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37548 discussions

[Users] Problem with avp_db_load in OpenSER 1.2
by Daryl Sanders 29 Apr '07

29 Apr '07

I cannot seem to get my usr_preferences to load using the following load statement. avp_db_load("$avp(s:caller_username)", "*"); I have verified with xlog and avp_print() that $avp(s:caller_username) does contain a valid value before calling avp_db_load, however nothing is returned. It will work if I use this: avp_db_load("$fu/username", "*"); I really need to make it work via $avp(s:caller_username) if possible because my call forwarding and custom call routing features change this value often. Thanks, - Daryl

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[Users] Openser Summit 2007
by Flavio Goncalves 29 Apr '07

29 Apr '07

Hello, Is there any plans for an OpenSER summit 2007? Flavio.

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[Serusers] problem with MySQL
by Tuan Nguyen 29 Apr '07

29 Apr '07

Hi everyone, + I have problem with mysql when run ser: WARNING: no fork mode and more than one listen address found (will use only the the first one) 0(12395) init_tcp: using epoll_lt as the io watch method (auto detected) 0(12395) Maxfwd module- initializing 0(12395) new_connection: Access denied for user 'root'@'localhost' (using password: YES) 0(12395) gflags:mod_init: Error while connecting database 0(12395) init_mod(): Error while initializing module gflags ERROR: error while initializing modules in shell, when i run mysql -u root -pxxxxx, it works. but when run ser as root, it don't work, i tried to fix this problem 2 days but can't. can someone help me and tell me where is problem. + second question: in serusers archive don't have search box, when I want to search someone has the same problem like me and know how they solve it, i have to look every threads or we have other way to search efficient, I tried to search with google group but can't. thank you very much for your response. cheers Tuan.

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[Users] new CDRTool release 5.1.1
by Adrian Georgescu 29 Apr '07

29 Apr '07

The software can be downloaded from: http://download.dns-hosting.info/CDRTool/ Changelog from 5.0.10: cdrtool (5.1.1) unstable; urgency=low * Updated sample data for rating tables * Updated mysql schema * Fixed bug in csv import script cdrtool (5.1.0) unstable; urgency=low * Added historical rating engine, requires changes in database structure (you must apply changes from setup/mysql/ alter_tables.mysql) * Must add in global.inc "AccountsDBClass" => "DB_ser" for each SER data-source, there is no default anymore * Added function to obfuscate caller Ids, this changes the login rights The login accounts rights must be updated in the Accounts page * Added capability to do accounting for SIP URIs (e.g. paid helpdesk) * Improved compatibility with Firefox in media sessions screen * Added rights for login accounts to show pricing and caller id * Removed unused NGN rating (billing_customers table has been modified, please update your CSV file import tools) * Modify index of rates tables to accomodate multiple applications (you must apply changes from setup/mysql/alter_tables.mysql) * Default application set to 'audio', application field is now mandatory when importing rates CSV files * Use separate name spaces for memcache keys per data source * Allow quota checks for multiple OpenSER data-sources * Show memory usage in syslog * Print sql errors in syslog Regards, Adrian Georgescu

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[Users] A problem wiht openseradmin0.3--Add New Account Preference
by Lee Jacky 29 Apr '07

29 Apr '07

Hi,everyone,I am jacky.I use openser1.1+openseradmin0.3. Now I am troubled with a problem,and here is the detail: After I install the openseradmin, I add attribute Tel to the account_preferences_type,and then I click the link:Add New Account Preference. I fill the form like this: username:admin domain:voip.net attribute:Tel values:12456 Then I click the button:Create, and it redirect to a error page,whose content is as follows: ArgumentError in Voip account preferencesController#create wrong number of arguments (1 for 2) RAILS_ROOT: /usr/local/openseradmin-0.3/script/../config/.. Application Trace | Framework Trace | Full Trace /usr/local/ruby/lib/ruby/gems/1.8/gems/activerecord-1.15.1/lib/active_record/base.rb:1700:in `attribute=' /usr/local/ruby/lib/ruby/gems/1.8/gems/activerecord-1.15.1/lib/active_record/base.rb:1700:in `send' /usr/local/ruby/lib/ruby/gems/1.8/gems/activerecord-1.15.1/lib/active_record/base.rb:1700:in `attributes=' /usr/local/ruby/lib/ruby/gems/1.8/gems/activerecord-1.15.1/lib/active_record/base.rb:1699:in `each' /usr/local/ruby/lib/ruby/gems/1.8/gems/activerecord-1.15.1/lib/active_record/base.rb:1699:in `attributes=' /usr/local/ruby/lib/ruby/gems/1.8/gems/activerecord-1.15.1/lib/active_record/base.rb:1532:in `initialize_without_callbacks' /usr/local/ruby/lib/ruby/gems/1.8/gems/activerecord-1.15.1/lib/active_record/callbacks.rb:225:in `initialize' #{RAILS_ROOT}/app/controllers/voip_account_preferences_controller.rb:27:in `new' #{RAILS_ROOT}/app/controllers/voip_account_preferences_controller.rb:27:in `create' Request Parameters: {"voip_account_preference"=>{"username"=>"admin", "domain"=>"voip.net", "value"=>"12456", "attribute"=>"Tel"}, "commit"=>"Create"} I have considerde it for a long time ,but I still can't solve it, so I hope you can help me to give me some advice, Any suggestion will be appreciated. Best wishes, Yours, Jacky ___________________________________________________________ 雅虎免费邮箱3.5G容量，20M附件！ http://cn.mail.yahoo.com/

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[Users] WeSIP with openser 1.2.0 again
by tele 28 Apr '07

28 Apr '07

Hi, fresh install of OpenSER 1.2.0 and wesip-05_03_07-beta-0.1.3.tar.gz. i'm able to run wesip and successful deploy my app, but when the first INVITE come into wesip i get: java.lang.StringIndexOutOfBoundsException: String index out of range: -37 at java.lang.String.checkBounds(String.java:398) at java.lang.String.<init>(String.java:443) at com.voztele.javax.sip.header.ExtensionHeaderImpl.<init>(ExtensionHeaderImpl.java:40) at com.voztele.javax.sip.message.SIPMessage.createSIPHeader(SIPMessage.java:243) at com.voztele.javax.sip.message.SIPMessage.addSIPHeader(SIPMessage.java:230) at com.voztele.javax.sip.message.SIPMessage.<init>(SIPMessage.java:225) at com.voztele.javax.sip.message.SIPRequest.<init>(SIPRequest.java:154) at com.voztele.javax.sip.parser.StringMsgParser.parseSERMessage(StringMsgParser.java:355) at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:136) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Thread.java:619) Exception in thread "ExpressMChannel[2]" java.lang.NullPointerException at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:143) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Thread.java:619) I'm using JDK/JRE version 6. With old version of wesip and openser i've no problem. thanks :tele

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RE: [Users] WeSIP setup problem
by ventilo35＠free.fr 28 Apr '07

28 Apr '07

Hi Gines, thanks for your support > Your servlet named "neo" is being executed and looks like it is > proxying but what from what I see in the packet it is proxying to > itself. Could you try to proxy to another address and see if this > still happens ? Aside, could you send me your configuration files and > your servlet code ? The serlvet code is the following protected void doInvite(SipServletRequest callerInvite) throws ServletException, IOException { //Get a proxy for this request Proxy callerProxy = callerInvite.getProxy(); System.out.println("doInvite within neos servlet: requestURI is " + callerInvite.getRequestURI() ); callerProxy.proxyTo(callerInvite.getRequestURI()); } This code causes the "no route" exception. The println above gives : doInvite within neos servlet: requestURI is sip:5002@192.168.1.11 where 5002 is the called SIP phone and 192.168.1.11 is the address of my OpenSER + WeSIP box (i.e. the managed SIP domain) Following your advice I have tried to change the proxyTo to a different address: SipURI dest = sipFactory.createSipURI("5002","192.168.1.16"); callerProxy.proxyTo(dest); It that case, the called phone (5002, IP address 192.168.1.16) gets the INVITE correctly. However, something puzzles me here: the serlvet is not supposed to know the IP address of the called phone. In my understanding, the AS processes the INVITE, then it must send it back to the proxy server which is supposed to forward it to the next hop, is that right ? Could you please give me an example of a doInvite which basically does nothing and which lets the INVITE be forwarded to the called party ? Finally here are my config files: <<<<<<<<< Server.xml >>>>>>>>> <Server port="5005" shutdown="SHUTDOWN" debug="1"> <Service name="NEOS-Standalone"> <Connector className="org.apache.catalina.connector.http.HttpConnector" port="7080" minProcessors="5" maxProcessors="5" enableLookups="true" acceptCount="10" debug="0" /> <Connector className="com.voztele.sipservlet.connector.SipConnector" minProcessors="5" maxProcessors="5" enableLookups="true" acceptCount="10" debug="1" addresses="192.168.1.11:5060" pathName="com.voztele" protocol="udp" > <ExtraProperties> <Property key="com.voztele.javax.sip.SER_ADDRESS" value="127.0.0.1" /> <Property key="com.voztele.javax.sip.SER_PORT" value="5080" /> <Property key="com.voztele.STATISTICS" value="false" /> <Property key="javax.sip.STACK_NAME" value="neos_as" /> <Property key="javax.sip.THREAD_POOL_SIZE" value="10" /> <Property key="com.voztele.javax.sip.THREAD_POOL_SIZE" value="10" /> <Property key="com.voztele.javax.sip.SPIRAL_HDR" value="true" /> <Property key="com.voztele.javax.sip.REENTRANT_LISTENER" value="true" /> <Property key="javax.sip.AUTOMATIC_DIALOG_SUPPORT" value="false" /> <Property key="com.voztele.javax.sip.TRACE_LEVEL" value="3" /> </ExtraProperties> </Connector> <Engine name="Standalone" defaultHost="localhost" debug="1" className="com.voztele.sipservlet.core.ConvergedEngine"> <Mapper className="org.apache.catalina.core.StandardEngineMapper" protocol="HTTP/1.1"/> <Mapper className="org.apache.catalina.core.StandardEngineMapper" protocol="HTTP/1.0"/> <Mapper className="com.voztele.sipservlet.core.EngineSipMapper" protocol="SIP/2.0"/> <Logger className="org.apache.catalina.logger.FileLogger" timestamp="true"/> <Realm className="org.apache.catalina.realm.MemoryRealm" /> <Host name="localhost" nameSip="192.168.1.11_5060" debug="1" appBase="wesipapps" className="com.voztele.sipservlet.core.ConvergedHost"> <Logger className="org.apache.catalina.logger.FileLogger" timestamp="true"/> <Mapper className="com.voztele.sipservlet.core.SipHostMapper" protocol="SIP/2.0"/> <Mapper className="org.apache.catalina.core.HttpHostMapper" protocol="HTTP/1.1"/> <Mapper className="org.apache.catalina.core.HttpHostMapper" protocol="HTTP/1.0"/> </Host> </Engine> </Service> </Server> <<<<<<<<<<openser.cfg>>>>>>>>>>>>>> # # $Id: openser.cfg 1827 2007-03-12 15:22:53Z bogdan_iancu $ # # simple quick-start config script # Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes #log_stderror=no # (cmd line: -E) children=4 # Uncomment these lines to enter debugging mode #fork=no log_stderror=yes # listen = tcp:192.168.1.11:5060 listen = udp:192.168.1.11:5060 # ------------------ module loading ---------------------------------- #set module path mpath="/usr/local/lib/openser/modules/" # 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 "mi_fifo.so" # added for WeSIP loadmodule "seas.so" modparam("seas","listen_sockets","127.0.0.1:5080") # end added for WeSIP # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "auth.so" #loadmodule "auth_db.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- 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"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # 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(); # ftoutain essai de supprimer Ã§a ? ne change rien # 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); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); }; # 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("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); # exit; #}; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { if (is_present_hf("X-WeSIP-SPIRAL")) { if (!t_relay()) { sl_reply_error(); }; exit; }; if (! as_relay_t("neos_as") ){ t_reply("500", "NEOS service not found"); }; exit; }

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[Users] Re: Multiple invites
by raviprakash sunkara 28 Apr '07

28 Apr '07

Sorry Venkatesh an U me ..... to my mobile Am Not in hyd , I left My Company, In the mail list if U see the Klaus mail , he raply and U can do , if U knew the Avpops modules and pesudo attributes, http://www.openser.org/dokuwiki/doku.php/tutorials:avpops#serial_forking see that url , is the serial_forking , Do in the Same in parallel _ forking other call me , before on 2nd of the May. On 4/25/07, venkatesh.d <venkatesh.d(a)pyronetworks.com> wrote: > > hai raviprakash, > > i have tried for that multiple invites behind NAT and > Mediaproxy and calling from A(a)sipdomain.com to B(a)sipdomain.com and same > call have to forword to C(a)sipdomain.com . But C(a)sipdomain.com > invite message going to B system . > if you feel free can you send your openser.cfg file NATING > with out using media proxy and with multiple invites > > > Thanks and Regards > venkatesh.d > -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com

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[Users] Openser with Quintum voice gateway and an important load
by mailinglist-openser-users 27 Apr '07

27 Apr '07

Hi, Openser version: 1.2.0 We have 4 central Quintum CMS platform hosted in our computer room. We have several tens of Qintum voice gateway ( DX / AX / Call Relay ) on remote customer site. Our customer are mainly call centers so they generate a lot of small calls with an important number of call wich are never established ( around 50% ). The central platform is quite loaded, with around 600 simultaneous calls and around 20 new call per second. Up to now, we used H323 as the signalisation protocol. Now we are planning to migrate to SIP and use openser 1.2.0 for that. Does anybody already use openser with Quintum as voice gateway with so many calls ? I'd like to share experience with people using this kind of setup with this kind of load. There are no know problem between openser and Quintum SIP implementation ( we use firmware P104-12-16 ) ? Openser should be able to route all these call towards our 4 central gateway with LCR while as well sending radius accounting. Our main and backup SIP server are IBM X330 ( Bi-Pentium III 1266 Mhz with 2GB of memory ). Will it be enough to handle 600 simultaneous call and 20 new calls per seconds ? For radius accounting, we have 2 radius servers. It seems that when the first adius server is down, openser is trying 3 times the first radius server and then go to the second one but the children seems to be blocked during this time ( for example in debug mode, no more new call can be establised while radius accounting is not yet finished ). So should I start lots of children ( 20 or more ) so that if the first radius is down, openser will still be able to route new calls while some children are blocked for accounting reason ? As well I'm planning to use the "dialog" module, if this module largely increasing the load on openser ? Regards, Jerome. -- Jerome

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[Users] How can I pass pseudovariables to functions?
by Peter Lemenkov 27 Apr '07

27 Apr '07

Hello All! I have a question. If I declare variable like this: $var(ip_address) = "192.168.0.1"; I can't use this variable in functions like that one: revritehostport ($var(ip_address)); It leads me to the following message: omega1 ~ # /usr/local/sbin/openser -f /usr/local/etc/openser/openser_registrar.cfg -l 192.168.0.1 0(5723) parse error (316,25-37): syntax error 0(5723) parse error (316,37-38): bad argument, string expected ERROR: bad config file (2 errors) So the question is %subj%. -- With best regards!

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