April 2008 - sr-users - kamailio.org

[OpenSER-Users] openser devel guide
by Daniel-Constantin Mierla 07 Apr '08

07 Apr '08

Hello, I want to announce the availability of the first version for openser devel guide, online at: http://www.asipto.com/pub/openser-devel-guide/ Written together with Elena-Ramona Modroiu, the tutorial approaches the important components of openser: shared memory, locking, pseudo-variables, transformations, mi commands, statistics, db api, timer, module development a.s.o. Hopefully, the guide will help new developers to understand easier the internals of openser and contribute back new code or improvements to existing code. Has a set of examples and references to the code to be able to learn quicker how to use the APIs. The document is under polishing process towards clean docbook and pretty printing (pdf) format, then the sources will be included in the openser repository to be more accessible for improvements. It will be a continuing work-in-progress document, please report any error you find there. It is also mirrored at: http://www.openser.org/docs/openser-devel-guide/ Cheers, Daniel -- http://www.asipto.com

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[OpenSER-Users] OpenSER Summit 2008 - Post facts
by Bogdan-Andrei Iancu 07 Apr '08

07 Apr '08

Hello everybody, The most important OpenSER event, community oriented, the OpenSER Summit 2008 marked a new point in OpenSER project history. Gathering more than 60 participants from both industry and community side, OpenSER Summit brought in to the lights major project people and industry representatives who help with the evolution of the project. Major topics that were discussed: A. how can OpenSER be used as a core component in industry proofed solutions and how/where to get the necessary knowledge B. what are the development directions that needs to be taken by the project - these were moderated open discussions. All this theoretical discussions were back-end up by the OpenSER Pavilion, were several companies showed the practical part of OpenSER software, via services, products and knowledge. All the material that were presented during the Summit are now available for download: A. via the summit agenda - http://www.openser.org/mos/view/OpenSER-Summit-2008/#agenda - here, by browsing the agenda, you can download a certain presentation; B. via the pub/ section of the web site - http://www.openser.org/pub/OpenSER-Summit-2008/ - gets all the presentations from here. Pictures and video captures will be soon available on the web site ;) Best regards, Bogdan

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Re: [OpenSER-Users] CPL and Postgres problems.
by Bogdan-Andrei Iancu 07 Apr '08

07 Apr '08

So, you only changed the column type (for cpl_bin) from BLOB to BYTEA ? Regards, Bogdan Ola Karlsson wrote: > Bogdan.. > > I changed the column type in the database to bytea for cgi-bin and > stuff started working , seems the decode only happened for BLOB type > not for text. > > When debugging I got a \ instead of 0x01 så figured something was > wrong getting stuff out of the database. > > /Ola > > On Mon, Apr 7, 2008 at 10:50 AM, Bogdan-Andrei Iancu > <bogdan(a)voice-system.ro <mailto:bogdan@voice-system.ro>> wrote: > > Hello Ola, > > That is strange as the len of the script is taken from the > postgres driver (via openser API) - the returned columns must be > BLOB type. Check the types of the cpl_* columns; otherwise I > suspect a postgres driver (in openser) issue. > > Best regards, > Bogdan > > Ola Karlsson wrote: > > Hi Users! > > Have a problem with CPL scripts and postgres. > I've tried to make the simplest possible script for debugging > which resulted in the following in the 'cpl_xml' column. > > '<cpl>\012 <incoming>\012 <reject status="reject" > reason="I reject anonymous calls"/>\012 > </incoming>\012</cpl>\012\012' > > Loading it with openserctl that gets translated to this in the > cpl_bin field. > > '\001\001\000\000\000\006\002\001\000\000\000\006\026\000\002\000\000\000\002[\000\001\000\031I > reject anonymous calls\000\000' > > The reason I figure that postgres could have something to do > with this is that I've got this working before(before being > quite a while ago though. ) with MySQL. > > Turning on debugging gives this information. > > Mar 31 18:13:18 fuj /sbin/openser[601]: cpl-c B4script > Mar 31 18:13:18 fuj /sbin/openser[601]: > DBG:cpl-c:get_dest_user: trying to get user from new_uri > Mar 31 18:13:18 fuj /sbin/openser[601]: > DBG:cpl-c:get_dest_user: trying to get user from R_uri > Mar 31 18:13:18 fuj /sbin/openser[601]: > DBG:cpl-c:get_user_script: fetching script for user <olak> > Mar 31 18:13:18 fuj /sbin/openser[601]: > DBG:cpl-c:get_user_script: we got the script len=0 > Mar 31 18:13:18 fuj /sbin/openser[601]: cpl-c After Script > > OpenSER says that the length of the script is 0 whatever > script I load?. > I changed that in the CPL-C module so we get the right length > of the compiled script but then it falls over on the sanity > check for NODE_CPL = 001 , and if you skip that check it > starts complaining about unknown nodes ( in my case 92 ) ... > > So my question is if anyone uses postgres and CPL, if so can > you give me some pointers? > Would be good to know if this happens to be a bug or an > untested area in that case I have no problem fixing the > problem just don't want to burn alot of time on something > already fixed or me doing something stupid with configuration. > > Regards > /Ola Karlsson > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > >

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[OpenSER-Users] sdp_mangle_ip()
by Travis Hegner 07 Apr '08

07 Apr '08

Hello All, I have briefly looked through the bug list but couldn't find anything related to my problem. I thought I would check here first to see if perhaps I have a configuration issue. I'm currently using the latest 1.3 version downloaded with svn. When attempting to call "sdp_mangle_ip()" I end up with a malformed sip packet. Upon inspection, I realized that the packet is correct except for the "Content-Length:" header, which seems to have the new (correct) content-length of the new packet written twice into the data stream. Here is what a snippet of the packet looks like: Content-Length: 156\r\n156\r\n Obviously, a correct packet should look like: Content-Length: 156\r\n Has anyone experienced this error before? Is there some parameter I should be setting for the mangler module, or another function I should call before this one? Should I roll up my sleeves and see if I can patch it myself? Any info is greatly appreciated... Travis

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[OpenSER-Users] CPL and Postgres problems.
by Ola Karlsson 07 Apr '08

07 Apr '08

Hi Users! Have a problem with CPL scripts and postgres. I've tried to make the simplest possible script for debugging which resulted in the following in the 'cpl_xml' column. '<cpl>\012 <incoming>\012 <reject status="reject" reason="I reject anonymous calls"/>\012 </incoming>\012</cpl>\012\012' Loading it with openserctl that gets translated to this in the cpl_bin field. '\001\001\000\000\000\006\002\001\000\000\000\006\026\000\002\000\000\000\002[\000\001\000\031I reject anonymous calls\000\000' The reason I figure that postgres could have something to do with this is that I've got this working before(before being quite a while ago though. ) with MySQL. Turning on debugging gives this information. Mar 31 18:13:18 fuj /sbin/openser[601]: cpl-c B4script Mar 31 18:13:18 fuj /sbin/openser[601]: DBG:cpl-c:get_dest_user: trying to get user from new_uri Mar 31 18:13:18 fuj /sbin/openser[601]: DBG:cpl-c:get_dest_user: trying to get user from R_uri Mar 31 18:13:18 fuj /sbin/openser[601]: DBG:cpl-c:get_user_script: fetching script for user <olak> Mar 31 18:13:18 fuj /sbin/openser[601]: DBG:cpl-c:get_user_script: we got the script len=0 Mar 31 18:13:18 fuj /sbin/openser[601]: cpl-c After Script OpenSER says that the length of the script is 0 whatever script I load?. I changed that in the CPL-C module so we get the right length of the compiled script but then it falls over on the sanity check for NODE_CPL = 001 , and if you skip that check it starts complaining about unknown nodes ( in my case 92 ) ... So my question is if anyone uses postgres and CPL, if so can you give me some pointers? Would be good to know if this happens to be a bug or an untested area in that case I have no problem fixing the problem just don't want to burn alot of time on something already fixed or me doing something stupid with configuration. Regards /Ola Karlsson

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[OpenSER-Users] OpenSER on top of OpenIMSCore
by Habib, Yeasin 07 Apr '08

07 Apr '08

Hi, I want to install OpenSER server on top of OpenIMSCore. I am using the new VMWare image provided by Vladimir Romanov. I have tried to install it from the source from svn. But up to installation I was successful. But when I try to run the mysql database script it's not working. I am trying to run the script /usr/local/etc/openser/openserctlrc, but it's not working anyway. Because it doesn't seems like a shell script. If anybody installs OpenSER on top of OpenIMSCore VMWare image then please help me. If anybody has a working image of OpenSER then it will be really grateful if I have the link or tutorial of that image. I am looking forward to hearing from all of you. Best regards Yeasin Habib

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[OpenSER-Users] How to detect lack of response to SIP nat ping?
by Mike Fedyk 07 Apr '08

07 Apr '08

Hi, I'm trying to come up with some code in openser that will log when a SIP UA does not respond to a SIP NAT ping. The best I can think of so far is using the sip timers module, but it is based on the INVITE method instead OPTIONS that is the nat ping default. Does anyone have any ideas that can point me in the right direction? Thanks, Mike

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[OpenSER-Users] dlg_end_dlg with extra headers
by Alex Massover 07 Apr '08

07 Apr '08

Hi! I'm trying to use dlg_end_dlg with extra headers. As far as I understood I can send fifo command like: [root@SER openser]# openserctl fifo dlg_end_dlg 123 456 'Header: here\r\n' 400 Too few or too many arguments It doesn't seem to accept third extra-header parameter. Am I doing something wrong? -- Best Regards, Alex Massover VoIP Development Manager Jajah Inc.

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[Serusers] SIP Registration fail
by William Zhang 06 Apr '08

06 Apr '08

Hi All, I am new to SER. I might just made some simple mistakes. So be easy on me:) I appreciate any inputs. Basically I was able to install and run SER 2.0.0 RC1 and make call between SIP clients SJPhone 1.65 with a helloworld configuration. Now I am trying to add the Authentication and MySQL. Here are what I did so far and no luck to get it work:( 1. Generate the configuration files with configuration file buildsystem and the content of the file is in the bottom of this message. 2. Create the SER database in MySQL server by running the ser_mysql.sh create command 3. Set SIP_DOMAIN variable: export SIP_DOMAIN="unknown.org" 4. Add domain and user with SER_CTL 2.0 with following commands ser_ctl domain add unknown.org ser_ctl user add 777(a)unknown.org -p 1212 The ser_ctl domain show and user show return: unknown.org unknown.org --------w--c-s 777(a)unknown.org uri 777(a)unknown.org 777(a)unknown.org credentials username=777 realm=unknown.org password=1212 777(a)unknown.org attr datetime_created=2008-04-06 03:25:43 I believe this should set the user and domain correctly to the database. But I also got some weired output with other command like ser_ctl ps, ser_ctl version. Same error: ser_ctl: (97, 'Address family not supported by protocol'): error'> And sercmd also give me the follow errors:( ERROR: connect_unix_sock: connect(/tmp/ser_ctl): Connection refused [111] 5. Run the ser with error to console and no fork. 6. Configure the SJPhone with above registration information. The client can not register and got 401 Unauthorized response. Below are the SIP packets contents: 2008-04-06 04:49:29.843 UDP LOCAL->192.168.0.182:5060 REGISTER sip:unknown.org SIP/2.0 Via: SIP/2.0/UDP 192.168.0.179;branch=z9hG4bKc0a800b30000004947f8565900000d5f00000019;rport From: "unknown" <sip:777@unknown.org>;tag=4ff22665b92 To: <sip:777@unknown.org> Contact: <sip:777@192.168.0.179> Call-ID: C69D49CA6580482CAFB5D305213ACE0D0xc0a800b3 CSeq: 18 REGISTER Max-Forwards: 70 User-Agent: SJphone/1.65.377a (SJ Labs) Content-Length: 0 21:49:29 SIP.Network DEBUG 2008-04-06 04:49:29.843 UDP 192.168.0.182:5060->LOCAL SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.179;branch=z9hG4bKc0a800b30000004947f8565900000d5f00000019;rport=5060 From: "unknown" <sip:777@unknown.org>;tag=4ff22665b92 To: <sip:777@unknown.org>;tag=3594b9925f930dcc8b38a6cda152d6b5.f97a Call-ID: C69D49CA6580482CAFB5D305213ACE0D0xc0a800b3 CSeq: 18 REGISTER Server: Sip EXpress router (2.0.0-rc1 (i386/linux)) Content-Length: 0 Warning: 392 192.168.0.182:5060 "Noisy feedback tells: pid=14897 req_src_ip=192.168.0.179 req_src_port=5060 in_uri=sip:unknown.org out_uri=sip:unknown.org via_cnt==1" The contents of the ser.cfg file: # Debug level on logging debug=3 # memory debug log level memdbg=10 # memory statistics log level memlog=10 # Fork and create children (no=run single process for debugging purposes) fork=no # Log to stderr (only use this of fork=no) log_stderror=yes # Log to the defined syslog facility log_facility=LOG_LOCAL0 # IP address to listen to listen=192.168.0.182 # Port to listen on port=5060 # Aliases used by uri!=myself checks to identity locally destined messages alias=mytestser.org # Number of children to fork per port per protocol children=4 # Is this host multihomed? mhomed=no # Should dns resolving be used? dns=no # Should reverse dns resolving be used? rev_dns=no # Should reverse dns resolving be used? disable_core_dump=yes # *************Your local directives ***************** # *** Common # *** Specific # Load the stateless transaction module (sl_reply etc) loadmodule "/usr/local/lib/ser/modules/sl.so" # Load the transaction stateful module (t_relay() etc) loadmodule "/usr/local/lib/ser/modules/tm.so" # Load the record routing module. Needed for making sure messages go through our server loadmodule "/usr/local/lib/ser/modules/rr.so" # Load maxfwd for stopping loops loadmodule "/usr/local/lib/ser/modules/maxfwd.so" # Load the user location module to store and retrieve user locations loadmodule "/usr/local/lib/ser/modules/usrloc.so" # Load the registrar module to handle registrations loadmodule "/usr/local/lib/ser/modules/registrar.so" # Load the xlog module for advanced logging with parameters loadmodule "/usr/local/lib/ser/modules/xlog.so" # Load the ctl module for the control socket interface. loadmodule "/usr/local/lib/ser/modules/ctl.so" # Load the gflags, global flags module. loadmodule "/usr/local/lib/ser/modules/gflags.so" # Load textops used for searching/selecting text in messages and manipulating headers. loadmodule "/usr/local/lib/ser/modules/textops.so" # Load the sanity check module to do integrity checks on incoming messages. loadmodule "/usr/local/lib/ser/modules/sanity.so" # Load the mysql connector module. loadmodule "/usr/local/lib/ser/modules/mysql.so" # Load the auth module with all the core authentication functions. loadmodule "/usr/local/lib/ser/modules/auth.so" # Load the auth module with the database functions. loadmodule "/usr/local/lib/ser/modules/auth_db.so" # The uri_db module uses database to check the uri. loadmodule "/usr/local/lib/ser/modules/uri_db.so" # The avp_db module allows loading of user attributes from database. loadmodule "/usr/local/lib/ser/modules/avp_db.so" loadmodule "/usr/local/lib/ser/modules/avp.so" # Load domain module used to verify the local domains. loadmodule "/usr/local/lib/ser/modules/domain.so" # *** Your loaded modules # Flags set be various modules # We set database mode to 0 = do not save to database backend (keep in memory), 1 = write-through to database modparam("usrloc", "db_mode", 1) # ctl module params # by default ctl listens on unixs:/tmp/ser_ctl if no other address is # specified in modparams; this is also the default for sercmd modparam("ctl", "binrpc", "unixs:/tmp/ser_ctl") # listen on the "standard" fifo for backward compatibility modparam("ctl", "fifo", "fifo:/tmp/ser_fifo") # We turn on full lr (adds lr=on as uri parameter, not lr). # This should have higher compatibility with some user agents. modparam("rr", "enable_full_lr", 1) # Encrypt dialog cookies with this secret modparam("rr", "cookie_secret", "ChangeThisIfYouStoreSecretStuffInCookie") # The authentication related modules and which database to use. modparam("auth_db|usrloc|uri_db", "db_url", "mysql://ser:heslo@localhost/ser") # We can store some of our important attributes (inside the messages) avpflags dialog_cookie; # ************* Your local module parameters ***************** # *** Common # *** Specific # Main route, executed for all messages route { # ------------------------------------------------------------------------ # Sanity Check Section # ------------------------------------------------------------------------ sanity_check(); # Check whether we might be in a loop. if (!mf_process_maxfwd_header("10")) { sl_reply("483", "Too Many Hops"); exit; } if (msg:len > max_len) { sl_reply("513", "Message Overflow"); exit; } # Log all incoming messages to see start of processing if(is_gflag("29")){ xlog("L_NOTICE","INCOMING - %rm\nFrom: %is(%fu)\nTo: %ru\nCall-ID: %ci\n"); } # ------------------------------------------------------------------------ # Handling messages with Route headers (aka loose routing) # ------------------------------------------------------------------------ if (loose_route()) { if(is_gflag("31")){ xlog("L_NOTICE"," loose routed - %rm\nFrom: %is(%fu)\nTo: %ru\nCall-ID: %ci\n"); } route(RELAY); exit; } # ------------------------------------------------------------------------ # Record Route Section # ------------------------------------------------------------------------ else if (method!="REGISTER") { record_route(); }; # If the ACK has more hops before reaching the user agent, it will have a Route header # and be forwarded above. If next hop is a user agent, the Request URI will be the ip:port # found in Contact of the original OK from the user agent, but will not be loose routed. # Just relay the ACK to the user agent. if (method=="ACK") { if(is_gflag("31")){ xlog("L_NOTICE"," ACK relayed using ruri - %rm\nFrom: %is(%fu)\nTo: %ru\nCall-ID: %ci\n"); } route(RELAY); exit; } # check if the caller is from a local domain (f=from, d=domain, $f will be set) lookup_domain("$fd", "@from.uri.host"); # check if the callee is at a local domain (t=to, d=domain, $t will be set) # If we have no To tag (dialog creating messages), we use request uri, but # but if this is an in-dialog message (like BYE with last hop before UA here) # we use the To header. if((a)to.tag=="") { lookup_domain("$td", "@ruri.host"); } else { lookup_domain("$td", "@to.uri.host"); } # we dont know the domain of the caller and also not # the domain of the callee -> someone uses our proxy as # a relay if (!$t.did && !$f.did) { sl_reply("403", "Relaying Forbidden"); exit; } # ------------------------------------------------------------------------ # Message Type Processing Section # NOTE!!! It is assumed that all messages reaching this point are destined # for the local server (REGISTER) or for UAs registered locally. (see nonlocal.m4) # ------------------------------------------------------------------------ if (method=="REGISTER") { route(REGISTER); exit; } # Authenticate route(AUTHENTICATE); # All local, non-ACKs, non-REGISTER are assumed to be part of a potential # new dialog (standalone MESSAGE and NOTIFYs without dialog-creation will ex. also reach here) route(NEWDIALOG); # By now, the request uri should be correct, relay if(is_gflag("29")){ xlog("L_NOTICE","No route matched for the request\nCall-Id: %ci\n"); }; sl_reply("404", "No route matched"); } # end main route route[RELAY] { # Log all outgoing messages if(is_gflag("29")){ xlog("L_NOTICE","OUTGOING - %rm\nFrom: %is(%fu)\nTo: %ru\nCall-ID: %ci\n"); } # ------------------------------------------------------------------------ # Default Message Handler # ------------------------------------------------------------------------ # t_relay will act based on the current request URI, as well as any # overruling received parameters (i.e. for NATed user agents) # UDP to TCP and vice versa works as well. if (!t_relay()) { sl_reply_error(); }; exit; } route[REGISTER] { # ------------------------------------------------------------------------ # REGISTER Message Handler # ------------------------------------------------------------------------ # From lookup_domain, $t.did should be set if we found the domain as ours. if (!$t.did) { if(is_gflag("29")){ xlog("L_NOTICE","%rm From <%fu> Attempted REGISTER from non-local domain\nCall-ID: %ci\n"); }; sl_reply("403", "Attempting to register with unknown domain"); exit; } # We check that Authorization header in the message contains valid credentials found in the # table with the same name. if (!www_authenticate("$fd.digest_realm", "credentials")) { # $? represents the return error code from www_authenticate() if ($? == -2) { if(is_gflag("31")){ xlog("L_NOTICE"," REGISTER - %rm From <%fu> 500 Internal Server Error\nCall-ID: %ci\n"); }; sl_reply("500", "Internal Server Error"); } else if ($? == -3) { if(is_gflag("31")){ xlog("L_NOTICE"," REGISTER - %rm From <%fu> 400 Bad Request\n"); }; sl_reply("400", "Bad Request"); } else { # After www_authenticate(), $digest_challenge should be set if no credentials were found. # We append the challenge to reply and send 401 Unauthorized. if ($digest_challenge) { append_to_reply("%$digest_challenge"); } sl_reply("401", "Unauthorized"); } # We didn't have a valid set of credentials, so we exit. exit; } # www_authenticate set $uid, so let's store it. $authuid=$uid; # Check if the authenticated user is the same as the target user. # We use the uri in the To field to lookup the user's uid. If none is found, the To user is unknown. if (!lookup_user("$tu.uid", "@to.uri")) { sl_reply("404", "Trying to register unknon user (To)"); exit; } # $f.uid set by www_authenticate() should match the uid we should looked up if ($authuid != $tu.uid) { sl_reply("403", "Authentication and To-Header mismatch"); exit; } # We use the uri in From to look up the uid. if (!lookup_user("$fu.uid", "@from.uri")) { sl_reply("404", "Expect a known user in From"); exit; } # The looked up $fu.uid is checked against the Authentication user if ($authuid != $fu.uid) { sl_reply("403", "Authentication and From-Header mismatch"); exit; } if (!save_contacts("location")) { sl_reply("400", "Invalid REGISTER Request"); exit; }; exit; } # End route[REGISTER] route[AUTHENTICATE] { # RFC3261 says you are not allowed to challenge these methods if (method=="CANCEL" || method=="ACK") { return; } # We check that Authorization header in the message contains valid credentials found in the # table with the same name. if (!proxy_authenticate("$fd.digest_realm", "credentials")) { # $? represents the return error code from proxy_authenticate() if ($? == -2) { if(is_gflag("31")){ xlog("L_NOTICE"," AUTHENTICATE - %rm From <%fu> 500 Internal Server Error\nCall-ID: %ci\n"); }; sl_reply("500", "Internal Server Error"); } else if ($? == -3) { if(is_gflag("31")){ xlog("L_NOTICE"," AUTHENTICATE - %rm From <%fu> 400 Bad Request\n"); }; sl_reply("400", "Bad Request"); } else { # After www_authenticate(), $digest_challenge should be set if no credentials were found. # We append the challenge to reply and send 407 Proxy Authentication Required. if ($digest_challenge) { append_to_reply("%$digest_challenge"); } sl_reply("407", "Proxy Authentication Required"); } # We didn't have a valid set of credentials, so we exit. exit; } # proxy_authenticate set $uid, so let's store it. $authuid=$uid; # Use the uri in the From header and look up the uid. If not found, delete the uri attribute. if (!lookup_user("$fu.uid", "@from.uri")) { del_attr("$uid"); } # Check the found uid against the authenticated uid. if ($fu.uid != $authuid) { sl_reply("403", "Fake Identity"); exit; } # load the user attributes (preferences) of the caller. $fu.yourattr should now be set if set in the # user preferences table. load_attrs("$fu", "$f.uid"); } route[NEWDIALOG] { # If we don't know the user, it is not for a local user. if (!lookup_user("$tu.uid", "@ruri")) { # User is not known and request uri does not match PSTN dialplan. return; } if (!lookup_contacts("location")) { if(is_gflag("30")){ xlog("L_NOTICE"," NEWDIALOG - 480 User Offline\nCall-Id: %ci\n"); }; sl_reply("480", "User Offline"); exit; } # Relay it to the location(s) of the found user route(RELAY); exit; } # End route[NEWDIALOG]

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[OpenSER-Users] Res: Openser with freeradius not accounting BYE method
by Flavio Goncalves 05 Apr '08

05 Apr '08

Set the flag 2 and 3 in the loose route for BYE requests. if (loose_route()) { if (is_method("BYE")) { setflag(2); setflag(3); }; }; ----- Mensagem original ---- De: MagoDaViola <magodaviola(a)ig.com.br> Para: users(a)lists.openser.org Enviadas: Sábado, 5 de Abril de 2008 16:50:56 Assunto: [OpenSER-Users] Openser with freeradius not accounting BYE method I´m has a problem that my openser sometimes not accounting bye method and this cause a call without pay. See the openser.cfg: # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=4 # debug level (cmd line: -dddddddddd) ##-- # # Uncomment these lines to daemonize fork=yes log_stderror=no # (cmd line: -E) ##-- # # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes ##-- alias=sip.processavoip.net listen=201.32.70.20 #syn_branch=1 memlog=3 sip_warning=1 server_signature=yes reply_to_via=0 mhomed=0 disable_dns_blacklist=yes unix_sock="/tmp/openser.sock" check_via=0 # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 mpath="/usr/local/lib/openser/modules/" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database 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 "auth.so" loadmodule "auth_radius.so" loadmodule "uri.so" loadmodule "domain.so" loadmodule "permissions.so" loadmodule "mediaproxy.so" loadmodule "nathelper.so" loadmodule "acc.so" loadmodule "xlog.so" loadmodule "dbtext.so" loadmodule "uri_db.so" loadmodule "mi_fifo.so" loadmodule "group.so" # ----------------- setting module-specific parameters --------------- # -- mi-fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 6) modparam("usrloc", "domain_column", "fk_domains") modparam("usrloc", "user_column", "fk_accounts") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) ## -- database params -- modparam("acc|usrloc|group|domain|uri_db|permissions", "db_url", "mysql://root:master@localhost/openser") # -- radius params -- modparam("acc|auth_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # -- radius permissions -- modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") # -- group params -- modparam("group", "table", "accounts_groups") modparam("group", "user_column", "fk_accounts") modparam("group", "domain_column", "fk_domains") modparam("group", "group_column", "pk_accounts_groups") modparam("group", "use_domain", 0) ## -- acc params -- modparam("acc", "radius_missed_flag", 3) modparam("acc", "radius_flag", 2) #modparam("acc", "radius_extra", "via=$hdr(Via[*]); email=$avp(s:email)") modparam("acc", "log_level", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") modparam("acc", "radius_extra", "User-Name=$fU;Suffix=$fd;Sip-Translated-Request-URI=$ru") modparam("acc", "service_type", 15) modparam("acc", "failed_transaction_flag", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_flag", 1) ## -- acc params without radius -- #modparam("acc", "db_missed_flag", 3) #modparam("acc", "db_flag", 2) #modparam("acc", "db_table_acc", "call_history") #modparam("acc", "db_table_missed_calls", "call_history_missed") # -- uri_db params -- modparam("uri_db", "subscriber_table", "accounts") modparam("uri_db", "subscriber_domain_column", "fk_domains") modparam("uri_db", "subscriber_user_column", "pk_accounts") # -- domain params -- modparam("domain", "domain_table", "domains") modparam("domain", "domain_col", "pk_domains") # -- nathelper, mediaproxy nd registrar params -- modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@openser.org") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("registrar|nathelper", "received_avp", "$avp(i:42)") # to configure nat at openser 1.2, 1.3 or 1.4 modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "natping_interval", 30) #modparam("mediaproxy", "sip_asymmetrics", "/etc/openser/sip-clients") #modparam("mediaproxy", "rtp_asymmetrics", "/etc/openser/rtp-clients") #modparam("registrar", "received_avp", "$avp(s:rcv)") # to configure nat at openser 1.2, 1.3 or 1.4 # -- 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 ) #Tempo maximo do ring antes de atender #modparam("tm", "fr_timer", 1) #modparam("tm", "wt_timer", 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","Muitos Hops"); log(1, "486: Muitos Hops\n"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message to big"); log(1, "513: Message too bi \n"); exit; }; ##-- # Eh feita a gravacao de todas as mensagens para se certificar que todas as mensagens # passem pelo proxy o que eh importante particularmente se as entidades estao usando # um protocolo de transporte diferente ##-- if (method!="REGISTER") { record_route(); }; route(6); ##-- # Secao de finalizacao da chamada ##-- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; ##-- # mensagens subsequentes dentro de um dialogo deveriam tomar um caminho pre-determinado # pela gravacao das rotas (record route) ##-- if (loose_route()) { # marca a logica de roteamento no pedido append_hf("P-hint: rr_enforced\r\n"); if (has_totag() && (method=="INVITE" || method=="ACK")) { xlog("L_ERR", "Main Route: Fixing SDP to $fu $rm (loose route)\n"); if (client_nat_test("3") || isflagset(5)) { setbflag(6); use_media_proxy(); }; }; xlog("L_ERR", "Main Route: Loose Route relaying\n"); route(1); exit; }; if (!uri==myself) { # marca a logica de roteamento no pedido append_hf("P-hint: rr_enforced\r\n"); xlog("L_ERR", "Main Route: Relaing mesage to proxy destination \n"); route(4); route(1); }; ##-- # Verifica baseado na tabela de dominios se a parte host da R-URI comtem um dos dominios # quais o proxy eh responsavel ##-- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbinden..."); } exit; }; ##-- # A rotina abixo eh chamada para marcar o registro com NAT ##-- if (method=="CANCEL") { route(1); exit; } else if (method=="REGISTER") { route(2); exit; } else if (method=="INVITE") { route(3); exit; }; ##-- # Verifica se existem aliases para este endereco e se eles nao pertencem # a teste dominio encaminhamos para frente ##-- lookup("aliases"); if (uri!=myself) { xlog("L_ERR", "Main Route: aliases relayin to other domain proxy\n"); route(4); route(1); exit; }; ##-- # Verifica se encontra o usuario na base de localizacao ##-- if (!lookup("location")) { sl_send_reply("404", "Usuario nao encontrado"); exit; }; route(1); exit; } route[1] { ##-- # Envie isto agora; use "statefull fowarding" pois funciona de forma confiavel de UDP para TCP # Ativa a contabilizacao das chamadas ##-- xlog("L_ERR", "Route[1]: Relaying message\r\n"); setflag(2); ##-- # Ativa a contabilizacao das chamadas perdidas ##-- setflag(3); t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { ##-- # Tratamento padrao para as mensagens de Register ##-- sl_send_reply("100", "Registrando...."); xlog("L_ERR", "Route[2]: Trying Auth: $fu $si\n"); if (isflagset(5) && client_nat_test("7")) { xlog("L_ERR", "Route[2]: Client behind NAT - flag(6)\n"); setbflag(6); fix_nated_register(); force_rport(); # se quiser usar natping options descomente a linha abaixo # setbflag(7); }; if (!radius_www_authorize("")) { www_challenge("", "0"); xlog("L_ERR", "Route[2]: Client Unauthorized: $fu $si\r\n"); sl_send_reply("401", "Unauthorized"); exit; }; if (!check_to()) { xlog("L_ERR", "Route[2]: Client Unauthorized: $fu $si\r\n"); sl_send_reply("401", "Unauthorized"); exit; } else { xlog("L_ERR", "Route[2]: Authenticate: $fu $si\r\n"); }; consume_credentials(); if (!save("location")) { xlog("L_ERR", "Route[2]: Can't save register location\r\n"); sl_reply_error(); }; } route[3] { ##-- # Tratamento padrao para as mensagens de INVITE ##-- xlog("L_ERR", "Route[3]: INVITE: $fu -> $tu $ci\r\n"); # setflag(1); if (!allow_trusted()) { if (!radius_proxy_authorize("")) { proxy_challenge("", "0"); # Realm will be autogenerated exit; } else if (!check_from()) { sl_send_reply("403", "Usue from=ID"); xlog("L_ERR", "Route[3]: Error 403 $fu -> $tu\r\n"); exit; }; } consume_credentials(); ##-- # Verifica se existem aliases para este endereco e se eles nao pertencem # a teste dominio encaminhamos para frente ##-- lookup("aliases"); if (uri!=myself) { xlog("L_ERR", "Main Route: aliases relayin to other domain proxy\n"); route(4); route(1); }; ##-- # VoIP Local ##-- if (uri=~"sip:[1-9][0-9]+@.*") { xlog("L_NOTICE", "OpenSER: Ligacao VoIP local: $fU -> $tu\n"); } else { ##-- # Nacional ##-- if (uri=~"sip:0[1-9][0-9]+@.*") { xlog("L_NOTICE", "OpenSER: Ligacao Nacional: $fU -> $tu\n"); } else { ##-- # Internacional ##-- if (uri=~"sip:00[1-9][0-9]+@.*") { xlog("L_NOTICE", "OpenSER: Ligacao Interacional: $fU -> $tu\n"); }; }; route(5); exit; }; if (!lookup("location")) { append_hf("P-hint: usuario nao encontrado\r\n"); sl_send_reply("404", "Usuario nao encontrado"); exit; }; append_hf("P_hint: usrloc applied\r\n"); t_on_failure("1"); route(4); route(1); } route[4] { ##-- # Secao de travessia do nat ##-- if (isflagset(5)) { setbflag(6); }; if (isflagset(5) || isbflagset(6) || isbflagset(7)) { use_media_proxy(); xlog("L_ERR", "Rota[4]: Travessia de NAT mediaproxy\n"); }; } route[5] { ##-- # Encaminhamento para a rede publica (PSTN) ##-- xlog("L_ERR", "Route[5]: Relaying to PSTN\n"); rewritehostport("201.32.70.21"); route(4); route(1); } route[6] { force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; } failure_route[1] { if (t_check_status("487")) { exit; }; if (isbflagset(6) || isflagset(5)) { end_media_session(); }; onreply_route[1] { ##-- # Bloco de tratamento de respostas ##-- if ((isflagset(5) || isbflagset(6)) && status=~"(180)|(183)|(2[0-9][0-9])") { use_media_proxy(); # search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); } if (isbflagset(6)) { fix_nated_contact(); } } Alcindo Schleder Sistema Processa Acessoria e Comunicações Ltda. MSN: a_schleder(a)hotmail.com Skipe: alcindo_schleder Demétrio P. dos Santos, 705 - 95670-000 - Gramado-RS-Brasil Celular+ 55 54 9966 7591 alcindo(a)processa.org Fone+55 54 3286-1738comercial(a)processa.org -----Anexo incorporado----- _______________________________________________ Users mailing list Users(a)lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users

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[OpenSER-Users] Openser with freeradius not accounting BYE method
by MagoDaViola 05 Apr '08

05 Apr '08

Untitled Document I´m has a problem that my openser sometimes not accounting bye method and this cause a call without pay. See the openser.cfg: # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=4 # debug level (cmd line: -dddddddddd) ##-- # # Uncomment these lines to daemonize fork=yes log_stderror=no # (cmd line: -E) ##-- # # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes ##-- alias=sip.processavoip.net listen=201.32.70.20 #syn_branch=1 memlog=3 sip_warning=1 server_signature=yes reply_to_via=0 mhomed=0 disable_dns_blacklist=yes unix_sock="/tmp/openser.sock" check_via=0 # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 mpath="/usr/local/lib/openser/modules/" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database 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 "auth.so" loadmodule "auth_radius.so" loadmodule "uri.so" loadmodule "domain.so" loadmodule "permissions.so" loadmodule "mediaproxy.so" loadmodule "nathelper.so" loadmodule "acc.so" loadmodule "xlog.so" loadmodule "dbtext.so" loadmodule "uri_db.so" loadmodule "mi_fifo.so" loadmodule "group.so" # ----------------- setting module-specific parameters --------------- # -- mi-fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 6) modparam("usrloc", "domain_column", "fk_domains") modparam("usrloc", "user_column", "fk_accounts") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) ## -- database params -- modparam("acc|usrloc|group|domain|uri_db|permissions", "db_url", "mysql://root:master@localhost/openser") # -- radius params -- modparam("acc|auth_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # -- radius permissions -- modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") # -- group params -- modparam("group", "table", "accounts_groups") modparam("group", "user_column", "fk_accounts") modparam("group", "domain_column", "fk_domains") modparam("group", "group_column", "pk_accounts_groups") modparam("group", "use_domain", 0) ## -- acc params -- modparam("acc", "radius_missed_flag", 3) modparam("acc", "radius_flag", 2) #modparam("acc", "radius_extra", "via=$hdr(Via[*]); email=$avp(s:email)") modparam("acc", "log_level", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") modparam("acc", "radius_extra", "User-Name=$fU;Suffix=$fd;Sip-Translated-Request-URI=$ru") modparam("acc", "service_type", 15) modparam("acc", "failed_transaction_flag", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_flag", 1) ## -- acc params without radius -- #modparam("acc", "db_missed_flag", 3) #modparam("acc", "db_flag", 2) #modparam("acc", "db_table_acc", "call_history") #modparam("acc", "db_table_missed_calls", "call_history_missed") # -- uri_db params -- modparam("uri_db", "subscriber_table", "accounts") modparam("uri_db", "subscriber_domain_column", "fk_domains") modparam("uri_db", "subscriber_user_column", "pk_accounts") # -- domain params -- modparam("domain", "domain_table", "domains") modparam("domain", "domain_col", "pk_domains") # -- nathelper, mediaproxy nd registrar params -- modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@openser.org") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("registrar|nathelper", "received_avp", "$avp(i:42)") # to configure nat at openser 1.2, 1.3 or 1.4 modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "natping_interval", 30) #modparam("mediaproxy", "sip_asymmetrics", "/etc/openser/sip-clients") #modparam("mediaproxy", "rtp_asymmetrics", "/etc/openser/rtp-clients") #modparam("registrar", "received_avp", "$avp(s:rcv)") # to configure nat at openser 1.2, 1.3 or 1.4 # -- 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 ) #Tempo maximo do ring antes de atender #modparam("tm", "fr_timer", 1) #modparam("tm", "wt_timer", 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","Muitos Hops"); log(1, "486: Muitos Hops\n"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message to big"); log(1, "513: Message too bi \n"); exit; }; ##-- # Eh feita a gravacao de todas as mensagens para se certificar que todas as mensagens # passem pelo proxy o que eh importante particularmente se as entidades estao usando # um protocolo de transporte diferente ##-- if (method!="REGISTER") { record_route(); }; route(6); ##-- # Secao de finalizacao da chamada ##-- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; ##-- # mensagens subsequentes dentro de um dialogo deveriam tomar um caminho pre-determinado # pela gravacao das rotas (record route) ##-- if (loose_route()) { # marca a logica de roteamento no pedido append_hf("P-hint: rr_enforced\r\n"); if (has_totag() && (method=="INVITE" || method=="ACK")) { xlog("L_ERR", "Main Route: Fixing SDP to $fu $rm (loose route)\n"); if (client_nat_test("3") || isflagset(5)) { setbflag(6); use_media_proxy(); }; }; xlog("L_ERR", "Main Route: Loose Route relaying\n"); route(1); exit; }; if (!uri==myself) { # marca a logica de roteamento no pedido append_hf("P-hint: rr_enforced\r\n"); xlog("L_ERR", "Main Route: Relaing mesage to proxy destination \n"); route(4); route(1); }; ##-- # Verifica baseado na tabela de dominios se a parte host da R-URI comtem um dos dominios # quais o proxy eh responsavel ##-- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbinden..."); } exit; }; ##-- # A rotina abixo eh chamada para marcar o registro com NAT ##-- if (method=="CANCEL") { route(1); exit; } else if (method=="REGISTER") { route(2); exit; } else if (method=="INVITE") { route(3); exit; }; ##-- # Verifica se existem aliases para este endereco e se eles nao pertencem # a teste dominio encaminhamos para frente ##-- lookup("aliases"); if (uri!=myself) { xlog("L_ERR", "Main Route: aliases relayin to other domain proxy\n"); route(4); route(1); exit; }; ##-- # Verifica se encontra o usuario na base de localizacao ##-- if (!lookup("location")) { sl_send_reply("404", "Usuario nao encontrado"); exit; }; route(1); exit; } route[1] { ##-- # Envie isto agora; use "statefull fowarding" pois funciona de forma confiavel de UDP para TCP # Ativa a contabilizacao das chamadas ##-- xlog("L_ERR", "Route[1]: Relaying message\r\n"); setflag(2); ##-- # Ativa a contabilizacao das chamadas perdidas ##-- setflag(3); t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { ##-- # Tratamento padrao para as mensagens de Register ##-- sl_send_reply("100", "Registrando...."); xlog("L_ERR", "Route[2]: Trying Auth: $fu $si\n"); if (isflagset(5) && client_nat_test("7")) { xlog("L_ERR", "Route[2]: Client behind NAT - flag(6)\n"); setbflag(6); fix_nated_register(); force_rport(); # se quiser usar natping options descomente a linha abaixo # setbflag(7); }; if (!radius_www_authorize("")) { www_challenge("", "0"); xlog("L_ERR", "Route[2]: Client Unauthorized: $fu $si\r\n"); sl_send_reply("401", "Unauthorized"); exit; }; if (!check_to()) { xlog("L_ERR", "Route[2]: Client Unauthorized: $fu $si\r\n"); sl_send_reply("401", "Unauthorized"); exit; } else { xlog("L_ERR", "Route[2]: Authenticate: $fu $si\r\n"); }; consume_credentials(); if (!save("location")) { xlog("L_ERR", "Route[2]: Can't save register location\r\n"); sl_reply_error(); }; } route[3] { ##-- # Tratamento padrao para as mensagens de INVITE ##-- xlog("L_ERR", "Route[3]: INVITE: $fu -> $tu $ci\r\n"); # setflag(1); if (!allow_trusted()) { if (!radius_proxy_authorize("")) { proxy_challenge("", "0"); # Realm will be autogenerated exit; } else if (!check_from()) { sl_send_reply("403", "Usue from=ID"); xlog("L_ERR", "Route[3]: Error 403 $fu -> $tu\r\n"); exit; }; } consume_credentials(); ##-- # Verifica se existem aliases para este endereco e se eles nao pertencem # a teste dominio encaminhamos para frente ##-- lookup("aliases"); if (uri!=myself) { xlog("L_ERR", "Main Route: aliases relayin to other domain proxy\n"); route(4); route(1); }; ##-- # VoIP Local ##-- if (uri=~"sip:[1-9][0-9]+@.*") { xlog("L_NOTICE", "OpenSER: Ligacao VoIP local: $fU -> $tu\n"); } else { ##-- # Nacional ##-- if (uri=~"sip:0[1-9][0-9]+@.*") { xlog("L_NOTICE", "OpenSER: Ligacao Nacional: $fU -> $tu\n"); } else { ##-- # Internacional ##-- if (uri=~"sip:00[1-9][0-9]+@.*") { xlog("L_NOTICE", "OpenSER: Ligacao Interacional: $fU -> $tu\n"); }; }; route(5); exit; }; if (!lookup("location")) { append_hf("P-hint: usuario nao encontrado\r\n"); sl_send_reply("404", "Usuario nao encontrado"); exit; }; append_hf("P_hint: usrloc applied\r\n"); t_on_failure("1"); route(4); route(1); } route[4] { ##-- # Secao de travessia do nat ##-- if (isflagset(5)) { setbflag(6); }; if (isflagset(5) || isbflagset(6) || isbflagset(7)) { use_media_proxy(); xlog("L_ERR", "Rota[4]: Travessia de NAT mediaproxy\n"); }; } route[5] { ##-- # Encaminhamento para a rede publica (PSTN) ##-- xlog("L_ERR", "Route[5]: Relaying to PSTN\n"); rewritehostport("201.32.70.21"); route(4); route(1); } route[6] { force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; } failure_route[1] { if (t_check_status("487")) { exit; }; if (isbflagset(6) || isflagset(5)) { end_media_session(); }; onreply_route[1] { ##-- # Bloco de tratamento de respostas ##-- if ((isflagset(5) || isbflagset(6)) && status=~"(180)|(183)|(2[0-9][0-9])") { use_media_proxy(); # search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); } if (isbflagset(6)) { fix_nated_contact(); } } -------------------------------------------------------------------------- Alcindo Schleder Sistema Processa Acessoria e Comunicações Ltda. a.. MSN: a_schleder(a)hotmail.com b.. Skipe: alcindo_schleder Demétrio P. dos Santos, 705 - 95670-000 - Gramado-RS-Brasil Celular + 55 54 9966 7591 alcindo(a)processa.org Fone +55 54 3286-1738 comercial(a)processa.org

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[OpenSER-Users] How to modify the "Referred-by" field inside a REFER
by Amelie CAMELIERE 04 Apr '08

04 Apr '08

Hi everybody, Is it possible to modify the "Referred-by" field inside a REFER SIP message? In fact I want to modify the name displayed by the phone when I use a REFER. Thanks for your help Amélie

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[OpenSER-Users] Openser with RTPProxy in a different scenario
by morton.olson 04 Apr '08

04 Apr '08

Hi all! I have a problem with Openser an RTPProxy. I am using nathelper module to establish a statefull connection with RTPProxy in between the both clients. I player around with it now for a while but I cant get it to work. All the examples in the internet a for clients behind NAT, etc... Maybe one of you might help me. I am using Openser 1.3.0 with RTPProxy 20040107 on Debian Linux "Etch". My scenario looks like this: UA(public ip) ----- Firewall (public IP; 192.168.0.1/24) ---- Openser+RTPProxy (192.168.0.5/24) ----- UA (192.168.0.23/24) The Firewall forward Port 5060 to the Openser which is listening on 5060 as well. Also Ports from 60000-61000 are forwarded to the Openser aswell. (RTPProxy was compile with these Ports) I have no access to the firewall and it does not manipulate SIP Messages. Is there a chance to get this to work? If so, I willl post my configuration file to see what I did wrong. Thanks a lot. Greetz Martin -- View this message in context: http://www.nabble.com/Openser-with-RTPProxy-in-a-different-scenario-tp16349… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 1

[Serusers] Asterisk + OpenSER integration successful
by satish patel 04 Apr '08

04 Apr '08

Dear ser guys.. I have done asterisk with openser integration after spending 2 and 3 days...not time to get nice cold beer... I am user openser front of Asterisk. All sip client register on OpenSER server and call to local extension without Asterisk. caz Asterisk is a media server which are useing only for PSTN calling. When any sip client dialing outside number means mobile number or any outside number Openser forward call to Asterisk and asterisk forward it on PSTN use the E1 or FXO line. ( In my case it is E1 ). OpenSER only user for SIP users or SIP proxy server it can only handle sip request. Asterisk is Media server which can handle IVR , confrance , VoiceMail, and Meetmee or PSTN application. when call come down from PSTN Asterisk forward it on OpenSER as per dialplan configuration. It can handle 1000 of call and you can use it on large production. [sip_client]------------->[Openser]--------->[Asterisk]----[PSTN] My Openser configuration file. #cat ser.cfg debug=6 fork=yes log_stderror=yes listen=<openser_ipaddress> # INSERT YOUR IP ADDRESS HERE port=5060 children=4 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) 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" # 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", "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 (msg:len >= 2048 ) { 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("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; # requests for Media server if(method=="INVITE") { if (uri =~ "sip:0[1-9]@*"){ route(3); 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(); }; } route[3]{ if (uri =~ "sip:0[1-9]@*"){ log(1, "Forwarding to Asterisk\n"); rewritehostport("<asterisk_ip>:5060"); route(1); break; } } #---------END------------# #Asterisk extension.conf [globals] SERADDRESS=203.124.16.109 [outgoing] exten => 5500,1,Playback(hello-world) exten => 5000,1,Answer() exten => 5000,2,Dial(SIP/5000) exten => 5000,3,Hangup() ; XXXX use to forward call on openser extensions. exten => _XXXX,1,Dial(SIP/${EXTEN}@${SERADDRESS},20,r) $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Explore your hobbies and interests. Click here to begin.

1 0

[Serusers] How to set a global variable in ser.cfg and use it in following route() functions?
by lin9k＠yahoo.com 04 Apr '08

04 Apr '08

Hi, Is ser.cfg allow such feature? I'm thinking to use it to hold some variable that can be used for the life of ser.cfg. Thanks in advance, Ling ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com

2 1

[Serusers] calling rewriteuser and rewriteuri from a module?
by lin9k＠yahoo.com 04 Apr '08

04 Apr '08

Hi, I want to provide a module of our own. How do I call rewriteuser and rewriteuri functions from a module? Thanks in advance, Ling ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com

2 1

[OpenSER-Users] 1.3.x package?? forLinux??
by Rey Mulero 04 Apr '08

04 Apr '08

Hello, I am new to OpenSER. I wanted to know if there was a package already compiled to support a P4 Intel machine. I was looking in the website but it really does not spell it out. I see file names openser-1.3.0-tls_linux_i386.tar.gz and openser-1.3.0-notls_linux_i386.tar.gz I am wondering what the difference is? What does tls or notls stand for? Your help is appreciated. Thank you Rey ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

2 1

[Serusers] Asterisk + SER call forwading
by satish patel 04 Apr '08

04 Apr '08

Dear all I want to achive forwading call from SER to Asterisk system i have read about it on voip-info real-time integration of SER & asterisk but its base on mysql i dont have mysql database server i just want to use asterisk as PSTN media means all SIP register on SER and "0XXXXXXXXXXX" number goes thorugh asterisk Ser and asterisk runing on defferent machine. I have put this code in my ser.cfg.sample file.. [globals] SERADDRESS=XXX.XXX.XXX.XXX:5060 [context] exten => <yourexten>,1,Dial(SIP/${EXTEN}@${SERADDRESS},20,r) In ser.cfg: if (method == "INVITE") { if (uri =~ "sip:1[0-9]{10}@*"){ log(1, "Forwarding to Asterisk\n"); rewritehostport("XXX.XXX.XXX.XXX:5060"); t_relay(); break; } } But it is not working u got error "User not Found" i dont know way this happending... $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Get the freedom to save as many mails as you wish. Click here to know how.

1 0

[OpenSER-Users] re : Cert advisory and OpenSER V1.3.1
by Jignesh Gandhi 03 Apr '08

03 Apr '08

Hello, I recently upgraded from SER.8.14 to OpenSER 1.3.1 and had a quick question regarding the SIP CERT advisory suite of tests. Has OpenSER been tested against the SIP CERT advisory suite of test cases, that can be downloaded from http://www.ee.oulu.fi/research/ouspg/protos/testing/c07/sip/. This is the SIP CERT website http://www.cert.org/advisories/CA-2003-06.html thanks, Jignesh Gandhi Movius Interactive Corp

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[Serusers] SER for NAT client
by satish patel 03 Apr '08

03 Apr '08

Dear all I have a bit confusen about server and nat its stupid question but i want to clear it wil your suggestions.. I am going to impliment SER for my customers with domain base authentication. Now if my client behind the NAT then how my RTP make connection between two IP phone [SER] / \ / \ / \ [A] [B] In this case RTP between [A] to [B] right But when both [A] and [B] behind the same NAT router then what about RTP what is the path ??? What is the pass of RTP between [A] and [B] in this case [SER] | | | [NAT] / \ / \ [A] [B] $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Now you can chat without downloading messenger. Click here to know how.

2 3

[Serusers] SER and ptime ?
by Mike Trest - Personal 03 Apr '08

03 Apr '08

Hi, I have registering phones (Linksys) that use g711 or g729. I have a media question in a situation when I attempt to do pure-proxy without media gateway or proxy. Because these user are always messing with their adapter settings, I have variable call setup when I forward certain destinaitons because of ptime miss-match between Linksys and destination. Is it possible to accpet / reject / revise the inbound call when the "ptime" is not specifically 20ms or 30ms (or any specific value) before t_relay on to destination? I see the ptime value in SDP but before I embark on journey trying to "manage" call setup based on that, I want to know if others are working with this issue. ..mike..

1 0

[OpenSER-Users] [Fwd: contact uri]
by Amit Vijayvargiya 03 Apr '08

03 Apr '08

2 1

[Serusers] How to return a string value of an exported functions from a module to be used in ser.cfg?
by lin9k＠yahoo.com 03 Apr '08

03 Apr '08

Hi, I am thinking to write a module and export a function to be called from ser.cfg. I need the exported function to return a string value to be used by another function in ser.cfg. How do I do that in ser.cfg? For example: in ser.cfg: function1(my_function()); in my_module: str *my_function(){ ... return my_string; } Thanks in advance, Ling ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com

2 1

[OpenSER-Users] Res: My first OpenSER post
by Flavio Goncalves 02 Apr '08

02 Apr '08

Hi Yakout, Check SerMyAdmin at http://sourceforge.net/projects/sermyadmin. Regards, Flavio E. Goncalves ----- Mensagem original ---- De: Yakout Esmat <yesmat(a)xtra.co.nz> Para: users(a)lists.openser.org Enviadas: Terça-feira, 1 de Abril de 2008 16:08:44 Assunto: [OpenSER-Users] My first OpenSER post Hi All, This is my first port on the OpenSER forum, I come from Asterisk world looking for scalability and redundancy. I have managed to install openSER 1.3 for the first time yesterday on a CentOS 5 test server. I used SVN, which worked fine. My questions is what is the recommendation in regards to web interface. I can see that we have 2 options SerWEB and Admin Web. If there are any links or leads on HOWTO install either one that would be of great help. Which one is recommended for Admin and also for client interface? Does any of them provide FULL admin functionality, as in creating extensions, configuring voice mail etc….? Many thanks for your assistance. No virus found in this outgoing message. Checked by AVG. Version: 7.5.519 / Virus Database: 269.22.1/1352 - Release Date: 3/31/2008 10:13 AM -----Anexo incorporado----- _______________________________________________ Users mailing list Users(a)lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users

2 1

[Serusers] rtpproxy address filling
by Andres 02 Apr '08

02 Apr '08

Hi, I have a question regarding the way rtpproxy handles 'address filling'. After a session has been created, the rtpproxy pre-fills the caller and callee's addresses and we see that in the rtpproxy output like: pre-filling caller's address with 192.116.246.234:41000 pre-filling callee's address with 192.116.246.234:20000 Then when it sees the actual packets coming in from a different source port, it updates the address and we see it in the log like: callee's address filled in: 192.116.246.234:1024 (RTP) The audio then flows fine both ways. My question is, what would happen it the actual packets came in from a different IP while the rtpproxy was waiting between the state of 'pre-filling' and 'address filled' states? Will the rtpproxy accept such a change that includes a new IP? Will it ignore packets from a different IP and continue the session normally? Or will it abort the session completely? Thanks, Andres http://www.telesip.net

4 10

[Serusers] Fwd: Unix Socket Problem
by frq ser 02 Apr '08

02 Apr '08

Hi Stefan Thanks a lot for your prompt response. I can hear the announcement now. Previously my audio setup was not correct. Best Regards, Frq On Wed, Apr 2, 2008 at 12:11 PM, Stefan Sayer <stefan.sayer(a)iptego.com> wrote: > Hello, > > frq ser wrote: > > > Hi > > > > Thanks for your reply Stefan and my apologies for coming back to this > > issue after such a long time. As you suggested, I included SDP headers in my > > Sipsak INVITE msg and the SEMS seems to like it. I am trying to play the > > announcement using a INVITE message to SEMS at :5070. The SEMS sends a 200 > > OK message, but no audio was being played. In the log, I got these messages > > (just including the important messages)... > > > > .. > > (17358) DEBUG: execute (AmInterfaceHandler.cpp:172): Request OK: > > dispatch it! (17358) DEBUG: getAnnounceFile (Announcement.cpp:79): trying > > '/usr/local/lib/sems/audio/192.168.3.208/200.wav < > > http://192.168.3.208/200.wav>' (17358) DEBUG: getAnnounceFile > > (Announcement.cpp:84): trying '/usr/local/lib/sems/audio/200.wav' > > .. > > .. > > (17358) DEBUG: getCompatiblePayloads (AmSdp.cpp:348): using global > > address: 192.168.3.208 <http://192.168.3.208/> (17358) DEBUG: > > negotiate (AmSession.cpp:262): remote party doesn't support telephone events > > > > Seeing the last line, I then used an Xlite softphone (instead of sipsak) > > to contact SEMS. Again, the call gets established for about 10 seconds and > > then just gets disconnected and no audio can be heard. Nothing in > > > To me the logs look normal. seems like it plays the announcement and then > hangs up, as it should. the default_en.wav (or 200.wav, if you've copied it > there), is only a few seconds long. are you sure that audio is setup > correctly (volume etc), i.e. does the xlite test call work? if you call e.g. > sip:conference@iptel.org <sip%3Aconference(a)iptel.org>, do you hear > something? > > can you run tcpdump -i any or run wireshark to see whether packets flow > between SEMS and xlite? > > you could also PM me a wireshark dump so i could have a look at what's > going on. > > Stefan > >

1 0

Re: [Serusers] Unix Socket Problem
by frq ser 02 Apr '08

02 Apr '08

Hi Thanks for your reply Stefan and my apologies for coming back to this issue after such a long time. As you suggested, I included SDP headers in my Sipsak INVITE msg and the SEMS seems to like it. I am trying to play the announcement using a INVITE message to SEMS at :5070. The SEMS sends a 200 OK message, but no audio was being played. In the log, I got these messages (just including the important messages)... .. (17358) DEBUG: execute (AmInterfaceHandler.cpp:172): Request OK: dispatch it! (17358) DEBUG: getAnnounceFile (Announcement.cpp:79): trying '/usr/local/lib/sems/audio/192.168.3.208/200.wav' (17358) DEBUG: getAnnounceFile (Announcement.cpp:84): trying '/usr/local/lib/sems/audio/200.wav' .. .. (17358) DEBUG: getCompatiblePayloads (AmSdp.cpp:348): using global address: 192.168.3.208 (17358) DEBUG: negotiate (AmSession.cpp:262): remote party doesn't support telephone events Seeing the last line, I then used an Xlite softphone (instead of sipsak) to contact SEMS. Again, the call gets established for about 10 seconds and then just gets disconnected and no audio can be heard. Nothing in the SEMS log (see below) suggests that there is any problem (as far as I can understand). I've checked that the file /usr/local/lib/sems/audio/200.wav is present and I can play it directly in Linux using play command. (The file 200.wav is just a copy of default_en.wav file) I just want to know why the announcement file is not being played and why the call between softphone and SEMS just gets disconnected after a while. I am sure it would be something very basic :( Thanks a lot for your help Frq

2 1

[OpenSER-Users] openser social networking events - April-June 2008
by Daniel-Constantin Mierla 02 Apr '08

02 Apr '08

Hello, in the period of time April - June 2008, there will be a series of openser social networking events organized in different places world wide, during the time some other Open Source or VoIP events take place in the same location. The events are most probably in the form of a dinner, organized together with local representatives of openser community. The events are intended to be rather small size, to allow each one to know the others, in a nice and friendly discussion environment. The primary goal is to strengthen the relations within community and give opportunity to network for collaboration and business. Participation to any of these events is free, everybody pays for its expenses. Anyone interested in openser, sip or voip in general is welcome to join. Next events are: - Bucharest, Romania, April 12, 2008, tied up with ROSDEV2008 (http://www.rosdev.ro - Romanian site only) - Orlando, FL, USA, April 22, 2008, tied up with OPENSER-Asterisk SIP Masterclass (http://www.edvina.net) - Barcelona, Spain, May 6, 2008, tied up with OPENSER-Asterisk SIP Masterclass (http://www.avanzada7.com/) - Vienna, Austria, May 23, 2008, on the way to AsteriskTag (http://www.asterisk-tag.org/wiki/Hauptseite) - Berlin, Germany, May 28, 2008, tied up with LinuxTag (http://www.linuxtag.org) - Amsterdam, Netherlands, June 3, 2008, tied up with VoN.x Europe (http://www.von.com/web/index.php) The hour and place for each event will be announced few days before as we know the number of attending people and we make the reservation. Please send email to me to reserve a place, although you are welcome to come at any time and without prior notice, you may find that there is no more available seat to join the table. If you live in the surroundings of these cities, and want to help with organization, you know a nice and quite place suitable for the event, let me know. Cheers, Daniel -- http://www.asipto.com

1 0

[OpenSER-Users] Building Carrierroute Tables
by Douglas Garstang 02 Apr '08

02 Apr '08

So, I was kind of wondering how those using carrierroute where populating their routes. I was hoping to build a cost table that contained the costs for all our routes and all our carriers, and then construct the carrierroute table as a view derived from the cost table. I guess my SQL (and those around me I asked about it) isn't that good, because I haven't been able to come up with a way to do it yet. Doug. ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com

2 1

[OpenSER-Users] My first OpenSER post
by Yakout Esmat 01 Apr '08

01 Apr '08

Hi All, This is my first port on the OpenSER forum, I come from Asterisk world looking for scalability and redundancy. I have managed to install openSER 1.3 for the first time yesterday on a CentOS 5 test server. I used SVN, which worked fine. My questions is what is the recommendation in regards to web interface. I can see that we have 2 options SerWEB and Admin Web. If there are any links or leads on HOWTO install either one that would be of great help. Which one is recommended for Admin and also for client interface? Does any of them provide FULL admin functionality, as in creating extensions, configuring voice mail etc….? Many thanks for your assistance. No virus found in this outgoing message. Checked by AVG. Version: 7.5.519 / Virus Database: 269.22.1/1352 - Release Date: 3/31/2008 10:13 AM

1 0

[OpenSER-Users] Problem activating auth and auth_db modules using openser 1.3.1
by Pascal Maugeri 01 Apr '08

01 Apr '08

Hi My 1.3.1 server does not start when I activate the modules auth and auth_db modules with the following configuration: loadmodule "auth.so" loadmodule "auth_db.so" [...] modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "user_column", "username") modparam("auth_db", "password_column", "password") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "db_url", "mysql://openser:openserrw@192.168.1.4 /openser") modparam("auth_db", "load_credentials", "") There is no error in the log file but no other module is loaded whenever I enable auth and auth_db. Is there anything else I should enable to configure these two modules ? When I disable the loading of these two modules everything works fine. Regards, Pascal

3 3

[Serusers] help required disabling DNS lookup
by geethanjali.csg＠wipro.com 01 Apr '08

01 Apr '08

Hi, I am trying to set up calls between 2 domains without using DNS lookup. UA -> domain1 -> domain2 -> UA I am using static forwarding route the all requests. Requests should be routed using this forward logic only. Both DNS lookup and forward logic are happening. Since I have not configured the DNS, an error "Unresolvable Destination" is seen by the UA. Ser log displays errors related to DNS lookup. I don't want the DNS lookup to happen. Requests should be routed using the forward logic only. Can someone help me solve this problem. Regards, Geethanjali C S Wipro Technologies The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

3 3

[OpenSER-Users] [openser 1.3] loading module auth.so stop my server
by Pascal Maugeri 01 Apr '08

01 Apr '08

Hi I'm still unable to have a working server 1.3 that loads the module auth.so. It is weird because there is no error message in my log file also I set the debug level to maximum. But after starting the openser service, it says it is successful but there is no running process (pgrep openser does not show anything, when it does if I remove the load auth.so)! How can I get more debugging info about the loading and initialization of auth ? Can anybody share with me his or her configuration file ? I may miss something in mine: loadmodule "sl.so" [...] loadmodule "auth.so" (I believe it is sufficient). Is there any known issue using Redhat Enterprise 4 ? Regards, Pascal

1 0

[OpenSER-Users] Problem in finding invite tsxn for ACK.
by Vikas CG. 01 Apr '08

01 Apr '08

Hi All, We were using the openser server as reg/proxy to test sip gateway. We observed that if we run openser in TCP mode it will add 2 record-routes one for TCP and UDP. But when we put hold UA-client i.e. when we send reinvite the openser server is adding only one record-routes But in ACK for reinvite it is adding 2 record-routes. I am also attaching the sip flow. It would be grateful if any body tells us is this behavior is correct? Because we are depending on record route to find the invite tsxn for ack. Thanks, Vikas C.G.

2 1

Re: [OpenSER-Users] PUA/PUA_MI: More than one Publish sent
by Schumann Sebastian 01 Apr '08

01 Apr '08

Sorry, one short mistake: Of course, after last expire, the expires only removed the a.1207041424.4027.8.0, which was the last one published. The a.1207041424.4027.7.0 from the same publication remains in the table. It has nothing to do with odd and even numbers. Regards Sebastian -----Original Message----- From: Schumann Sebastian Sent: Tuesday, 01. April 2008 10:32 AM To: 'Anca Vamanu' Cc: users(a)lists.openser.org Subject: RE: [OpenSER-Users] PUA/PUA_MI: More than one Publish sent Dear Anca Thank you for the response. The publication of more than one message in your described scenario I understand. In my case, immediately after the first PUBLISH is sent, a second one is sent afterwards. OpenSER does not use the most recent etag for an update, but creates a new one, that is never been removed automatically. If I send an expires=0 message after the publication, only the first published etag is removed. Here my publication script: #!/bin/bash # FIFO command to PUBLISH hard-presence state of user via PUA # #config values OP_SYSUSER="alice(a)10.96.115.129" OP_SIPURI="sip:$OP_SYSUSER" OP_STATE="open" OP_EXPIRES="-1" #build reply fifo file rm -f /tmp/openser_fifo_reply mkfifo /tmp/openser_fifo_reply cat /tmp/openser_fifo_reply& cat >/tmp/openser_fifo <<EOF :pua_publish:openser_fifo_reply $OP_SIPURI $OP_EXPIRES presence application/pidf+xml . <?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:x ml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' ntity='$OP_SIPURI'><tuple id='gigpoej8'><status><basic>$OP_STATE</basic></status><rpid:user-input> Har d-State</rpid:user-input></tuple><dm:person id='0jfqoiet'><rpid:activities><rpid:unknown/></rpid:activities><dm:note >Hard-State</dm:note></dm:person></presence> EOF And my expires script: #!/bin/bash # FIFO command to PUBLISH hard-presence state of user via PUA # #config values OP_SYSUSER="alice(a)10.96.115.129" OP_SIPURI="sip:$OP_SYSUSER" OP_STATE="closed" OP_EXPIRES="0" #build reply fifo file rm -f /tmp/openser_fifo_reply mkfifo /tmp/openser_fifo_reply cat /tmp/openser_fifo_reply& cat >/tmp/openser_fifo <<EOF :pua_publish:openser_fifo_reply $OP_SIPURI $OP_EXPIRES presence application/pidf+xml . <?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:x ml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' ntity='$OP_SIPURI'><tuple id='giqpoej8'><status><basic>$OP_STATE</basic></status><rpid:user-input> Har d-State</rpid:user-input></tuple><dm:person id='0jfqoiet'><rpid:activities><rpid:unknown/></rpid:activities><dm:note > ard-State</dm:note></dm:person></presence> EOF I start first publish, then expire, then publish again and so on. The following etags are created normally: e.g. a.1207041424.4027.1.0 a.1207041424.4027.3.0 a.1207041424.4027.5.0 At some point, another one is created together with the "normal" one, e.g. a.1207041424.4027.7.0 and a.1207041424.4027.8.0 The expires only removed the a.1207041424.4027.7.0 (all odd numbers). The published one with the even number stays forever. I don't know what can be the reason. Here my OpenSER configuration file extract from pua module: modparam("pua", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("pua", "db_table", "pua") modparam("pua", "update_period", 10) modparam("pua", "default_expires", 600) Thanks for help. Best regards Sebastian -----Original Message----- From: Anca Vamanu [mailto:anca@voice-system.ro] Sent: Tuesday, 25. March 2008 3:05 PM To: Schumann Sebastian Cc: users(a)lists.openser.org Subject: Re: [OpenSER-Users] PUA/PUA_MI: More than one Publish sent Hello Sebastian, Indeed sometimes more that one Publish messages are sent. I will explain when and why. It is possible for someone to request through the MI command to publish a state that should be active for 5000 seconds (you probably know that -1 in expires filed means infinite time). But the max expires in presence server is lets say 3600 sec. If no other publish request is received before the first one expires, the record is deleted. So, to respect the requested time, the pua module generates itself another publish when the first one is close to expire just to update the validity time. Now, if for a subsequent mi command you do not sent the etag parameter ( set it to '.' value), the most recent etag will be found in pua module, and a correct Publish message will be sent. I have just now tried the scenario you described and it works perfectly for me. So, I want to ask you to say exactly what you are doing because there must be something you have missed out. Is the presentity uri exactly the same for the first and the subsequent mi requests? After how much time have you sent the mi request that fails? Regards, Anca Vamanu Schumann Sebastian wrote: > Dear all > > I am using OpenSER 1.3.1 branch release R3936. I face some bad > behaviour using pua and pua_mi modules. > > I publish presence states via the management interface. Sometimes, it > happens that multiple PUBLISH message are sent by one request. > Unfortunately, there is no "system" or oder when this occurs. It just > happens after some publications. > > In case of an expiration (Expires=0, E-Tag .) it is not that bad, I > receive the log message: ERROR:presence:update_presentity: No E_Tag match. > > In case of publication (Expires=-1, E-Tag .) it is very bad: After > publication, he receives also two similar messages, and threats both > PUBLISH messages as two different ones. Thus, creates two different > presentity entries with different E-Tags. If the state is expired now > via MI (Expires: -1, E-Tag .) only one of the E-Tags is removed. The > second stays in the presentity table and the user still appears online. > > Any ideas what can be the reason and how to solve it? > > Best regards > Sebastian > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users >

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Re: [OpenSER-Users] unable to find dialog for BYE
by Jayesh Nambiar 01 Apr '08

01 Apr '08

Hi, This happens only when the call is disconncted from the called party side. When the call initator disconnects there is no such warning in it. Although the dialog ends properly. w/regards, Jayesh ----- Original Message ---- From: Ovidiu Sas <osas(a)voipembedded.com> To: Jayesh Nambiar <voip_freak(a)yahoo.co.in> Cc: users(a)lists.openser.org Sent: Monday, 31 March, 2008 7:21:35 PM Subject: Re: [OpenSER-Users] unable to find dialog for BYE Are you getting this all the time? What kind of matching algorithm are you using? Please post a SIP trace. There are several scenarios that can trigger this warning and it is difficult to debug without providing extra information about your scenario. Regards, Ovidiu Sas On Wed, Mar 26, 2008 at 6:00 AM, Jayesh Nambiar <voip_freak(a)yahoo.co.in> wrote: > > > > Hi All, > I have recently upgraded to openser 1.3.1 and I am trying to explore the > Dialog module. > I want to maintain the dialog for specific set of calls, so I have defined a > route and I have written setflag(4) in that route so that all calls getting > into that route will have the dialogs maintained. Flag 4 is defined for > dialog module. > > Now when the call initiator hangs up; the dialog starts and ends properly > but when the called party hangs up, I get this message in the syslog: > " unable to find dialog for BYE" > > Can someone please guide me on how to interpret this WARNING and what > measures should be taken to avoid the same. > > Thanks in advance > > w/regards, > Jayesh > ________________________________ > Save all your chat conversations. Find them online. > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > Unlimited freedom, unlimited storage. Get it now, on http://help.yahoo.com/l/in/yahoo/mail/yahoomail/tools/tools-08.html/

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[OpenSER-Users] writing provider info in sip_trace mysql table.
by Itamar - IspBrasil 01 Apr '08

01 Apr '08

hi. anyone know if is possible to store information in sip_trace mysql table about the company who have completed the call ? example: outbund_company = xxx.xxx.xxx.xxxx -------------------- Itamar Reis Peixoto e-mail/msn: itamar(a)ispbrasil.com.br skype: itamarjp icq: 81053601 +55 11 4063 5033 +55 34 3221 8599

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Re: [OpenSER-Users] dialog matching criteria
by Jayesh Nambiar 01 Apr '08

01 Apr '08

Hi Bogdan, Thanks for your reply. I tried of acheiving the same as follows: Enabled DB support for dialog module. If the called number is 123456, I first query the dialog table to retreive the count on to_uri and compare it with a certain limit value. If the count is equal to or greater than the limit value, just reply as 403, Forbidden. I am about to finish this today and will let you know the results. I just wanted an expert advice on if the logic is ok and efficient. Also there is a column in the dialog table which is caller_route_set and callee_route_set, can someone please tell me from where these values are retreived from as it stays blank while checking the dlg_list. Also is it possible to insert some custom value in this column to implement any further logic. Thanks, w/regards, Jayesh ----- Original Message ---- From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> To: Jayesh Nambiar <voip_freak(a)yahoo.co.in> Cc: users(a)lists.openser.org Sent: Monday, 31 March, 2008 5:34:05 PM Subject: Re: [OpenSER-Users] dialog matching criteria Hi Jayesh, Such support is not available, but it is on the roadmap for 1.4. Regards, Bogdan Jayesh Nambiar wrote: > Hi All, > Is there a way in dialog module that I can keep count of active > dialogs based on certain parameters like ruri, to_uri etc. > For example, if i need to do a check of active dialogs on a called > number which is 123456 and take certain actions if the number of > active dialogs to called number 123456 is greater than or equal to > certain integer value. > > Is the above requirement a possibility in openser? > Any help regarding this will be highly appreciated. > > Thanks in advance. > > w/regards, > Jayesh > > ------------------------------------------------------------------------ > Get the freedom to save as many mails as you wish. Click here to know > how. > <http://in.rd.yahoo.com/tagline_mail_5/*http://help.yahoo.com/l/in/yahoo/mai…> > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > Forgot the famous last words? Access your message archive online at http://in.messenger.yahoo.com/webmessengerpromo.php

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sr-users April 2008