sr-users February 2006

sr-users@lists.kamailio.org

189 participants
373 discussions

[Users] avpops: new function avp_db_query()

by Daniel-Constantin Mierla

Hello, avpops module has a new function which allow to execute raw SQL queries and store the result in AVPs. avp_db_query(query, dest); The query given as parameter can contain pseudo-variables. Using this function you can benefit of full database system features, being able to do joins, unions, etc. Old db-related functions are in place since they are faster for their usage case. The documentation of the of avpops module was updated and posted at: http://openser.org/docs/modules/1.1.x/avpops.html A small example of usage: limit the number of calls done in the last day: if(is_method("INVITE") && !has_totag()) { if(avp_db_query("select count(*) from acc where username='$fU' and domain='$fd' and method='INVITE' and timestamp>=$Ts-24*3600", "$avp(i:234)")) { if(avp_chech("$avp(i:234)", "ge/i:10")) { sl_send_reply("403", "too many calls in the last day"); exit(); } } } Cheers, Daniel

18 years, 1 month

[Serusers] TCP connect and other related stuff problems

by Christian Thomas

Hi everyboy, I have a problem when my SER 0.9.4 has to connect to a carrier thru tcp transport. I'm working in stateful mode, of course, and I use t_relay_to_tcp(ipcarrier,port) when I can detect that the carrier GW wants to speak with SER thru TCP. It works not very well as there is the following errors when the carrier GW send a 200 OK to SER after the INVITE procedure : tcp_blocking_connect : timeout(10) tcpconn_connect : connect failed tcp_send: connect failed tcp_send failed msg_send : tcp_send failed t_forward_nonack : sending request failed The UAS which sent the INVITE receive a 477 : next hop error Is it a timeout problem ? If so, how to resolve it? How to increase this timeout for TCP? Thanks a lot Christian

18 years, 1 month

[Serusers] How to remove Contact HF in 200 reply for REGISTER

by Arek Bekiersz

Hi, How to disable sending Contact HF in OK send to a successful (i.e. authenticated) REGISTER request? Just use remove_hf() from textops or there is other option? I've got stupid UA that behaves badly when working behind NAT. First it sends REGISTER without anything (no Contact, Expires). On successful authentication SER responds 200 OK with Contact HF, set to public IP, exactly how SER sees UA behind NAT (like some_IP:61001). Then the stupid UA sends another REGISTER message, this time with Contact and Via HF set to this public value (i.e. some_IP:61001). It spoils my nathelper/rtpproxy detection of devices behind NAT. Of course I can enable it by regexp on User-Agent HF, but his is not an option, as plenty of those UAs runs not behind NAT. -- Regards, Arek Bekiersz

18 years, 2 months

[Users] openser & sems memory error

by Ibrahim TUNALI

Hi, i installed openser (1.0 stable) and sems (0.9 stable). I recive "ERROR:tm:write_to_fifo: writev failed: Cannot allocate memory" error when openser send request to sems voice mail. Is it compile error or do you have any solution for this. regards openser voicemail seciton; -------------------------- # switch to stateful mode: if (!t_newtran()){ sl_send_reply("500","could not create transaction"); return; }; # prevent timeout on the other side: t_reply("100","Trying - just wait a minute !"); # actively absorb ACKs if (method == "ACK") { t_relay(); return; } if (method=="INVITE"){ # no service number, redirect to voicemail. # do not forget to load AVPs so that voicemail gets the # callee's email address. avp_db_load("$ruri","i:34/$email_scheme"); if(!t_write_req("/tmp/am_fifo","voicemail/vm_email")) { t_reply("500","error contacting sems"); }; return; } else if (method=="BYE" || method=="CANCEL") { # Sems should already know which plug-in is handling that # call. "bye" is no plug-in name. It is a reserved name which # tells Sems to terminate the call. if(!t_write_req("/tmp/am_fifo","bye")) { t_reply("500","error contacting sems"); }; }; openser log; ------------ Feb 23 17:18:17 voip /usr/sbin/openser[28398]: ERROR:tm:write_to_fifo: writev failed: Cannot allocate memory Feb 23 17:18:17 voip /usr/sbin/openser[28398]: ERROR:tm:t_write_req: write_to_fifo failed Feb 23 17:18:17 voip /usr/sbin/openser[28396]: Warning: sl_send_reply: I won't send a reply for ACK!! Feb 23 17:18:18 voip /usr/sbin/openser[28392]: Warning: sl_send_reply: I won't send a reply for ACK!! Feb 23 17:18:20 voip /usr/sbin/openser[28396]: Warning: sl_send_reply: I won't send a reply for ACK!! Feb 23 17:18:24 voip /usr/sbin/openser[28398]: Warning: sl_send_reply: I won't send a reply for ACK!! Feb 23 17:18:28 voip /usr/sbin/openser[28394]: Warning: sl_send_reply: I won't send a reply for ACK!! Feb 23 17:18:32 voip /usr/sbin/openser[28398]: Warning: sl_send_reply: I won't send a reply for ACK!! Feb 23 17:18:36 voip /usr/sbin/openser[28394]: Warning: sl_send_reply: I won't send a reply for ACK!! Feb 23 17:18:40 voip /usr/sbin/openser[28398]: Warning: sl_send_reply: I won't send a reply for ACK!! Feb 23 17:18:44 voip /usr/sbin/openser[28394]: Warning: sl_send_reply: I won't send a reply for ACK!! sems log; ------------ Feb 23 17:19:45 voip Sems[28477]: Error: (AmUtils.cpp)(str2i)(147): str2i: unexpected char s in ser Feb 23 17:19:45 voip Sems[28477]: Error: (AmUtils.cpp)(str2i)(147): str2i: unexpected char s in ser Feb 23 17:19:45 voip Sems[28480]: Debug: (sems.cpp)(main)(386): Loading application plug-ins Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(load)(87): loading /usr/lib/sems/plug-in/apps/announcement.so ... Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAppPlugIn)(281): application 'announcement' loaded. Feb 23 17:19:45 voip Sems[28480]: Debug: (SemsConfiguration.cpp)(reloadModuleConfig)(70): using inline configuration for module announcement Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(load)(87): loading /usr/lib/sems/plug-in/apps/bot.so ... Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAppPlugIn)(281): application 'bot' loaded. Feb 23 17:19:45 voip Sems[28480]: Debug: (AmServer.cpp)(registerFct)(401): AmUnServer::registerFct: function 'bot_start' has been registered. Feb 23 17:19:45 voip Sems[28480]: Debug: (AmServer.cpp)(registerFct)(208): AmFifoServer::registerFct: function 'bot_start' has been registered. Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(load)(87): loading /usr/lib/sems/plug-in/apps/conference.so ... Feb 23 17:19:45 voip Sems[28482]: Debug: (AmThread.cpp)(_start)(63): Thread 28482 is starting. Feb 23 17:19:45 voip Sems[28480]: Debug: (AmThread.cpp)(start)(85): Thread 28482 is just created. Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAppPlugIn)(281): application 'conference' loaded. Feb 23 17:19:45 voip Sems[28480]: Debug: (SemsConfiguration.cpp)(reloadModuleConfig)(70): using inline configuration for module conference Feb 23 17:19:45 voip Sems[28480]: Warning: (Conference.cpp)(onLoad)(77): no join_sound specified in configuration Feb 23 17:19:45 voip Sems[28480]: Warning: (Conference.cpp)(onLoad)(78): file for module conference. Feb 23 17:19:45 voip Sems[28480]: Warning: (Conference.cpp)(onLoad)(84): no drop_sound specified in configuration Feb 23 17:19:45 voip Sems[28480]: Warning: (Conference.cpp)(onLoad)(85): file for module conference. Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(load)(87): loading /usr/lib/sems/plug-in/apps/echo.so ... Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAppPlugIn)(281): application 'echo' loaded. Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(load)(87): loading /usr/lib/sems/plug-in/apps/isdngw.so ... Feb 23 17:19:45 voip Sems[28480]: Info: (IsdnConfiguration.cpp)(reload)(91): isdngw version 0.2 loading (supported backends: isdn4linux) ... Feb 23 17:19:45 voip Sems[28480]: Warning: (SemsConfiguration.cpp)(reloadModuleConfig)(66): no configuration found for module isdngw, maybe you want to specify config.isdngw in config file. Feb 23 17:19:45 voip Sems[28480]: Info: (IsdnGatewayFactory.cpp)(IsdnGatewayFactory)(81): isdngw: isdn4linux support disabled in configuration Feb 23 17:19:45 voip Sems[28480]: Info: (IsdnGatewayFactory.cpp)(IsdnGatewayFactory)(148): isdngw initialization successful. Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAppPlugIn)(281): application 'isdngw' loaded. Feb 23 17:19:45 voip Sems[28480]: Debug: (IsdnGatewayFactory.cpp)(onLoad)(199): registering fifo function isdninfo for web access... Feb 23 17:19:45 voip Sems[28480]: Debug: (AmServer.cpp)(registerFct)(208): AmFifoServer::registerFct: function 'isdninfo' has been registered. Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(load)(87): loading /usr/lib/sems/plug-in/apps/number_reader.so ... Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAppPlugIn)(281): application 'number_reader' loaded. Feb 23 17:19:45 voip Sems[28480]: Debug: (SemsConfiguration.cpp)(reloadModuleConfig)(70): using inline configuration for module number_reader Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(load)(87): loading /usr/lib/sems/plug-in/apps/voicemail.so ... Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAppPlugIn)(281): application 'voicemail' loaded. Feb 23 17:19:45 voip Sems[28480]: Debug: (SemsConfiguration.cpp)(reloadModuleConfig)(70): using inline configuration for module voicemail Feb 23 17:19:45 voip Sems[28480]: Debug: (AnswerMachine.cpp)(loadEmailTemplates)(103): loading /etc/sems//default.template ... Feb 23 17:19:45 voip Sems[28480]: Debug: (sems.cpp)(main)(390): Loading audio plug-ins Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(load)(87): loading /usr/lib/sems/plug-in/audio/gsm.so ... Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAudioPlugIn)(218): codec id 3 inserted Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAudioPlugIn)(233): payload 'GSM'inserted with id 3 Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(load)(87): loading /usr/lib/sems/plug-in/audio/ilbc.so ... Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAudioPlugIn)(218): codec id 4 inserted Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAudioPlugIn)(237): payload 'iLBC'inserted with id 96 Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAudioPlugIn)(257): file format iLBC inserted Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(load)(87): loading /usr/lib/sems/plug-in/audio/speex.so ... Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAudioPlugIn)(218): codec id 6 inserted Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAudioPlugIn)(237): payload 'speex'inserted with id 96 Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(load)(87): loading /usr/lib/sems/plug-in/audio/wav.so ... Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAudioPlugIn)(218): codec id 1 inserted Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAudioPlugIn)(218): codec id 2 inserted Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAudioPlugIn)(233): payload 'PCMU'inserted with id 0 Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAudioPlugIn)(233): payload 'PCMA'inserted with id 8 Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAudioPlugIn)(257): file format Wav inserted Feb 23 17:19:45 voip Sems[28480]: Debug: (sems.cpp)(main)(394): Starting session container Feb 23 17:19:45 voip Sems[28486]: Debug: (AmThread.cpp)(_start)(63): Thread 28486 is starting. Feb 23 17:19:45 voip Sems[28480]: Debug: (AmThread.cpp)(start)(85): Thread 28486 is just created. Feb 23 17:19:45 voip Sems[28480]: Debug: (sems.cpp)(main)(397): Starting mailer Feb 23 17:19:45 voip Sems[28480]: Debug: (AmThread.cpp)(start)(85): Thread 0 is just created. Feb 23 17:19:45 voip Sems[28480]: Debug: (sems.cpp)(main)(400): Starting RTP scheduler Feb 23 17:19:45 voip Sems[28480]: Debug: (AmThread.cpp)(start)(85): Thread 0 is just created. Feb 23 17:19:45 voip Sems[28480]: Debug: (sems.cpp)(main)(403): Starting ICMP watcher Feb 23 17:19:45 voip Sems[28480]: Debug: (AmThread.cpp)(start)(85): Thread 0 is just created. Feb 23 17:19:45 voip Sems[28480]: Debug: (AmThread.cpp)(start)(85): Thread 0 is just created. Feb 23 17:19:45 voip Sems[28480]: Info: (AmServer.cpp)(run)(147): FIFO server started Feb 23 17:19:45 voip Sems[28487]: Debug: (AmThread.cpp)(_start)(63): Thread 28487 is starting. Feb 23 17:19:45 voip Sems[28488]: Debug: (AmThread.cpp)(_start)(63): Thread 28488 is starting. Feb 23 17:19:45 voip Sems[28489]: Debug: (AmThread.cpp)(_start)(63): Thread 28489 is starting. Feb 23 17:19:45 voip Sems[28489]: Error: (AmIcmpWatcher.cpp)(run)(47): ICMP Watcher: could not create RAW socket: Operation not permitted Feb 23 17:19:45 voip Sems[28489]: Error: (AmIcmpWatcher.cpp)(run)(48): ICMP Watcher: try to run SEMS as root or suid. Feb 23 17:19:45 voip Sems[28489]: Debug: (AmThread.cpp)(_start)(68): Thread 28489 is ending. Feb 23 17:19:45 voip Sems[28490]: Debug: (AmThread.cpp)(_start)(63): Thread 28490 is starting. Feb 23 17:19:45 voip Sems[28490]: Info: (AmServer.cpp)(run)(321): Unix socket server started

18 years, 2 months

[Serusers] Proposal for new features

by Arek Bekiersz

Hi, As my current work may be interesting for the rest, I wonder if you will find those new features useful. Every opinion is appreciated. Some time ago I commited a small 'ldap' module, all-in-one. As there was some (small :-) ) feedback from people using this module, I propose to split it into following modules: 1) auth_ldap: set of authentication routines, working with standard or user-defined Ldap schemas 2) group_ldap: group membership module based on Ldap backend 3) uri_ldap: uri mangling module, based on ldap backend 4) avp_ldap: (new) module for operating on AV-pairs from Ldap, working with both user-defined or standard dbase schemas Later, it is possible to write small SER 'soap' modules, using one of freely available C/C++ soap libraries. Those modules will perform various tasks: 5) soap_ipdr: module for reporting service usage information (billing) in XML, to be used by external billing systems. Information will include all call attempts (also failed ones like CC,CAD,UCN,UCI and possibly CID completion codes from IPDR's VoIP service definition). It will return following XML documents (using SOAP as transport): * IPDRDoc - usage for specific user (user@domain) * IPDRSettlementDoc - usage for domain NOTE: that using IPDRSettlementDoc for domain is based on model where our cooperants have one virtual SIP domain. it is then easiest way to make settlement using IPDRSettlementDoc for one domain. 6) soap_parlayx: Such module would act as a gateway between SER and external Parlay X Web Services software. When called, this module will analyze SIP request and forward it to external Parlay X software. That software will fulfill request and return ReturnValue, with specific Action. Then SER soap_parlayx module will perform desired action on SIP request. More on Parlay X Web services v.2.0 can be found at: http://www.parlay.org/en/specifications/ NOTE: This behaviour mimics a hypothetic SIP-CGI gateway, but this time it is a SIP-SOAP gateway. EXAMPLE: could be the HandleBusy operation from "Part 3: Call Notification" of Parlay X Web services: 1. SIP encounters "486 Busy" situation 2. SER Script writer calls parlayx_handle(HandleBusy) function from soap_parlayx module 3. soap_parlayx module calls external software, sending handleBusyRequest, containing CallingParty, CalledParty 4. external software returns handleBusyResponse with Action structure, containing ActionToPerform (like 'route' on not route), RoutingAddress, Charging (charge for using of service) 5. SER soap_parlayx module modifies SIP request and forwards it to voicemail_SIP location. In addition it stores relevant info in IPDR data recorder for later use. External Parlay X software will be basically a SOAP engine, it can be written in any language. Unlike a soap_parlayx module itself, that engine will not be a part of SER. 7) SIP SOAP initiative Seems that IP world is slowly evolving in direction of service enabled architecture, usually based on SOAP/UDDI standards. For this reason maybe it is worth to initiate work on SIP SOAP draft standard? It would basically mimic existing IETF SIP CGI draft standard, but with far greater opportunities for creating new services, that SOAP standard brings. It will beautifuly fulfil current work in progress in my company and an European project, codename VISP (Virtual Internet Service Provider), that we are coordinator of. Project is focusing on enabling smooth company-to-company interoperability in mutual creation of new services in fast changing IP world. Including VoIP. More information can be found on Web site: http://www.visp-project.org/ -- Regards, Arek Bekiersz

18 years, 2 months

[Users] Using t_reply()

by Douglas Garstang

Why can't I do this??? Arrgh! I want to be able to call onreply_route[35] when we get a reply. So, not surprisingly, in onreply_route[35] I want to see what sort of reply I got, and then send ANOTHER reply back to the originater of the message telling them something was wrong. Why can't I use t_reply in a reply_route? What's the most likely away around this, because I sure must be missing something fundamental. route[10] { t_on_failure("11"); t_on_reply("35"); rewritehostport("pbx1-mgt0.ipt.oneeighty.com:5060"); if ( !t_relay() ) { xlog ("L_INFO","t_relay() returned error"); } else { xlog ("L_INFO","t_relay() returned ok"); } } onreply_route[35] { xlog ("L_INFO","Got some sort of reply"); if (t_check_status("603")) { xlog ("L_INFO","got a decline"); t_reply("603","Declined"); } return; }

18 years, 2 months

[Users] Checking reply status

by Douglas Garstang

How can I check for SIP responses, say 'Decline' for example? Do I have to use t_check_status()? I tried but I think it wants me to create a transaction first, and seeing as though the OpenSER docs are so bad, I don't know how to do this. Is there a stateless way to check for a SIP response? In general, it seems as though you can't check SIP response messages. There's the 'if method = <something>' approach, say for example 'if method == INVITE', but I have not seen any examples of how you can check for SIP response messages just as easy in a stateless fashion. Say, something like 'if response == DECLINE' or 'if response == 'Not Found'. How can this be done??? Thanks, Doug.

18 years, 2 months

[Users] TLS DH cipher problem

by Tóth Péter

Helo! I find a list, where the same problem occured with EDH-RSA-DES-CBC-SHA cipher... http://marc.theaimsgroup.com/?l=openssl-users&m=109844857431850&w=2 The offered solution is the following: "You need to include a call to SSL_CTX_set_tmp_dh() which the DH parameters too." How can I include the above? Thans: Tóth Péter

18 years, 2 months

[Serusers] SDP of 180

by Ever Zalazar

Hi people. I want to know if it's possible to configure SER to modify the SDP of the 180. Best Regards Ever Zalazar

18 years, 2 months

[Serusers] SER and Asterisk on FOSDEM Brussels, 25 February 2006

by Arek Bekiersz

Hi, I just returned from FOSDEM, held today at ULB Free University, Brussels. It was a great event, helping to encourage growth of free software. You could see many designers and developers working to make Open Source better. I must say I "recharged my batteries", raising my spirit a little. I joined SER presentation by Jan Janak, held at 14:00 at "Chavanne" room, building "U". Jan's appearance was recepted very warmly by audience. Presentation was professional and it fitted well in schedule, still leaving a lot of time for questions from the public. I must say that new SER logo (the big "S") looked very good on the big screen, during presentation. Jan presented origins of SER project, overview of architecture and functionality and also few typical SER installation scenarios. For those knowing less about SER or willing to learn it was a best primer. He also presented answers for frequent questions like "what SER is and what is not" and what can be done to get additional functionalities. There is a photograph attached to this letter. It's was taken with 3G phone but you can still see Jan on the right, with one of his slides. :-) After SER there was Asterisk presentation and workshop, held by Mark Spencer and his colleagues. He's a great developer too and in addition he revealed his showman skills, adding many jokes for the audience to laugh at. I was particularly disarmed by funny voice prompts with Asterisk's IVR voice of Alice, although one could found some them a little abrupt :-) -- Regards, Arek Bekiersz mailto:arek@perceval.net

18 years, 2 months

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sr-users February 2006