sr-users March 2007

sr-users@lists.kamailio.org

192 participants
380 discussions

by jh＠tutpro.com

Ovidiu Sas writes: > I'm not familiar with the dbtext module and therefore I don't have a > patch for it, just the lcr patch that I already posted in the > tracker. ovidiu, but my point was that it doesn't make sense to remove DB_STRING test from one module and leave it in some other modules. i still believe that the right approach is to fix dbtext module by adding support for DB_STRING test there. so you should remove the bug tracker item on lcr and move it to dbtext module. -- juha

17 years, 8 months

[Users] name for branch 1.2.x

by Daniel-Constantin Mierla

Hello, so far, when a new major release a new branch was created with the first version in that branch: rel_1_0_0 - was for 1.0.x rel_1_1_0 - was for 1.1.x Some complained that it is not suggestive when minor release in the branch was done, like 1.0.1 or 1.1.1. So, let's try to find a way more convenient. So we will have release a.b.c, where c can be incremented several time, being like a patch update release for a.b.0. Proposed options: 1) a_b_0 - stick to current module 2) a_b_x - use x to signal that it is a changing number for same major release 3) a_b - ignore patch release number Other suggestions are welcome. Please feedback as soon as possible. The branch will be created in few hours. Cheers, Daniel

17 years, 8 months

[Serusers] Question for incoming PSTN calls

by Giovanni Chirico

Hi, I have my subscribers registered with a specific domain (ie: 0123456789(a)company.com) in subscribers table. The subscribers are geographic numbers and can made PSTN calls (outgoing calls). The question is for incoming PSTN calls: the gateway (Cisco) transfer calls to SER as <mailto:0123456789@192.168.0.1> 0123456789(a)192.168.0.1 (the IP of SER Server) and receive the error Forbidden because the subscriber is registered as <mailto:0123456789@company.com> 0123456789(a)company.com. Can I replace @192.168.0.1 with @company.com or disabile this check? Can someone help me? Thanks in advance, Giò

17 years, 8 months

[Users] RTPproxy just for inside calls

by Joao Pereira

Hello Im configuring a server with hundreds of users and Im not using RTPproxy in order to allow more calls / second. But my clients that are behind NAT need OpenSER to force the RTP to pass through the server.... how can I configure RTPproxy to be used only in the calls inside my domain? Thanks regards Joao Pereira

17 years, 8 months

[Users] cancelling forked requests + forcing TLS use

by Andrew T Gin

Hi I have two questions. 1. Receiver(a)1.1.1.1 and Receiver(a)2.2.2.2 register with ser. Client sends an INVITE to ser for Receiver. ser forks these INVITES in parallel to both Receiver(a)1.1.1.1 and Receiver(a)2.2.2.2. As both Receivers are online, they both reply with 200 OK. SER then relays BOTH 200 OK's back to the Client. My question: Shouldn't SER send CANCEL to one of them, and relay only ONE 200 OK back to the client? How would I do this? It is not the client's responsibility to send a CANCEL to the SER to relay to one of the receivers, is this correct? 2. When I have tls_verify_client = 1 tls_require_client_certificate = 1 in my cfg file, ser still accepts tcp and udp connections, despite tls_require_client_certificate=1. I did a search, and found this way: if (proto != TLS) { sl_send_reply("403", "Forbidden"); exit; }; However SER is really still listening on UDP and TCP port 5060. Is there a way to configure SER such that it only listens and acknowledges TLS connections? Thanks Andrew

17 years, 8 months

[Users] OpenSER's doxygen documentation available online

by Daniel-Constantin Mierla

Hello, ... although mainly intended for developers, it might be useful for everybody ... Doxygen-generated documentation for development version is now available for browse online at: http://www.openser.org/doxygen-devel/ It is a very good tool to check where a variable or function was defined or referenced, look at internal structures without downloading sources locally. Credits to Norman Brandinger for the config files of doxygen. Those files are now included in source tree, in 'doc/doxygen', so if you want to generate a local copy of such documentation, use: # doxygen doc/doxygen/openser-doxygen The outup is in 'doc' directory. you can tune it from doxygen config files. Cheers, Daniel

17 years, 8 months

[Users] lcr module parameters

by jh＠tutpro.com

Daniel-Constantin Mierla writes: > deadline was till today. Please no changes in the SVN only bug fixes if > found in last minute. Not other updates to C code -- docs are always > welcome to improve. ok, i'll commit the changes to trunk after 1.2 has been released. -- juha

17 years, 8 months

Re: [Users] Dispatcher and failover in 1.2.0

by jh＠tutpro.com

Daniel-Constantin Mierla writes: > > There is another hidden bomb inside the lcr module. > yes, know about that. Talked with Juha and there seems to be loat of > changes to me made in time for this release. It is his decision how far > to go. I changed to many modules, where I had good knowledge. > Unfortunately lcr is not one I played too much, so the best is to have > maintainer do it. For next release, these should be labeled as "to be > fixed". i made the changes, but was out of "office" for a few days and have not tested them. when is the deadline? -- juha

17 years, 8 months

[Users] Help needed with mediaproxy and rewritehostport configuration

by Berry Bartels

Im trying to set up a SIP and RTP proxy server to provide sip trunks to other SIP servers and PABX. The trunk to the PSTN will be provided by an other SIP server. This server will only handle signaling and rtp streams from our server. To test the configuration Im using a softphone (eyebeam 1.5), a cisco 7912 with SIP IOS, both logged in on my own SIP server, and a mobile phone as a PSTN client. Above shown in a diagram. http://praatpaal.com/images/sipdiagram.jpg The following occurs: 1. When I call from the softphone (eyebeam) the the Cisco 7912 the call is established correct. As far as I can debug the clients are forced to use the mediaproxy running on our server. This also works the other way around. 2. When I call from PSTN to a SIP client, I can only hear audio from the PSTN on the SIP client. 3. When I call from a SIP client to the PSTN, I can only hear audio from the SIP client on the PSTN client. So, my conclusion is: When a call is setup from or to the PSTN network, audio is only working from the client the call is originated. Below is the configuration from our SIP server and The Mediaserver. Please help me out here. Let me know if you have any questions or need more information Kind regards, Berry Bartels RoutIT B.V. OPENSER.CFG listen=XXX.XXX.XXX.XXX alias=" XXX.XXX.XXX.XXX " alias="voipit.nl" alias="sip01.voipit.nl" alias="sipproxy01.voipit.nl" debug=3 fork=no log_stderror=yes check_via=yes # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) sip_warning=yes syn_branch=yes server_signature=yes port=5060 children=4 fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:********@localhost/openser" # --- load modules mpath="/usr/lib/openser/modules/" loadmodule "mysql.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "domain.so" loadmodule "nathelper.so" loadmodule "mediaproxy.so" loadmodule "xlog.so" loadmodule "acc.so" # --- module params # rr # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # mediaproxy modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "natping_interval", 0) modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "sipping_from", "sip:ping@voipit.nl") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("usrloc","db_url","mysql://openser:********@localhost/openser") modparam("acc", "db_url", "mysql://openser:********@localhost/openser") modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) # request routing logic route { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: ROUTE script\n"); # sanity checks if (!mf_process_maxfwd_header("10")) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Too many hops\n"); sl_send_reply("483","Too Many Hops"); exit; } if (msg:len >= max_len) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Message too big\n"); sl_send_reply("513", "Message too big"); exit; } if(loose_route()) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Loose Route -> route[1]\n"); t_relay(); } if (!method=="INVITE") { record_route(); } lookup("aliases"); if (uri==myself) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Registration :: URI is myself\n"); if (method=="REGISTER") { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Registration\n"); # Uncomment this if you want to use digest authentication if (!www_authorize("voipit.nl", "subscriber")) { www_challenge("voipit.nl", "0"); xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Registration :: Failed\n"); exit; } xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Registration :: Success\n"); save("location"); xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Registration :: Location Saved\n"); exit; } if (is_method("INVITE")||is_method("ACK")) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Invite\n"); if (lookup("location") && from_uri=~".*(a)voipit.nl") { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Lookup Location :: Found :: Internal Call\n"); xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Internal Call -> Route[1]\n"); route(1); exit; } else if (lookup("location") && from_uri!=~".*(a)voipit.nl") { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Lookup Location :: Found :: Inbound Call\n"); xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Inbound Call -> Route[1]\n"); route(1); exit; } else { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Lookup Location :: NOT Found :: Outbound Call\n"); # sl_send_reply("404","Not found :: Outbound Call"); xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Outbound Call -> Route[2]\n"); route(2); exit; } } if (is_method("SUBSCRIBE")) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Subscribe :: Subscribed in Database\n"); exit; } append_hf("P-hint: usrloc applied\r\n"); } xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: End of script -> Route[1]\n"); route(1); } route[1] { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: route[1]\n"); if (is_method("INVITE")) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: route[1] :: INVITE\n"); t_on_reply("1"); # t_on_failure("1"); use_media_proxy(); } else if (is_method("BYE|CANCEL")) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: route[1] :: BYE OR CANCEL\n"); end_media_session(); } if (!t_relay()) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Route[1] :: T_RELAY error\n"); sl_reply_error(); end_media_session(); } else { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Route[1] :: T_RELAY\n"); } exit; } route[2] { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: route[2]\n"); if (is_method("INVITE")) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: route[2] :: INVITE\n"); rewritehostport("YYY.YYY.YYY.YYY:5060"); t_on_reply("2"); use_media_proxy(); } else if (is_method("BYE|CANCEL")) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: route[2] :: BYE OR CANCEL\n"); end_media_session(); } if (!t_relay()) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Route[2] :: T_RELAY error\n"); sl_reply_error(); end_media_session(); } else { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Route[2] :: T_RELAY\n"); } exit; } onreply_route[1] { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: onreply_route[1]\n"); if (status =~ "(183)|(2[0-9][0-9])") { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: onreply_route[1] :: Status is 183 or 200-299\n"); if (client_nat_test("1")) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: onreply_route[1] :: nat_test(1) :: TRUE\n"); fix_contact(); } else { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: onreply_route[1] :: nat_test(1) :: TRUE\n"); } use_media_proxy(); } } onreply_route[2] { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: onreply_route[2]\n"); if (client_nat_test("1")) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: onreply_route[2] :: nat_test(1) :: TRUE\n"); fix_contact(); } else { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: onreply_route[2] :: nat_test(1) :: TRUE\n"); } use_media_proxy(); } MEDIAPROXY.INI [Dispatcher] start = yes socket = /var/run/proxydispatcher.sock group = openser defaultProxy = /var/run/mediaproxy.sock [MediaProxy] start = yes socket = /var/run/mediaproxy.sock group = openser listen = none allow = Any proxyIP = XXX.XXX.XXX.XXX portRange = 60000:65000 TOS = 0xb8 idleTimeout = 60 holdTimeout = 180 forceClose = 0 [Accounting] accounting = none [Database] user = openser password = ******** host = 127.0.0.1 database = radius table = radacct [Radius] secret = secret server = localhost authport = 1812 acctport = 1813 dictionaries = /etc/radiusclient-ng/dictionary, /etc/openser/radius/dictionary, /usr/local/mediaproxy/dictionary retries = 2 timeout = 3

17 years, 8 months

[Users] Temporarily suspend or disable subscriber ???

by Daryl Sanders

Hi Everyone, Does OpenSER have any mechanism to temporarily suspend or disable a subscriber? For example, if they don't pay their bill. If not, how are other people doing this? I know I could modify their username or password in the subscriber table, but that seems like a bit of a hack. Any ideas? Thanks! Daryl

17 years, 8 months

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sr-users March 2007