancu Exp $ >>>>>> > main.c compiled on 14:27:50 Oct 24 2006 with gcc 4.0.0 >>>>>> > >>>>>> > Thanks >>>>>> > Frank >>>>>> > >>>>>> > >>>>>> > */Bogdan-Andrei Iancu <bogdan at voice-system.ro>/* wrote: >>>>>> > >>>>>> > Hi Frank, >>>>>> > >>>>>> > I'm successfully using a SNOM 320 with TLS and no problem so >>>>>> far..... >>>>>> > What openser version are you using?...do you get any core file >>>>>> > after the >>>>>> > crash?? >>>>>> > >>>>>> > regards, >>>>>> > bogdan >>>>>> > >>>>>> > Frank DInnocenzo wrote: >>>>>> > >>>>>> > > Hello, >>>>>> > > >>>>>> > > I'm new to openSer and wondered if there was any >>>>>> resolution to this >>>>>> > > post??? I'm hitting this same crash with openSer (with TLS >>>>>> enabled) >>>>>> > > using a Snom phone (300 with latest Beta firmware - >>>>>> > > snom300-6.5-SIP-j.bin ). I've gone >>>>>> > > through the archives looking for help getting this setup >>>>>> > working. I've >>>>>> > > also tried the Snom Softphone with the same result. >>>>>> > > >>>>>> > > Thanks >>>>>> > > Frank >>>>>> > > >>>>>> > > >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>> > > >>>>>> > > Hi, >>>>>> > > >>>>>> > > I have tried to register to OpenSER CVS 31.03 with my SNOM >>>>>> 360 >>>>>> > > (5.3.6sw), disabled pretty all modules from OpenSER (radius, >>>>>> > xlog, auth, >>>>>> > > ...) and always the server crashes. >>>>>> > > >>>>>> > > Is there something I have missed in configuration ? >>>>>> > > >>>>>> > > Thanks a lot for help, >>>>>> > > -Mika >>>>>> > > >>>>>> > > The error message and debug info seems like this: >>>>>> > > >>>>>> > > 11(24351) tcpconn_new: new tcp connection to: XXX.XX.XX.XXX >>>>>> > > 11(24351) tcpconn_new: on port 2171, type 3 >>>>>> > > 11(24351) tls_tcpconn_init: Entered: Creating a whole new ssl >>>>>> > connection >>>>>> > > 11(24351) tls_tcpconn_init: Looking up tls domain >>>>>> > [XXX.XX.XX.XXX:5061] >>>>>> > > 11(24351) tls_tcpconn_init: Using default tls server settings >>>>>> > > 11(24351) tls_tcpconn_init: Setting in ACCEPT mode (server) >>>>>> > > 11(24351) tcpconn_add: hashes: 442, 1 >>>>>> > > 11(24351) handle_new_connect: new connection: 0xb608f6a0 24 >>>>>> > flags: 0002 >>>>>> > > 11(24351) send2child: to tcp child 0 7(24347), 0xb608f6a0 >>>>>> > > 7(24347) received n=4 con=0xb608f6a0, fd=19 >>>>>> > > 7(24347) DBG: io_watch_add(0x810e580, 19, 2, 0xb608f6a0), >>>>>> fd_no=1 >>>>>> > > 7(24347) tls_update_fd: New fd is 19 >>>>>> > > 11(24351) DBG: handle_tcp_child: dead tcp child 0 (pid >>>>>> 24347, no 7) >>>>>> > > (shutting down?) >>>>>> > > 11(24351) DBG: io_watch_del (0x810e420, 17, -1, 0x0) >>>>>> fd_no=16 called >>>>>> > > 11(24351) ERROR: receive_fd: EOF on 15 >>>>>> > > 11(24351) DBG: handle_ser_child: dead child 7, pid 24347 >>>>>> > (shutting down?) >>>>>> > > 11(24351) DBG: io_watch_del (0x810e420, 15, -1, 0x0) >>>>>> fd_no=15 called >>>>>> > > 0(24339) child process 24347 exited by a signal 11 >>>>>> > > 0(24339) core was generated >>>>>> > > 0(24339) INFO: terminating due to SIGCHLD >>>>>> > > 6(24346) INFO: signal 15 received >>>>>> > > 6(24346) Memory status (pkg): >>>>>> >>>>>> _______________________________________________ >>>>>> Users mailing list >>>>>> Users(a)openser.org >>>>>> http://openser.org/cgi-bin/mailman/listinfo/users >>>>>> >>>>> >>>>> >>>>> >>>> >>>> >>> >>> >>> >> >> >

3 2

[Serusers] enum module error ser2.0
by tzieleniewski 01 Mar '07

01 Mar '07

Hi! I was trying to utilize the enum module of ser but whenever I try to do the check for the user part of the request uri I get the following error: ser@sen:~/ser/ser-cvs/proxy$ ./ser -c -f ser.cfg 0(5401) parse error (721,14-15): unknown command, missing loadmodule? I have the enum module loaded and the line for this happens is: if (is_user_from_e164()) { I also tried is_e164() which I found in the module header file but with the same result. it this is not going to work how can I make the check by hand? for instance having such uri sip:+4822102@domain.com will somthing like this work: uri=~ "sip:+[0-9]{7}@.*" ?? Bests Tomasz

2 2

[Users] OpenSER+ Freeradius+MySQL
by Ion Minzu 01 Mar '07

01 Mar '07

Hello , I have 2 servers: 1st is OpenSER, second is FreeRadius with MySQL. Till now the authentication was making through mysql, but from now I want make authentication and accounting through radius. In mysql are 2 DB: radius and openser. -To create a new user, in which DB it will be inserted? -Till now I add new users with openserctl, but with radius how to add new users? -- Best regards, Ion Minzu, mailto:ion.minzu@cts.md

3 4

[Users] Multiple hops Accounting
by Manuel Guesdon 28 Feb '07

28 Feb '07

Hi, I try to have exploitable accounting for multiple domains and multi hops routing. I've previously tried for weeks on asterisk without success, even with adding c modules and playing with cdr fork :-( He is a case: - sip.domainA and sip.domainB point to the same Openser server. - 4(a)sip.domainA is dbaliased to 5(a)sip.domainB - 5(a)sip.domainB is dbaliased to 6@asteriskServer - 6@asteriskServer make a dial to phone2(a)sip.domainA I dial 4 on phone1 and get this INVITE routing phone1(a)sip.domainA === RURI:4@sip.domainA ==> openser openser === RURI:6@asteriskServer ==> asteriskServer asteriskServer == RURI:phone2@sip.domainA ==> openser openser ==> phone2 this works and phone2 ring, I take the call and hangup phone1 after some time. Now in acc table I find 15 records (7 INVITEs, 5 ACKs and 3 BYEs) The 3 BYE records seems correct (one from phone 1 to openser, one from openser to asteriskBox and the last from asteriskBox to phone 2 via openser). But I can't find a way to match the 7 INVITEs to the 3 BYEs, even with adding src&dst legs src_ip info and so on. >From acc records, I'd like to have make the following records report: == Call ID, src, dst, total duration, billing duration == - 1234, phone1@sip.domainA,4@sip.domainA,10,8 - 1234, 4@sip.domainA,5@sip.domainB,10,8 - 1234, 5@sip.domainB,6@asteriskBox,10,8 - 1234, 6@asteriskBox,phone2@domainA,10,8 Is it a complete dream or do you have an idea ? Manuel

2 1

[Serusers] problem in calling voicemail!!
by Meidiana Ten 28 Feb '07

28 Feb '07

Hi.. Thanx for the solution It really works well But now I face another problem Really need your help.. I configure SER for voicemail usage I'm using eyebeam as the UA. To check voicemail, I set it to call 113, and to send voicemail, it will call 114 based on the SER script. But it failed to forward to voicemail even the icon "send to voicemail" has appeared on the eyebeam. Below is my ser.cfg and ser_sems.cfg Please check it for me and tell me what's wrong. ser.cfg: debug=9 fork=yes log_stderror=yes listen=202.95.149.2 # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" fifo_mode=0666 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/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" modparam("auth_db|permissions|uri_db|usrloc","db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") 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", "/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("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("tm", "fr_timer", 10 ) modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "wt_timer", 10 ) modparam("msilo", "db_url", "mysql://ser:heslo@localhost/ser") modparam("msilo", "db_table", "silo") modparam("msilo","registrar","sip:registrar@pcr.ac.id") modparam("msilo","expire_time",259200) modparam("msilo","check_time",30) modparam("msilo","clean_period",5) alias="pcr.ac.id" route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; # ----------------------------------------------------------------- # Offline Message Store Section # ----------------------------------------------------------------- if (is_from_local()) { if (method=="REGISTER") { save("location"); log("REGISTER received -> dumping messages with MSILO\n"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; break; }; # domestic SIP destinations are handled using our USRLOC DB if(!lookup("location")) { if (! t_newtran()) { sl_reply_error(); break; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; break; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("0")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; break; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); t_relay(); break; }; # forward to current uri now; use stateful forwarding that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(5); route(1); break; }; if (uri==myself) { if (method=="ACK") { route(6); break; } else if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { break; }; log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("1")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:\ +\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("pcr.ac.id","subscriber")) { www_challenge("pcr.ac.id","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (!allow_trusted() && nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("pcr.ac.id","subscriber")) { proxy_challenge("pcr.ac.id","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (uri =~ "sip:1.*@") { log("Forwarding to 5070"); rewritehostport("localhost:5070"); t_relay_to_udp("localhost","5070"); break; } if (uri=~"^sip:9[0-9]*@") { route(4); break; }; if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { route(4); break; }; sl_send_reply("404", "User Not Found"); break; }; if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { if(isflagset(6)) { unforce_rtp_proxy(); } sl_reply_error(); }; } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehostport("202.95.149.2:5060"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; route(1); } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:\ +0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } ser_sems.cfg: # ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # ser for sems configuration # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) listen=202.95.149.2 port=5070 children=4 #check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" fifo_mode=0666 unix_sock="/tmp/ser_sock" # ------------------ module loading ---------------------------------- 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/textops.so" loadmodule "/usr/local/lib/ser/modules/avp.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" # ----------------- setting module-specific parameters --------------- alias="pcr.ac.id" # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # configure avpops db connection #modparam( "avpops", "avp_url", "mysql://ser:heslo@localhost/ser" ) #modparam( "avpops", "avp_table", "subscriber" ) #modparam( "avpops", "uuid_column", "id" ) # configure aliases, the number doesn't matter as long as there are no collisions) modparam( "avpops", "avp_aliases", "email=i:67" ) # scheme to access the database #modparam( "avpops", "db_scheme", # "email_scheme:table=subscriber;value_col=email_address;value_type=string") #modparam("tm", "pass_provisional_replies", 1) # appends for INVITE to voicemail modparam("tm", "tw_append","voicemail_headers:hdr[User-Agent];P-Email-Address=avp[$email]") # appends for dtmf per INFO modparam( "tm", "tw_append","info_append:hdr[Content-Length];hdr[subject];msg[body]") # 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 > max_len ) { sl_send_reply("513", "Message too big"); break; }; if (method != "ACK" && method != "INVITE" && method != "BYE" && method != "CANCEL" && method != "INFO" ){ log("unsupported method\n"); sl_send_reply("500","unsupported method"); 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 record_route(); # 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"); if (!t_relay()) { sl_reply_error(); } break; }; # make transaction if (!t_newtran()){ log("could not create transaction\n"); sl_send_reply("500","could not create transaction"); break; } # actively absorb ACKs if (method == "ACK") { t_relay(); break; } # pass INFO to SEMS if (method=="INFO") { if(!t_write_unix("/tmp/sems_sock","sems/info_append")){ log("could not contact sems\n"); t_reply("500","could not contact media server"); } } if (uri =~ "sip:101.*@") { if (!t_write_unix("/tmp/sems_sock","myapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:102.*@") { if (!t_write_unix("/tmp/sems_sock","myconfigurableapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:103.*@") { if (!t_write_unix("/tmp/sems_sock","myannounceapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:104.*@") { if (!t_write_unix("/tmp/sems_sock","myjukebox")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:105.*@") { if (!t_write_unix("/tmp/sems_sock","mycc")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:106.*@") { if (!t_write_unix("/tmp/sems_sock","ivr_announce")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } ############ default sems apps if (uri =~ "sip:110.*@") { if (!t_write_unix("/tmp/sems_sock","echo")){ log("could not contact echo\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:111.*@") { if (!t_write_unix("/tmp/sems_sock","announcement")){ log("could not contact announcement\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:112.*@") { if (!t_write_unix("/tmp/sems_sock","conference")){ log("could not contact conference\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:113.*@") { if (!t_write_unix("/tmp/sems_sock","mailbox")){ log("could not contact mailbox\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:114.*@") { if (!t_write_unix("/tmp/sems_sock","early_announce")){ log("could not contact early_announce\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:115.*@") { # load email avp with some email address avp_write("root@localhost","$email"); # use voicemail_headers append to pass it to sems if (!t_write_unix("/tmp/sems_sock","voicemail/voicemail_headers")){ log("could not contact voicemail\n"); t_reply("500","could not contact media server"); break; } break; } t_reply("404","Not found"); } Please tell me if you need my voicemail.conf..I'll send it to you. Thanx for the help Regards, Meidiana ____________________________________________________________________________________ We won't tell. Get more on shows you hate to love (and love to hate): Yahoo! TV's Guilty Pleasures list. http://tv.yahoo.com/collections/265

2 1

[Users] DB migration from 1.1 to 1.2
by Bogdan-Andrei Iancu 28 Feb '07

28 Feb '07

Hi, today I uploaded on SVN trunk the new logic from migrating the existing DBs from 1.1 format to 1.2. The strategy is different than in the previous version. The old "reinstall" proved to be suboptimal because of the lack of flexibility and mainly it didn't provide a smooth transition (downtime was required in order to swap the DBs, unsafe - data could be lost if error occurred during reinstall - and data could be lost because of DB changes). A new approach was chosen this time - migration. A new DB is created in parallel with the old one and data is processed and transferred into the new DB via sql commands (and nor dumps), using strict mapping between old and new columns. This way the original data is kept and complex column renaming and data processing is possible (like lcr address for example). for the moment this migration is done only for mysql, but is possible to do it for postgres also but as I'm not to much into, I will need some help from some postgres guru ;). in the mean while testing is welcomed ;) regards, bogdan

1 0

[Serusers] Call from Asterisk to SER to a foreign subscriber
by karim basraoui 28 Feb '07

28 Feb '07

---------- Forwarded message ---------- From: karim basraoui <basraouik(a)gmail.com> Date: 28 févr. 2007 20:33 Subject: Re: [Serusers] Call from Asterisk to SER to a foreign subscriber To: ram <talk2ram(a)gmail.com> I have found the thing! in Asterisk dial plan, when i'm trying to call my UA1, i should not do this: exten =>xxxx,Dial(SIP/UA1) but : exten =>xxxx,Dial(SIP/192.168.4.65/UA1) ;where 192.168.4.65 is the new host address thank you for your help ! 2007/2/24, karim basraoui <basraouik(a)gmail.com>: > > UA2 tries to call UA1 not UA3, sorry for the mistake > > 2007/2/24, karim basraoui < basraouik(a)gmail.com>: > > > > i had : > > > > UA1 (192.168.4.66:5061) > > > > }------------->Asterisk----------->SER-----------> UA3 ( > > 192.168.4.65:5061) > > UA2 ( 192.168.4.66:5062) > > > > then UA1 moved to SER domain : > > > > > > UA3 ( 192.168.4.65:5061) > > UA2 ( 192.168.4.66:5062) > > ------------->Asterisk----------->SER-----------> > > > > UA1 (192.168.4.66:5061) > > > > UA1 registers to SER and SER forwards the registration to ASTERISK, > > Asterisk is aware of the new location. > > > > UA2 tries to call UA3. It sends the invite to Asterisk and Asterisk > > replies with a TRYING message. > > after a while the call fails with an error > > > > > > 2007/2/24, ram < talk2ram(a)gmail.com>: > > > > > > then register with SER, should do the job > > > > > > UA----ASterisk----SER-----Phone > > > > > > is this correct > > > > > > ram > > > > > > > > > On 2/24/07, karim basraoui <basraouik(a)gmail.com > wrote: > > > > > > > > Hi, > > > > > > > > the call i'm trying to establish is from ASTERISK to SER. > > > > the UA i'm calling was initially in ASTERISK domain and moved to SER > > > > domain , a roaming case. > > > > > > > > 2007/2/24, ram < talk2ram(a)gmail.com>: > > > > > > > > > > Hi > > > > > > > > > > your question is not clear > > > > > > > > > > how is the call path happening from ASterisk or from SER > > > > > or both are gateway proxies ? > > > > > > > > > > ram > > > > > > > > > > > > > > > On 2/23/07, karim basraoui < basraouik(a)gmail.com > wrote: > > > > > > > > > > > Hi, > > > > > > > > > > > > I've set up a small architecture in which an ASterisk proxy is > > > > > > connected to a SER proxy in two different domains. > > > > > > > > > > > > I have sip phones connected to both of them. when an asterisk > > > > > > subscriber "travels" to SER domain it tries to register to SER. I've > > > > > > configured SER to accept foreign > > > > > > subscribers and to REGISTER them to their initial domain by the > > > > > > command "t_replicate". Till now everything works good, the UA is registered > > > > > > and Asterisk is aware > > > > > > of its new location, it displays in peers list the UA and its > > > > > > new IP address and port. > > > > > > > > > > > > The problem is when i'm trying to call the foreign UA from an > > > > > > Asterisk subscriber, it doesn't work. i have a TRYING response but no call > > > > > > establishement. > > > > > > > > > > > > how can i overcome this problem please? > > > > > > > > > > > > -- > > > > > > BASRAOUI Karim > > > > > > www.basraouik.afrikart.net > > > > > > _______________________________________________ > > > > > > Serusers mailing list > > > > > > Serusers(a)lists.iptel.org > > > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > > > > > > > > > > > > > > > -- > > > > BASRAOUI Karim > > > > www.basraouik.afrikart.net > > > > > > > > > > > > > > > > -- > > BASRAOUI Karim > > www.basraouik.afrikart.net > > > > > > -- > BASRAOUI Karim > www.basraouik.afrikart.net > -- BASRAOUI Karim www.basraouik.afrikart.net -- BASRAOUI Karim www.basraouik.afrikart.net

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[Users] New OpenSER Config Generator features
by Andreas Granig 28 Feb '07

28 Feb '07

Hi all, Three new features for sip:wizard, the OpenSER Config Generator, have been released this week: - user-aliases: You can now assign aliases to local users. This is especially useful for mapping numeric numbers to SIP accounts for PSTN inward dialing. - cli-pai: This feature allows the provisioning of arbitrary SIP URIs, which are inserted into a P-Asserted-Identity Header for outgoing calls. This for example allows the signalling of a numeric CLI to a PSTN gateway (so it’s the opposite of user-aliases). - ring-timeout: You can now dynamically set the ring-timeout (or, more specifically, the fr_inv_timer) per local callee. Visit http://www.sipwise.com/wizard to give it a try. Regards, Andreas

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Re: [Users] 3xx requests and t_relay
by Bogdan-Andrei Iancu 28 Feb '07

28 Feb '07

Hi Shane, have you followed this ex: http://www.openser.org/docs/modules/1.2.x/uac_redirect.html#AEN251 can you be more specific about the errors you see? regards, bogdan Shane Burrell wrote: > I am calling the t_relay after getting the contacts. However internally I > see in openser it makes it back to the route and then dies in loose route. > I assumed there was something I need to check to allow the new call to be > processed. > > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Wednesday, February 28, 2007 12:26 PM > To: Shane Burrell > Cc: users(a)openser.org > Subject: Re: [Users] 3xx requests and t_relay > > Hi, > > let me see if I get it right...you get a 3xx reply and try to generate > INVITE to the new contacts? if so, take a look at the uac_redirect module. > > regards, > bogdan > > Shane Burrell wrote: > >> I have this working to the point it tries to relay the call but I see >> that internally openser is not making the new call. I assume t_relay >> calls a new route but how to I make sure that all relays get >> forwarded. Some of my logic else where seems to be killing the >> rerouted call and I need any 302 routed calls to be accepted and routed. >> >> >> >> TIA, >> >> >> >> Shane >> >> >> *************************************************************** >> >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > >

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[Users] 3xx requests and t_relay
by Shane Burrell 28 Feb '07

28 Feb '07

I have this working to the point it tries to relay the call but I see that internally openser is not making the new call. I assume t_relay calls a new route but how to I make sure that all relays get forwarded. Some of my logic else where seems to be killing the rerouted call and I need any 302 routed calls to be accepted and routed. TIA, Shane ***************************************************************

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sr-users February 2007