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[Users] Please help me to fix openser.cfg (rewritehostport)
by Ronald Wiplinger 13 Oct '06

13 Oct '06

I try to rewritehostport to my Asterisk box with all starting numbers with 6. The xlog never shows anything. if (method=="INVITE") { if (uri=~"^sip:6.+@") { rewritehostport("xx.xx.xx.xx:5060"); xlog("L_ERR", "transfer to Asterisk"); route(1); exit; }; }; I believe I did not put it at the right place. Below is my entire openser.cfg. Can you please tell me, where above paragraph should be? Other parts maybe also not in the right order. bye Ronald Wiplinger # # $Id: openser.cfg,v 1.6 2006/02/15 18:23:46 bogdan_iancu Exp $ # debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) log_facility=LOG_LOCAL7 log_name="Taipei-openser" check_via=no # (cmd. line: -v) dns=no # was no (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" mpath="/usr/local/lib64/openser/modules/" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "nathelper.so" loadmodule "xlog.so" loadmodule "enum.so" loadmodule "acc.so" loadmodule "auth.so" loadmodule "auth_db.so" modparam("usrloc|auth_db","db_url","mysql://shshshsh:shhhhrw@localhost/wodeopenser") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("registrar", "sip_natping_flag", 7) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("nathelper", "rtpproxy_sock", "udp:xx.xx.xx.xx:22222") # default port 22222 #modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_from", "sip:xx.xx.xx.xx") modparam("enum", "domain_suffix", "e164.org.") modparam("enum", "tel_uri_params", ";npdi") # ------------------------- request routing logic ------------------- # main routing logic route{ xlog("L_ERR", "request method: [$rm], from URI: [$fu], To URI: [$tu]"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 4096 ) { sl_send_reply("513", "Message too big"); exit; }; # ----------------------------------------------------------------- # NAT detection # ----------------------------------------------------------------- route(2); # ----------------------------------------------------------------- # if you want to make ENUM work with numbers starting with "00", # use the following to convert "00" it into a "+" # ----------------------------------------------------------------- if (uri=~"^sip:00[1-9][0-9]*@") { strip(2); prefix("+"); }; if (uri=~"sip:\+[0-9]+@") { if(!enum_query("e164.arpa.") && !enum_query("e164.org.") && !enum_query("e164.info.") && !enum_query("enum.org.")) { xlog("L_ERR", "Invalid enum? $uri Route to PSTN"); }; xlog("L_ERR","we matched just number $tu"); strip(1); # remove + t_relay(); exit; }; # ----------------------------------------------------------------- if (!method=="REGISTER") record_route(); if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!uri==myself) { route(1); }; # ----------------------------------------------------------------- if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("taipei.example.net", "subscriber")) { www_challenge("taipei.example.net", "0"); exit; }; if (isflagset(5)) { setflag(6); }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); if (method=="INVITE") { if (uri=~"^sip:6.+@") { rewritehostport("xx.xx.xx.xx:5060"); xlog("L_ERR", "transfer to Asterisk"); route(1); exit; }; }; }; route(1); } route[1] { if (subst_uri('/(sip:.*);nat=yes/\1/')){ setflag(6); }; if (isflagset(5)||isflagset(6)) { route(3); } if (!t_relay()) { sl_reply_error(); }; exit; } route[2]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; } route[3] { if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); t_on_failure("1"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); } failure_route[1] { if (isflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } }

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[Users] acc_radius AccountingStop question
by Pletli Antal 13 Oct '06

13 Oct '06

Hi, My system uses only AccountingStop radius request for accounting. When the openser receives BYE message, a radius AccountingStop request will be sended to back-end system. This request has only one event timestamp attribute, that is the time of the BYE message received. I would like to send the start and end time of the conversation in AccountingStop radius request. How can I store the timestamp of the INVITE message of this dialog? Thanks in advance Antal

2 1

[Serusers] Contact
by Teran Jesus 13 Oct '06

13 Oct '06

Hello i have a question, i want to change the Contact address in the INVITE message, How can i do this? J Teran

2 1

[Serusers] complete billing sip/h323 solution
by sip＠itecbolivia.com 13 Oct '06

13 Oct '06

Dear Users i'm selling a complete billing sip/h323 solution,wholesale, retail, etc If interested please contact. Regards SoTelecom ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program.

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[Serusers] Fail Over Route
by Suraj Kumar 13 Oct '06

13 Oct '06

Hi All, Any one does have done the failover route using the append_Branch ? I was trying to send the calls to a different route if the primary gateway is down. Here is what I had in my config, but not sure why the uri parameter is not accepted in the append_branch function as mentione below. If I use a constant instead, it workd fine! Thanks in advance for any suggestions. Regards, Sudhir ********************************************************** failure_route[1] { log (1, "*******************Failure route ********************"); if (status=~"[5-6]0[0-9]") { rewritehostport ("207.160.xxx.xxx:5060"); append_branch(uri); # append_branch("sip:33715619000116@207.160.xxx.xxx:5060"); t_relay(); break; }; } ********************************************************** _________________________________________________________________ Try the new Live Search today! http://imagine-windowslive.com/minisites/searchlaunch/?locale=en-us&FORM=WL…

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[Serusers] add parameters to the Record-Route
by Ricardo Carvalho 13 Oct '06

13 Oct '06

I need to add a parameter to the Record-Route of some sip messages, but with 0.9.6 version of Ser I guess there aren't functions implemented to do that , doesn't it? what do you recommend me? Thanks, Ricardo.

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[Users] Checking received contact against location table
by Dimo 13 Oct '06

13 Oct '06

Hi, I want to tweak my openser setup so it allows calls (INVITES) only from registered contacts. I was wondering if there is a way to check the received contact in the invite message against the location table in mysql database, where i store the registered users. I do not want to make additional digest authorization of the Invites. I see that the registered() function in registrar module has something like that, but it makes the check on the request uri, not the contact. Also if someone can think of a better way to allow calls only from registered contacts I would greatly appreciate any tips. Thanks. Best, D.

2 1

[Users] TLS certificate :Please help..
by Ferianto siregar 13 Oct '06

13 Oct '06

Dear all, Thank you very much for your time to read this message. Thank you All, I have qustion about certificate that we build in openser in order the openser support TLS. They are CA root certificate and client/server certificate. Here is my question: 1. What is certificate file that we have got after running ./gen_rootca.sh belogs to? I mean, what is cacert.pem, 01.pem, cakey.pem belongs to? Is it belongs to server? and for what does it use? 2. What is certificate file that we have got after runnign ./gen_usercert.sh belongs to? I mean, what is user-cert.pem,user-privkey.pem, user-calist.pem belongs to? Is it belongs to server or client? and what does it use? I do hope anybody can help me and give me an answer. Please.. Thank you. Regards, Ferianto --------------------------------- Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business.

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[Users] TLS certificate :Please help..
by Ferianto siregar 13 Oct '06

13 Oct '06

Dear all, Thank you very much for your time to read this message. Thank you All, I have qustion about certificate that we build in openser in order the openser support TLS. They are CA root certificate and client/server certificate. Here is my question: 1. What is certificate file that we have got after running ./gen_rootca.sh belogs to? I mean, what is cacert.pem, 01.pem, cakey.pem belongs to? Is it belongs to server? and for what does it use? 2. What is certificate file that we have got after runnign ./gen_usercert.sh belongs to? I mean, what is user-cert.pem,user-privkey.pem, user-calist.pem belongs to? Is it belongs to server or client? and what does it use? 3. What is TLS belongs to? It belongs to TCP protocol or UDP protocol? I do hope anybody can help me and give me an answer. Please.. Thank you. Regards, Ferianto --------------------------------- Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail.

1 0

[Serusers] ACK not loose_routed
by Cesc 12 Oct '06

12 Oct '06

Hello everyone! I have my system based on ser 0.9.6. Internal calls work just fine. I am now trying to interop with another system. It has a sort of asterisk functionality, but it is not asterisk, it is a private software company product. my.phone ---------> my.ser.proxy -------------> external.proxy -------------> external phone 10.111.0.119 10.111.0.50 10.111.0.20 10.111.0.144 There is no firewall or nat in the way. The problem, i think, is that the external.proxy is buggy. I told that to the company, but who knows when they will fix this. * Look at the OK (message #9 and #10). My.ser.proxy record-routes all invites. The Record-route headers reach the external.phone, which copies them in the OK message, sends the OK to the external.proxy ... and when this forwards it to my.ser.proxy, they are gone! The OK reaches my.phone, but then it generates my problem: the ACK. It contains NO ROUTE headers and the r-uri is also simply pointing to the external.proxy. If the ACK had the ROUTE headers, my.ser.proxy would loose_route the message and voila! But as loose_route() returns false, my (maybe bad) config file gets confused and treats it like a "new" call ... so it lookup("location") of the ACK r-ruri fails, and the ACK is dropped. Should i modify the config file so that ACK, if not loose_route'd, are simply t_relay'd? * They also modify the contact field. The reason is because behind the external.proxy could be H323 or SIP phones, so they sort of want to be a termination as far as signalling is concerned. I attach the message flow, hope it helps ... Thanks! Cesc

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