sr-users September 2005

sr-users@lists.kamailio.org

231 participants
507 discussions

[Serusers] ebay to pay $4 billion for skype [OT]

by Iqbal

well I guess there's hope for all the VOIP companies/entrepreneurs on this list...if anyone makes a billion, feel free to make donation, I accept all kinds of currencies :-) good luck Iqbal

19 years, 5 months

[Serusers] Issue 5.0 of GettingStarted Now Available....

by Simon

Dear List, Issue 5.0 of the successful GettingStarted document is now available for download from http://www.onsip.org/modules/mydownloads/viewcat.php?op= <http://www.onsip.org/modules/mydownloads/viewcat.php?op=&cid=13> &cid=13 This version builds upon previous versions and as well as fixing a few bugs in the configs we have added support for call forwarding. So if you are still struggling with NAT'ed devices, or PSTN gateways, or being challenged by your Customers to add more call routing functions then this new version of GettingStarted will help. We hope you like and enjoy, as always feedback is warmly welcomed. Paul, Simon, Greger www.onsip.org <http://www.onsip.org/>

19 years, 5 months

[Serusers] Re: Re: SER + Mediaproxy sl_send_reply: I won't send a reply for ACK!!

by GT

Thanks G I'll verify all that, as a matter of fact I am internal network(s) not public but beside a FW. In MediaProxy install, it is said "must" IP public. Do I need a distinctive public IP ? Kind regards, Gtel The config will not mark non-nated devices for proxying ('serctl ul show' will not show the flag set). I suggest you verifiy that none of the EUCs are behind NAT, it should then work regardless of mediaproxy, then ensure that mediaproxy is running properly and that sessions are created (see /var/log/messages). Finally, many people fall in the trap of an iptables firewall or firewall in front of ser. g-) ---- Original Message ---- From: GT To: serusers(a)lists.iptel.org Cc: greger(a)teigre.com Sent: Monday, September 12, 2005 06:11 AM Subject: SER + Mediaproxy sl_send_reply: I won't send a reply for ACK!! > Hello, > I am trying SER with Mediaproxy 1.2.1 as per Onsip sample config. > I am running SER & Mediaproxy on the same host rtp port range default > (35000:65000) but even for non-nated devices I get no audio until I > stop completely mediaproxy > Debugging but thinking there might be something wrong with this > config, any hint would be much appreciated, > Thanks, > Gtel

19 years, 5 months

[Serusers] Can' t receive calls from other domains

by harry gaillac

Hello, I use ser-0.9.3 and mediaproxy 1.4.2 I can dial sip addresses to other domains however i can't receive calls. I 've been spending time to fix this problem without success. In fact ser don't send public ip in sdp of the callee. Thanks for help Harry $Id: ser.cfg,v 1.27 2005/03/10 14:16:25 Exp $ # # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) #memlog=5 # memory debug log level #log_facility=LOG_LOCAL0 # sets the facility used for logging (see syslog(3)) /* 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" fifo_db_url="mysql://ser:heslo@serveur1.home.net/ser" user=root group=ser fifo_user=root # owner of the ser fifo fifo_group=ser fifo_mode=0660 # fifo's permissions #disable_core=yes #disables core dumping #open_fd_limit=1024 # sets the open file descriptors limit #mhomed=yes # usefull for multihomed hosts, small performance penalty #disable_tcp=yes #tcp_accept_aliases=yes # accepts the tcp alias via option (see NEWS) listen=80.119.6.86 # # ------------------ module loading ---------------------------------- 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" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/uri_db.so" loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/mediaproxy.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/permissions.so" loadmodule "/usr/lib/ser/modules/domain.so" loadmodule "/usr/lib/ser/modules/group.so" loadmodule "/usr/lib/ser/modules/acc.so" loadmodule "/usr/lib/ser/modules/msilo.so" loadmodule "/usr/lib/ser/modules/avpops.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params-- modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) # -- autdb_params-- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", 1) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- nathelper params -- modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "rtpproxy_disable", 1) # mediaproxy params -- modparam("mediaproxy", "natping_interval", 10) modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") # -- registrar params -- modparam("registrar", "nat_flag", 4) modparam("registrar", "use_domain", 1) # -- tm params -- modparam("tm", "fr_inv_timer", 15) modparam("tm", "fr_timer", 10) # --domain params -- modparam("domain", "db_url", "mysql://ser:heslo@serveur1.home.net/ser") modparam("domain", "domain_table", "domain") modparam("domain", "db_mode", 1) # Use caching # -- permissions params -- modparam("permissions", "db_url", "mysql://ser:heslo@serveur1.home.net/ser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "default_deny_file", "/etc/ser/permissions.deny") modparam("permissions", "default_allow_file", "/etc/ser/permissions.allow") # -- acc params -- modparam("acc", "db_url", "mysql://ser:heslo@serveur1.home.net/ser") modparam("acc", "log_fmt", "miocfsp") modparam("acc", "failed_transactions", 1) modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 3) # -- msilo params -- modparam("msilo", "db_url", "mysql://ser:heslo@serveur1.home.net/ser") modparam("msilo", "check_time", 10) # -- avpos params -- modparam("avpops","avp_url","mysql://ser:heslo@serveur1.home.net/ser") modparam("avpops","avp_table","usr_preferences") # ------------------------- 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(); }; if (method=="CANCEL" || method=="BYE") { end_media_session(); }; # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { if (has_totag() && method=="INVITE" || method=="ACK") { if (client_nat_test("3")) { setflag(4); force_rport(); fix_contact(); }; use_media_proxy(); }; route(1); break; }; # if the request is for our domains use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (is_uri_host_local()) { if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; } else if (method=="MESSAGE") { route(7); break; } else if (method=="SUBSCRIBE" || method=="NOTIFY") { route(8); break; }; route(1); break; }; # Requests to others domains if (!is_uri_host_local()) { # Checking Authorization if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("407", "Proxy Authentication Required"); break; }; if (client_nat_test("3")) { setflag(4); }; if (isflagset(4)) { force_rport(); fix_contact(); use_media_proxy(); }; # Message Handler to others domains # if (method=="MESSAGE") { route(1); break; }; # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; } route[1] { ########################### # Default Message Handler # ########################### t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } route[2] { ############################ # Register Message Handler # ############################ if (client_nat_test("3")) { setflag(4); force_rport(); fix_contact(); }; if (!www_authorize("", "subscriber")) { www_challenge("", "0"); sl_send_reply("401", "Unauthorized"); break; }; if (!is_from_local()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); break; }; m_dump(); } route[3] { ############################# # CANCEL and INVITE handler # ############################# # SIP destinations to PSTN if (uri=~"^sip:[0-9]{4}@.*$" || uri=~"^sip:[0-9]{10}@.*$" || uri=~"^sip:06[0-9]{8}@.*$" || uri=~"^sip:00.*@.*$") { route(4); route(1); break; }; # SIP destinations to IPBX if (uri=~"^sip:8[0-3]@.*$") { route(5); route(1); break; } # Check AVPs for Voicemail if ((avp_db_load("$ruri","s:fw_voicemail/usr_preferences") && avp_check("fw_voicemail", "eq/i:1"))) { route(6); route(1); break; }; # accounting db flags setflag(2); setflag(3); acc_db_request("404 missed call", "missed_calls"); # Check Nated User Agents if (client_nat_test("3")) { setflag(4); }; if (isflagset(4)) { force_rport(); fix_contact(); use_media_proxy(); }; # Check ACL for Voicemail if (is_user_in("Request-URI","voicemail")) { setflag(5); }; # native SIP destinations are handled using our USRLOC DB lookup("aliases"); if (!lookup("location")) { if (isflagset(5)) { route(6); route(1); break; } else { sl_send_reply("404", "User Not Found"); break; }; }; t_on_failure("1"); route(1); break; } route[4] { ########################## # Outgoing PSTN Handler # ########################## # Checking Authorization to PSTN if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("407", "Proxy Authentication Required"); break; }; if (uri=~"^sip:[0-9]{4}@.*$") { if (!is_user_in("From", "local")) { sl_send_reply("403", "Forbidden"); break; }; } else if (uri=~"^sip:[0-9]{10}@.*$") { if (!is_user_in("From", "local")) { sl_send_reply("403", "Forbidden"); break; }; } else if (uri=~"^sip:06[0-9]{8}@.*$") { if (!is_user_in("From", "cellphone")) { sl_send_reply("403", "Forbidden"); break; }; } else if (uri=~"^sip:00.@.*$") { if (!is_user_in("From", "int")) { sl_send_reply("403", "Forbidden"); break; }; }; prefix("0"); rewritehostport("asterisk.home.net:5050"); break; } route[5] { ########################## # Asterisk Handler # ########################## # Checking Authorization to Asterisk if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("407", "Proxy Authentication Required"); break; }; if (uri=~"^sip:8[0-3]@.*$") { if (!is_user_in("From", "local")) { sl_send_reply("403", "Forbidden"); break; }; }; rewritehostport("asterisk.home.net:5050"); break; } route[6] { ########################## # Voicemail Handler # ########################## revert_uri(); # resend to voicemail with original request URI prefix("u"); rewritehostport("asterisk.home.net:5050"); break; } route[7] { ########################## # Message Handler # ########################### # Checking Authorization if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("407", "Proxy Authentication Required"); break; }; lookup("aliases"); if (!lookup("location")) { if (method == "MESSAGE") { if (!t_newtran()) { sl_reply_error(); break; }; if (m_store("0")) { t_reply("202", "Accepted for Later Delivery"); break; }; t_reply("503", "Service Unavailable"); break; }; }; route(1); } route[8] { ################################ # SUBSCRIBE/NOTIFY Handler # ################################ # Checking Authorization if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("407", "Proxy Authentication Required"); break; }; lookup("aliases"); if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } onreply_route[1] { ########################## # Statefull Handler # ########################## if (isflagset(4) || isflagset(6) && status=~"(180) || (183) || 2[0-9][0-9]") { if(search("^Content-Type: .*$")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_contact(); }; } failure_route[1] { ########################## # Voicemail Handler # ########################## if (t_check_status("486")) { revert_uri(); # resend to voicemail with original request URI prefix("b"); rewritehostport("asterisk.home.net:5050"); append_branch(); t_relay_to_udp("asterisk.home.net","5050"); break; }; if (t_check_status("408")) { revert_uri(); # resend to voicemail with original request URI prefix("u"); rewritehostport("asterisk.home.net:5050"); append_branch(); t_relay_to_udp("asterisk.home.net","5050"); break; }; } ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

19 years, 5 months

[Serusers] Failure Route

by Sebastian Kühner

Hello, I have two different failure routes. One of them is for internal calls (function: mailbox on timeout or busy) and the other one is for PSTN calls (function: failover). If a user makes a call to a Virtual Number of another client (MyUser1 --> PSTN --> MyUser2), the failure routes are "mixtured". Example: Call from 44441 to 001234567890 --> Gateway --> Gateway forwards it to 44442 After a couple of seconds, the failure_route 1 (for internal calls) goes into the failure routing with the number 001234567890... but obvially there is no client with the number 001234567890. Any hints? Sebastian

19 years, 5 months

FW: [Fwd: Re: [Serusers] ul_add flag and serctl]

by Aisling

Also, the following errors appeared on the shell where I started ser ERROR: parse_sip_msg_uri: bad uri aisling(a)x.x.x.x WARNING: do_action:error in expression ERROR: parse_uri: bad uri, state 0 parsed: <aisl> (4) / aisling(a)x.x.x.x (22) ERROR: parse_sip_msg_uri: bad uri aisling(a)x.x.x.x WARNING: do_action:error in expression Warning: sl_send_reply: I won't send a reply for ACK!! -----Original Message----- From: Aisling [mailto:ashling.odriscoll@cit.ie] Sent: 24 August 2005 13:44 To: 'serusers(a)lists.iptel.org' Cc: 'Iqbal' Subject: RE: [Fwd: Re: [Serusers] ul_add flag and serctl] Hi Iqbal, Jan, I changed my log to /tmp/fifo in my ser.cfg and the serctl.dat file. I then tried adding an alias: Serctl alias add 7890 sip:aisling@x.x.x.x Again I got the 404 flags expected error and I checked the /tmp/fifo file. It was empty. I then did /opt/ser/sbin/serctl fifo ul_add aliases aisling(a)x.x.x.x 7890 0 1.00 0 128. (It would work unless I added those 4 parameters) It said 200 Added to table and the following appeared in my new log file: :ul_add:ser_receiver_28642 aliases aisling(a)x.x.x.x 7890 However while this said added above. When I dialed 7890 to reach the aisling(a)x.x.x.x client, a 404 was sent to my phone. Many Thanks, Aisling. -----Original Message----- From: Iqbal [mailto:iqbal@gigo.co.uk] Sent: 24 August 2005 12:29 To: Aisling O'Driscoll Subject: [Fwd: Re: [Serusers] ul_add flag and serctl] could try this -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

19 years, 5 months

[Users] Question regarding redundancy/load balancing

by Dave

Hello all: I would like to install two OpenSer servers, two database servers and a few mediaproxies. I want to achieve some level of redundancy. I have the following thoughts: 1) user DNS SRV records and this way I can do some load balancing. But my question is once a UA registers with one OpenSer server, how can I ensure that all future requests for SIP signalling will go the the same OpenSer server? What if the UA again does a DNS query before sending an INVITE and goes to a different OpenSer Server? Also how can I ensure that incoming INVITE requests will come to the UA from the "other" OpenSer server? 2) How do I handle the two database servers? If each DB server is connected to one OpenSer server, how do I ensure that what gets written to one DB is also written to the second database? Or is there some other way? I did go thru old postings on SER/OpenSER but although a lot has been discussed, no clear solution has been proposed. Any other pointers or thoughts will be highly appreciated. Dave --------------------------------- Find your next car at Yahoo! Canada Autos

19 years, 5 months

[Serusers] t_check_status in failure_route

by Andreas Granig

Hi, Lately there were some discussions about t_check_status in failure_route when cancelling, where Jan (IIRC) made clear that only the lowest reply-code can be matched with t_check_status. Now I have a related problem when routing to PSTN: I use lcr-module in failure-route for routing to the next GW when 408/500/503 is received. I have noticed that I sometimes get 408 signalled from the GW (a PSTN code is mapped to it). So on one GW I mapped 408 to 480 for testing purposes to skip the fallback, the others still return 408. Now if one of the not-remapped GWs returns 408, there's the fallback to another GW. If it's the remapped one, it returns 480, but t_check_status still matches 408. I know that this special issue will be solved when I configure all my GWs to map the 408 to 480, but the question arises if this situation could come up again in other circumstances? Because if the first GW fails with 408 triggered by fr_timer, other return codes >408 from subsequent GWs can't be detected and passed back to the calling UAC without another PSTN fallback? There was a discussion about introducing a method which checks if the call is cancelled to detect 487 (don't know anymore if on the openser-lists or here), but what about other codes? Comments? Andy

19 years, 5 months

[Serusers] Media proxy idle calls.

by Vikash Khatuwala

Hi, I have setup SER 0.9.4 with Mysql and media proxy by following the instructions from the onsip getting started guide and the config file from there too. I can manage to setup the accounts and also use a hard phone (PA1688 based) and also xten to login and place calls. However I get a few problems. When I disconnect, it takes xten about 5-10seconds before it can hang up and during this time it says hanging up. Secondly after it hangs up, I can still see the call as idle status when I run sessions.py This will go away after 1min, and sometimes if I try to call back immediately after hang up it doesnt work, I have to wait for the idle call to time out. Both phones are behind NAT and so the RTP has to go through mediaproxy. When the call is active the quality is good. I havnt made any changes to ser.cfg from onsip.org other then the realm in the www_authorize and proxy_authorize. Any advise will be appreciated. Thanks and Regards, Vikash.

19 years, 5 months

[Users] pa module

by Helge Waastad

Hi, Does anyone know if the pa module is going to be updated according to the changes in the SER cvs tree? Br, Hw

19 years, 5 months

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sr-users September 2005