sr-users July 2005

sr-users@lists.kamailio.org

199 participants
544 discussions

by fedora＠voice.dyndns.tv

Dear Example Mailbox: Just wanted to let you know you were just left a 0:13 long message (number 22) in mailbox 1234 from "8600" <8600>, on Thursday, July 14, 2005 at 07:06:13 AM so you might want to check it when you get a chance. Thanks! --Asterisk

19 years, 4 months

[Serusers] MySQL accounting SER

by GeoTel

Dear All, Loading SER no errors, but no records in my acc table, nor missed calls, .. What am I missing ? Config details: - acc.so is compiled by default with mysql as per Makefile DEFS+=-DSQL_ACC to enable SQL accounting - ser.cfg modparam("acc|uri_db", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 3) - ser -D -E Listening on udp: 10.0.0.2 [10.0.0.2]:5060 tcp: 10.0.0.2 [10.0.0.2]:5060 Aliases: tcp: localhost:5060 tcp: SIPSERver:5060 udp: localhost:5060 udp: SIPSERver:5060 SIPSERver root /root # stateless - initializing 0(0) Maxfwd module- initializing acc - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 42080 0(0) INFO: udp_init: SO_RCVBUF is finally 231808 1(3301) INFO: fifo process starting: 3301 1(3301) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... Remark, can not send any message as well 1(3301) INFO: fifo process starting: 3301 1(3301) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 5(3305) Warning: run_failure_handlers: no UAC support (0, 256) Any help ? Thanks

19 years, 4 months

[Serusers] Problem integrating ser + asterisk voicemail

by Yan Yu Lim

Hi guys, I currently have a sip proxy server (sip express router) which registers the sip phones. I need to add voice mail capability, i.e. sip express router will forward all incoming calls to Asterisk if the user does not pick up the call in 15 seconds. The voicemail recording stops when the user hangs up. However, the recording does not end if the user presses the # key, i.e. it is ignoring the user input. Similarly, when the user dials 2102 to access his voice mail, Asterisk plays the prompt, but it seems to ignore all the user input keys. Please kindly advise. Regards, YY ***************************************************** Config files ------------------------------ 1) Ser --------------------- ser.cfg (SER) --------------------- # -- 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) if (uri==myself) { if (method=="REGISTER") { # attempt handoff to PSTN if (uri=~"^sip:9[0-9]*@magnum.test.net") { ## This assumes that the caller log(1, "Forwarding to PSTN\n"); ## is registered in our realm forward(10.10.10.3, 5060); ## Our Cisco router break; }; # retrieve voicemail # if (uri=~"^sip:2[0-9]*@magnum.test.net") { log(1, "Retrieving voicemail\n"); # redirect now! rewritehostport("202.125.25.102:5061"); append_branch(); t_relay_to_udp("202.125.25.106","5061"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; timeout occurred ... now to forward to Asterisk's voicemail service if(method == "INVITE") { t_on_failure("1"); }; }; t_relay(); # leave voicemail # failure_route[1] { log(1,"Activating voicemail!!\n"); revert_uri(); # redirect now to Asterisk (on the same machine) ! rewritehostport("202.125.25.102:5061"); append_branch(); t_relay_to_udp("202.125.25.106","5061"); } -------------------- 2) Asterisk ------------ sip.conf ------------ [general] context=test port=5061 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; ip phone 1012, registered with SER [1012] type=friend username=1012 canreinvite=no context=test mailbox=1012 host=203.125.25.106 nat=no dtmfmode=info disallow=all allow=alaw allow=ulaw ----------------------- extensions.conf ------------------------- [test] ;leave voice messages exten => 1012,1,Voicemail(u1012) exten => 1012,2,Hangup ;play voice messages exten => 2012,1,VoiceMailMain,1012 exten => 2012,2,Hangup ------------------------- voicemail.conf ------------------------ [default] 1012 => 1234, YY, ylim(a)test.net

19 years, 4 months

[Serusers] SEMS ??

by Abdul Lateef

Hello, What is SEMS? Is important to include with SER? Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com ____________________________________________________ Sell on Yahoo! Auctions no fees. Bid on great items. http://auctions.yahoo.com/

19 years, 4 months

[Serusers] Linux VoIP Server LiveCD

by Seong bear

Hi all, I'm sure someone must have heard about Linux VoIP Server LiveCD http://www.wifi.com.ar/english/voip.html. By considering ROI, i planing to get one in with asterisk B2BUA solution package. I myself are really not good to build one, which could be time consume and might go wrong especially the billing part. Since the LiveCD is utilizing SER, I'm thinking to setup some asterisk servers for B2BUA and PSTN gateway purpose and integrate with it. Therefore, I would like to seek more opinions and advices from the experts. It's worth to get one or should i go for consultant help? And anyone play with it before? Thank regards kh'ng

19 years, 4 months

[Serusers] subscribe_fixup: pdomain name is registrar ( ito.)

by fedora＠voice.dyndns.tv

Jul 14 04:18:44 sip ser[29090]: db_url.len=35 Jul 14 04:18:45 sip ser[29090]: pa_domain=voice.dyndns.tv Jul 14 04:18:45 sip ser[29090]: pa_mod: use_db=1 db_url.s=mysql://ser:xxxxx@localhost/openser pa_domain=voice.dyndns.tv Jul 14 04:18:45 sip ser[29090]: pa_mod_init done Jul 14 04:18:45 sip ser[29090]: subscribe_fixup: pdomain name is registrar Jul 14 04:18:45 sip ser[29090]: INFO: udp_init: SO_RCVBUF is initially 65535 Jul 14 04:18:45 sip ser[29090]: INFO: udp_init: SO_RCVBUF is finally 131070 Jul 14 04:18:45 sip ser[29090]: INFO: udp_init: SO_RCVBUF is initially 65535 Jul 14 04:18:45 sip ser[29090]: INFO: udp_init: SO_RCVBUF is finally 131070 Jul 14 04:18:45 sip ser[29103]: INFO: fifo process starting: 29103 Jul 14 04:18:46 sip ser[29108]: MESSAGING: Offline messages dumped Jul 14 04:18:46 sip ser[29103]: SER: open_uac_fifo: fifo server up at /tmp/ser_fifo...

19 years, 4 months

[Users] Billing Stopped in Account Creation (SERWEB Like)

by Ozan Blotter

Dear List, Currently working in Billing Software for SER. The problem i've met is very interesting: >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", 1) <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< As you see above i force SER to use passwords in "password" column as plain texts. * I create some users via console by using openserctl / serctl command. * Register some ATA's to SER, no problem, auth and accounting is working perfect. * They can call each other and PSTN connectivity is fine. * I tried to code a PHP for creation/deletion/editing of users. * I managed to write values into database : "username", "domain", "password", "first_name", "last_name", "phone", "e_mail" as they come default in table, i added one more table for "address" which will be used to keep customer's address when i want to send invoices or post them something. * I can see some accounts created from PHP :) * They can register with SER, no problem. They can call to others and PSTN, no problem. Ka'Booom !!! When i look into "acc" table, i cannot see INVITEs and BYEs for customers who are created via PHP, only BYE lines written into database. I'm sure that INVITE, ACK and BYE's are written if i create a user with openserctl / serctl utility but not from PHP-created ones The difference between openserctl/serctl created ones and the PHP-created ones are : "phplib_id", "ha" and "ha1b" are empty in my examples and also "datetime_created" values are "0000-00-00 00:00:00" I do not know why caused accounting module to log only "BYE" messages and what is the solution ? btw : a very early demo for upcoming opensource billing for OpenSER / SER can be seen at http://www.intertet.net/goktar/?page=x Thanks, Ozan

19 years, 4 months

[Serusers] (no subject)

by Singh, Amit

Hi, I tried to turn on authetication in SER. I was able to add a new user to the subscriber table using serctl utility, but when I restarted the server, after adding the user, I am getting the following error: ======================================================================== ========================== SERVER LOG AND THE SERVER WONT START ======================================================================== ========================== 0(0) get_connection(): Connection not found in the pool 0(0) new_connection(): Access denied for user: 'ser@localhost' (Using password: YES) 0(0) db_init(): Could not create a connection 0(0) register_udomain(): Can not open database connection 0(0) domain_fixup(): Error while registering domain ERROR: error ffffffff while trying to fix configuration 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done 0(5165) shm_mem_destroy 0(5165) destroying the shared memory lock when I connect to the database directly using mysql, I can connect without any hassle. Now, I am not sure, whether it is the domain_fixup() which is causing problems or the error where it cannot connect to the database. But SER did start the first time I tried, and I was able to add a user. Can someone point me out, what is going wrong and needs to be fixed? ======================================================================== ========================== ADDED A USER BEFORE ======================================================================== ========================== [root@sip-ex-rtr root]# serctl add root root root(a)ser.com MySql password: ERROR 1045: Access denied for user: 'ser@localhost' (Using password: YES) introducing a new user to the database failed ERROR 1045: Access denied for user: 'ser@localhost' (Using password: YES) introducing a new user into uri table failed [root@sip-ex-rtr root]# serctl add root root root(a)ser.com MySql password: new user added new user into uri table added [root@sip-ex-rtr root]# serctl add root root root(a)ser.com MySql password: user already exists [root@sip-ex-rtr root]# serctl add root2 root2 root2(a)ser.com MySql password: [root@sip-ex-rtr root]# /usr/local/sbin/serctl: line 333: /tmp/ser_fifo: Interrupted system call ======================================================================== ========================== CONFIGURATION FILE ======================================================================== ========================== # # $Id: ser.cfg,v 1.24 2003/11/11 15:32:36 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=10 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (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" loadmodule "/usr/local/lib/ser/modules/pa.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" # ----------------- 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_url","sql://ser:ser@localhost/ser") 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) #modparam("pa", "new_tuple_on_publish", 1) #modparam("pa", "pa_domain", "orka.dhs.org") # ------------------------- 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 >= 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(); # 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") { # Uncomment this if you want to use digest authentication if (!www_authorize("ser.com", "subscriber")) { www_challenge("ser.com", "0"); break; }; save("location"); break; }; if (method=="SUBSCRIBE") { if (t_newtran()) { handle_subscription("registrar"); }; break; }; # if (method=="PUBLISH") { # if (t_newtran()) { # handle_publish("registrar"); # }; # break; # }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } Thanks & Regards, Amit Singh Co-op Engineer Sony Ericsson Mobile Communications, 7001 Development Drive, RTP Durham, NC 27709 Ph: +1-919-472-1601 _____________________________________________________________________ This e-mail has been scanned for viruses by MCI's Internet Managed Scanning Services - powered by MessageLabs. For further information visit http://www.mci.com

19 years, 4 months

[Serusers] Broken Calls not written to Database

by Ozan Blotter

Dear List, I'm making a call to PSTN, it's okay i can talk with the other party via ATA, but suddenly i unplug power from it, and in accounting module it does not write line with BYE message so i cannot understand whether the call is finished or not. only INVITE and ACK are written, no BYE :( This may be a security hole for customers, because they talk for a long time and they may unplug their units or cut power so it does not tell SER it's over. How i can prevent this, also what i need to add as a line for telling ACC module not to write calls from 833 to 833 into database, which are free SIP2SIP calls ? Btw, a Postpaid Billing System is on the way for OpenSER built in PHP, later i will need OpenSER group's help to place it onto hosting site. Shortly it will have features: * Nothing extra from package, default MySql Database which comes with OpenSER Release is being used, * All routing is done via OpenSER's ser.cfg , * Initial Rate, Initial Time, Increment Rate, Increment Time will be user variable , * Account Creation/Deletion done in PHP , * Invoice Generation and CDR will include many variables , * Rating will be in PHP, * And everything is totaly free :) If you have ideas and/or suggestion please write back to me so i can work for it too. Reqs: OpenSER & PHP & MySql & Apache Thanks, Ozan debug=3 fork=yes log_stderror=no listen=212.XXX.104.XXX port=5060 children=4 alias=212.XXX.104.XXX dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/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/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/domain.so" loadmodule "/usr/local/lib/openser/modules/acc.so" #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("auth_db", "use_domain", 1) modparam("domain", "db_mode", 1) modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/openser/sip-asymmetric-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/openser/rtp-asymmetric-clients") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) modparam("registrar", "default_expires", 60) modparam("registrar", "min_expires", 30) modparam("registrar", "nat_flag", 6) modparam("registrar", "use_domain", 1) modparam("rr", "enable_full_lr", 1) modparam("acc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) 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 and Acc section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("212.XXX.104.XXX:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); setflag(1); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # 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 (uri!=myself) { route(1); break; }; if (uri==myself) { 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(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","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 (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (uri=~"^sip:0[0-9]*@*") { rewritehost("212.XXX.59.XXX"); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; }

19 years, 4 months

[Serusers] reading avpops variables in ser.cfg

by Ramin Nikaeen

Valued Colleagues, Can anyone tell me how I can use/read the value of an avpops variable in my ser.cfg?! For example, if my intention is to obtain a new uri value from database and rewrite the RURI using this value, would the following work properly? if (avp_db_load("$ruri/username", "s:new-uri")) { rewriteuri("s:new-uri"); }; I appreciate your kind assistance in advance ramin

19 years, 4 months

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sr-users July 2005