sr-users September 2005

sr-users@lists.kamailio.org

231 participants
507 discussions

RE: [Serusers] Handoff to PSTN with a set prefix

by Colin Jordan

I need to send a prefix of 666# followed by full intl number without leading 0's. eg. 666#442071016211 The PSTN are connecting me to a Cisco SIP Proxy server, which uses Tech-prefixes eg 123# to separate and authenticate the calls into different domains. Can you please help me? -----Original Message----- From: Andreas Granig [mailto:andreas.granig@inode.info] Sent: Tuesday, September 20, 2005 10:50 AM To: Colin Jordan Cc: serusers(a)lists.iptel.org; serusers-request(a)lists.iptel.org Subject: Re: [Serusers] Handoff to PSTN with a set prefix Colin Jordan wrote: > # attemt handoff to PSTN > if (uri=~"^sip:9[0-9] *(a)gaco.co.za") { > log ("Forwarding to PSTN\n"); > t_relay_to_udp("217.15.11.87","5060"); > t_relay_to_tcp("217.15.11.87","5060"); > break; > }; Hm? Both UDP and TCP? Try: if (uri=~"^sip:9[0-9] *(a)gaco.co.za") { log ("Forwarding to PSTN\n"); prefix("666#"); t_relay_to_udp("217.15.11.87","5060"); break; }; I'm not sure about the "#" and if it's parsed correctly or if it's treated as start of a comment, haven't ever tried that. Andy

19 years, 2 months

[Serusers] Handoff to PSTN with a set prefix

by Colin Jordan

I need to set a fixed prefix of, 666#, to enable me to break out to the PSTN that I am sending my traffic. Will you please help me with the needed commands and where to place it? Please find below the ser.cfg file: ________________________________ # # $Id: ser.cfg,v 1.21.2.1 2003/07/30 16:46:18 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) # new insterted statement listen=66.8.91.200 listen=127.0.0.1 # hostname matching an allias will satify the condition uri-myself alias=gaco.co.za alias=66.8.91.200 /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) # this was commented port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/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 -- # database location - NEW #modparam("auth_db, "db_uri, "sql://ser:>heslo>@localhost/ser") # 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) # ------------------------- 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 record_route(); # loose-route processing if (loose_route()) { t_relay(); 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("gaco.co.za", "subscriber")) { www_challenge("gaco.co.za", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); 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(); }; # attemt handoff to PSTN if (uri=~"^sip:9[0-9] *(a)gaco.co.za") { log ("Forwarding to PSTN\n"); t_relay_to_udp("217.15.11.87","5060"); t_relay_to_tcp("217.15.11.87","5060"); break; }; } ________________________________ DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful.

19 years, 2 months

[Serusers] Enabling Radius communication between SER and Alepo RBS

by Thomas Tomter

Hello, We are working on establishing Radius-communication between SER and the billing system Alepo RBS version 4.2, but we are experiencing some trouble with this. We think the problem might be caused by the way we have defined SER as a network component in the Alepo RBS configuration file called 'NasCfgDbs' and maybe how the SER dictionary is built up. Does anyone have any experience with integration of SER with Alepo RBS? We would appreciate any kind of help or tip. Thanks in advance. Best regards, Thomas

19 years, 2 months

[Serusers] SER with asterisk as voicemail problem... please help!

by Ryan Pagquil

Hi, I configured SER to forward certain calls to asterisk for the voicemail functionality. But when I forward the SIP request to Asterisk my xlite softphone prompts "403", "Forbidden". This is my ser.cfg part where i forward the certain uri to the asterisk. if (uri=~"^sip:1[0-9]{3}@sip.philonline.com.*$") { strip(1); rewritehostport("202.84.24.47:5060"); t_relay(); break; }; Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

19 years, 2 months

[Serusers] 408 to Caller UA when CANCEL to Callee

by Tulika Pradhan

hi, i am facing the following situation. UA1 calls a user(UA2) who does not answer. the control comes to failure_route where i try another UA (UA3). but as UA3 rings, SER sends 408 Request timeout to UA1 and call gets disconnected. this is the SIP message flow. UA1 SER UA2 UA3 INVITE----------------> INVITE--------------> <----------------TRYING <----------------RINGING <------------------RINGING CANCEL--------------> <---------------------408 INVITE----------------------------------------> <---------------------487 ACK-------------------> <-----------------------OK <-------------------------------------------TRYING <--------------------------------------------RINGING (but UA already has got the busy tone) and does not hear this ringing. if 408 was not sent to UA1, then the call could have been established. what is going wrong, regards, tulika

19 years, 2 months

[Serusers] SER+MEDIAPROXY+ B2BUA

by Amit.Chawre＠relianceinfo.com

Hello I want to test a scenario with SER, MEDIAPROXY and Vovida B2BUA, but I'm having some problems. If someone has tested this before, I would like to ask you the config files to help me. When NATedUA1 calls NATedUA2, invite reaches NATedUA2. NATedUA2 sends responses back to SER (100 trying 180 ringing) but SER doesn’t forward it further to NATedUA1.I'm having problems when NatedUA2 send '100 Trying' and '180 ringing' to SER, it forwards it to B2BUA, but then there is no flow of messages. NATedUA2 ^ | 4.INVITE | 1.INVITE 2.INVITE NATedUA1 ----> SER+MEDIAPROXY ------> B2BUA <----- 3. INVITE ser.cfg is: # Example ser.cfg for mediaproxy functionality check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=72.21.40.107 port=5060 children=4 fifo="/tmp/ser_fifo" # fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/textops.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/domain.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" modparam("usrloc", "db_mode", 0) #modparam("usrloc", "db_url","mysql://ser:heslo@localhost/ser") modparam("mediaproxy", "natping_interval", 60) modparam("registrar", "nat_flag", 2) route{ if (!mf_process_maxfwd_header("10")) { if (method!="ACK") { sl_send_reply("483", "Too many hops"); }; break; }; if (msg:len >= max_len) { if (method!="ACK") { sl_send_reply("513", "Message too big"); }; break; }; if (method=="REGISTER") { if (is_from_local()) { # Mark as NAT'ed if (client_nat_test("3")) { setflag(2); force_rport(); fix_contact(); }; if (!radius_www_authorize("72.21.40.107")) { www_challenge("72.21.40.107", "0"); break; } else if (!check_to()) { sl_send_reply("403", "Username!=To not allowed"); break; }; if (!save("location")) { sl_reply_error(); }; } else { sl_send_reply("403", "This domain is not served here"); }; break; }; if (method=="INVITE") { if (!(is_from_local() || is_uri_host_local())) { sl_send_reply("403", "Relaying is forbidden"); break; }; t_on_failure("1"); } else if (method == "BYE" || method == "CANCEL") { end_media_session(); }; # Force subsequent messages to pass trough this proxy if (method == "INVITE") { record_route(); }; if (client_nat_test("3") && !search("^Record-Route:")) { # Mark as NAT'ed force_rport(); fix_contact(); }; #if (method=="INVITE") { t_on_reply("1"); #}; if (is_uri_host_local()) { if (!lookup("location")) { sl_send_reply("404", "User not found"); break; }; }; if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; if(!(src_port==5065 && src_ip==72.21.40.107)){ if (method=="INVITE" || method=="BYE") { log("amit: Forwarding to B2bua\n"); forward(72.21.40.107,5065); break; }; }; if (loose_route()) { if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; # end media session for BYE and CANCEL is done above # before entering the loose route. no need to call it here t_relay(); break; }; if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } failure_route[1] { end_media_session(); } onreply_route[1] { #if (status=~"(183)|(2[0-9][0-9])") { if (status=~"[1-2][0-9][0-9])") { if (client_nat_test("1")) { fix_contact(); }; use_media_proxy(); }; } Thanks for your help amit

19 years, 2 months

[Serusers] SER + MediaProxy or rtpproxy

by Frank Kostin

Hi All, Using Onsip SER with Mediaproxy --no-fork I can see caller/called/relayed. This is OK for local calls only. With or without mediaproxy I have audio both directions. When I call to /from another network the phone rigning but no audio. Can I relay /forward by IP address using mediaproxy and eventually redirect media streams throught a 3rd party ? Thanks, Frank --------------------------------- Yahoo! for Good Click here to donate to the Hurricane Katrina relief effort.

19 years, 2 months

[Users] request service to openser

by Loide Mara

How the UA requests a service for proxy (openser)? Is possible a direct communication between the UA and proxy? For example... How the UA requests the blacklist's service? Somebody has some suggestion? -------------------------------------------------- Loide Mara Verdes de Jesus 6º período - Ciência da Computação / UFAM Laboratório VoIP - UFAM

19 years, 2 months

[Users] Bug on pseudevariables in failure_route

by Franz Edler

Hi, I am actually testing with openser 0.10.0-dev12-tls and detected the following problem: When I use pseudo-variables $rs and $rr in a failure_route I get <null> instead of reply status and reply reason in case of a failure response. I think this is a bug. Regards Franz

19 years, 2 months

[Serusers] Instant Messaging cannot work with Windows Messenger 5.0

by kokmeng＠moij.biz

Hi, I have two PCs running Windows Messenger 5.0 and SER as registrar. I am able to get Voice, Video, App Sharing, Whiteboard, Presence (peer to peer only) working. However, I cannot get Instant Messaging to work. When I look at my Ethereal capture, I see that with 5.0, INVITE is used to start an IM session. The other client responds with TRYING, then OK. However, it also sends a BYE immediately after sending the OK. Subsequently when the originating client sends the MESSAGE, the other client responds with an Error because "Call Leg does not exist". Has anyone encountered this before? I've seen this problem being mentioned in Google searches, but no answer so far. Regards, KokMeng Loh

19 years, 2 months

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sr-users September 2005