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[Users] opeser, webinterface
by ram 05 Jun '06

05 Jun '06

Hi does any one can point me web interface for openser ram

2 1

[Users] please help me...........................................!!!!!!!!!!!!!!!!!!!!!!!!!!!11
by raviprakash sunkara 05 Jun '06

05 Jun '06

Hi, Please help me in this. " udp_rcv_loop: probing packet received from 192.168.2.21 50195 ": means And Integrate the Radius+ mysql with Openser. In logined and call are made. acct_type isfailed... In Radacct table od Radius acctstoptime ,acctSessionTime,connectinfo_start,conecctinfo_stop , these entities are insert as null, But AcctStartTime is inserting. Below one is openser.cfg file please help me #************************************************************************************************* modparam("usrloc", "db_mode", 2) modparam("auth_db", "password_column", "password") modparam("auth_db", "calculate_ha1", yes) modparam("usrloc|acc|auth_db|group|msilo", "db_url", " mysql://openser:openserrw@localhost/openser") ########333333 modparam("acc","log_level",1) modparam("acc","log_flag",1) modparam("acc","log_missed_flag",2) modparam("acc", "log_fmt", "cdfimorstup") modparam("acc", "failed_transaction_flag",3) modparam("acc", "report_cancels", 1) modparam("acc","report_ack",0) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc","radius_flag",1) modparam("acc","radius_missed_flag",2) modparam("acc","service_type",16) modparam("acc","radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf") ##########33 modparam("nathelper","natping_interval",30) modparam("nathelper","ping_nated_only",1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") #################33 modparam("rr", "enable_full_lr", 1) ############################################### route { # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("13")) { sl_send_reply("483","Too Many Hops........................!"); exit; }; if ( msg:len > max_len ) { 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 record_route(); # loose-route processing if (loose_route()) { t_relay(); setflag(1); exit; }; # account completed transactions via syslog setflag(1); setflag(2); if(uri==myself) { if(method=="REGISTER") { if(!radius_www_authorize("192.168.2.55")) { www_challenge("192.168.2.55","1"); exit; }; save("location"); if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; exit; }; if(method=="INVITE"|| method=="ACK") { acc_rad_request("Start"); record_route(); force_rtp_proxy(); #setflag(1); t_on_reply("1"); }; if (method=="BYE") { record_route(); }; if (method=="MESSAGE") { log(1, "MESSAGE\n"); setflag(1); /* set for accounting (the same value as in log_flag!) */ }; if (method=="BYE" || method=="CANCEL") { #log (1, "BYE or CANCEL\n"); #setflag(1); acc_rad_request("Stop"); setflag(1); }; if(!lookup("location")) { sl_send_reply("404","Woo......... NOt found"); }; }; lookup("aliases"); setflag(2); if(!t_relay()) { sl_reply_error(); }; } onreply_route[1] { if(status=~"[0-9][0-9][0-9]") { force_rtp_proxy(); }; } -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com <b>

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RE: [Users] Avpops tutorial
by Douglas Garstang 05 Jun '06

05 Jun '06

I'm going to have to side with Sam. I've read the avpops tutorial more times than I can remember, and it doesn't make much more sense than it did six months ago. The tutorial is bad. > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Friday, June 02, 2006 1:42 AM > To: Sam Lee > Cc: users(a)openser.org > Subject: Re: [Users] Avpops tutorial > > > Sam, > > the tutorial explains also each module parameter and functions - just > read it completely from the beginning. > > regards, > bogdan > > Sam Lee wrote: > > > Hi everyone, > > > > I'm a newbie when it comes to AVPops, so please bear with me. > > > > Am looking at the avpops tutorial , when i came across this web : > > http://www.voice-system.ro/docs/avpops/ar01s08.html#ex_bitmap_acl > > > > It teaches you how to use make use of avpops to create a > bitmap acl. > > For the whole of the website, nothing was done to explain > to you what > > each and every of those commands and the values in it are > for. It just > > doesn't make sense to me ?! > > For example , they said place this in :- > > > > > modparam("avpops","db_scheme","scheme0:username_col=username;d > omain_col=domain;value_col=acl;value_type=integer;table=subscriber") > > modparam("avpops","avp_aliases","acl=i:800") > > > > Can i ask what these 2 lines are talking about ??! > > Please help explain to me how to use that bitmap acl ! > Anything that > > we need to change at the database? > > Thanks alot! > > > > REgards, > > Sam > > > >------------------------------------------------------------- > ----------- > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

4 3

Re: [Users] Openser+Asterisk Voice mail
by Glenn Dalgliesh 05 Jun '06

05 Jun '06

Post your openser.cfg --------------------- Sent with ChatterEmail True push email for the Treo Smartphone www.chatteremail.com -----Original Message----- From: ram <talk2ram(a)gmail.com> Date: Sunday, Jun 4, 2006 11:46 am Subject: [Users] Openser+Asterisk Voice mail Hi Just started Learning OPENSER, installing at my office Now iam able to install and configured with my VoIP provider and iam able to make calls out, using Public IP, not tried yes NAT when i want to voice mail, openser, send email, But iam looking store and retrieve voice message, so i decided to install asterisk and send voice messages to Asterisk as per the site document i have configured iam running OPENSER 5060 port and Asterisk 5090 port on same server As guided document, maded Realtime config with my mysql, working Fine iam able to see both are contacting when i restart sip reload , so i able to see openser log, and console of asterisk Now the problem, when i call extention which is not registered I get fast busy tone, and i dont see any message in Asterisk console, but in the open ser log i can see User Not found can some one help me to achive this but when i dial *86, i can hear voice, saying that no messages, that means asterisk is working any help will be appriciated ram ------=_Part_20143_5639206.1149439588939 Content-Type: text/html; charset=ISO-8859-1 Content-Transfer-Encoding: 7bit Content-Disposition: inline <div>Hi</div> <div> </div> <div>Just started Learning OPENSER, installing at my office</div> <div> </div> <div>Now iam able to install and configured with my VoIP provider</div> <div> </div> <div>and iam able to make calls out, using Public IP, not tried yes NAT</div> <div> </div> <div>when i want to voice mail, openser, send email, </div> <div> </div> <div>But iam looking store and retrieve voice message, so i decided to install asterisk</div> <div>and send voice messages to Asterisk</div> <div> </div> <div>as per the site document i have configured</div> <div> </div> <div>iam running OPENSER 5060 port and Asterisk 5090 port on same server</div> <div> </div> <div>As guided document, maded Realtime config with my mysql, working Fine</div> <div> </div> <div>iam able to see both are contacting when i restart sip reload , so i able to see openser log, and console of asterisk</div> <div> </div> <div>Now the problem, when i call extention which is not registered</div> <div> </div> <div>I get fast busy tone,  and i dont see any message i[Users] Openser+Asterisk Voice mailram <talk2ram(a)gmail.com>To: users(a)openser.org

2 2

[Users] Openser on multi-homed server
by Simon Morvan 05 Jun '06

05 Jun '06

Hello there, We currently have the following configuration. An openser is listening on two interfaces. One is linked to an internal private network, on which users (sipphones) are. The other is link to a public network. We noticed several issues : - When an INVITE (for exemple) comes from the private network and has to be relayed to the public network, openser put the private address as the source of the outgoing message. (sendto() use the same socket from which the message came from ?) - To solve this issue, we set up openser to listen on 0.0.0.0 (only one socket) but with this config, openser was unable to put the correct address into 'Record-Route' headers. (i.e. Record-Route: <sip:0.0.0.0:5060> !) Is there a specific way of setting up openser in a multi-homed situation ? Thanks in advance. -- Simon Morvan

3 4

Re: [Users] SIP Invite Header Modification
by Bogdan-Andrei Iancu 05 Jun '06

05 Jun '06

Gene, subst() works only on the header, so the first line is not covered by this function. subst_uri() works only on the RURI (from the first line). so try: subst_uri("/sip:(.*)$/18888888888@xxx.xxx.xxx.xxx/") To header is not recommended to be changed since it's used to transaction matching - anyhow it has no purpose in routing. regards, bogdan Gene Cohen wrote: >Bogdan, > >I have no problem changing to To: but the INVITE doesn't match evn though I >use the same code. >I tried subst and subst_uri - here is the code: > > > if (subst("/^INVITE(.*)$/INVITE sip:18888888888@xxx.xxx.xxx.xxx;/") ) > > { > log("URISUBSTITUTED"); > > } > else > { > log("INVITE NOTSUBSTITUTED"); > > } > > > if (subst("/^To(.*)$/To: sip:18888888888@xxx.xxx.xxx.xxx;/") ) > > { > log("TO SUBSTITUTED"); > > } > else > { > log("TO NOT SUBSTITUTED"); > > } > > > >Thanks, >Gene > > >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: Friday, June 02, 2006 12:39 PM >To: Gene Cohen >Cc: users(a)openser.org >Subject: Re: [Users] SIP Invite Header Modification > > >Hi Gene, > >As I see, you actually want to change the first line of INVITE and not >the headers..for changing the ruri you have the subst_uri() function >(regexp based) in textops module: > http://openser.org/docs/modules/1.0.x/textops.html#AEN141 >make a regexp to get rid of the ruri params. > >regards, >bogdan > >Gene Cohen wrote: > > > >>I am working with a VoiceGenie SIP VXML server which is very sensitive >>to SIP headers on INVITE - if the format is not just so, it cannot >>handle the calls. >> >>I want to use openser to mask this problem by sending a consistent SIP >>INVITE that the VoiceGenie can handle. >> >>Does anyone have any advice on how to do this? >> >> >>This (simple) INVITE header works : >> >> INVITE sip:7035470041@69.60.182.155:5060 SIP/2.0 >> >>none of these work : >> >> INVITE sip:7035470041@69.60.182.155:5060;dtg=SIP SIP/2.0 >> INVITE sip:7035470041;npdi=yes;@69.60.182.155:5060;dtg=SIP SIP/2.0 >> INVITE sip:7035470041;cic=5119;@69.60.182.155:5060;dtg=SIP SIP/2.0 >> INVITE >> sip:7035470041;cic=5119;npdi=yes;@69.60.182.155:5060;dtg=SIP >> SIP/2.0 >> >> >> >> thanks, >> >> gene >> >>----------------------------------------------------------------------- >>- >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> >> > > > > >

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RE: [Users] Avpops tutorial
by Douglas Garstang 04 Jun '06

04 Jun '06

I'm not one of the developers. -----Original Message----- From: Juha Heinanen [mailto:jh@tutpro.com] Sent: Fri 6/2/2006 11:58 PM To: Douglas Garstang Cc: Bogdan-Andrei Iancu; Sam Lee; users(a)openser.org Subject: RE: [Users] Avpops tutorial Douglas Garstang writes: > I'm going to have to side with Sam. I've read the avpops tutorial > more times than I can remember, and it doesn't make much more sense > than it did six months ago. The tutorial is bad. perhaps you can write a better one. -- juha

2 1

[Serusers] openser+asterisk+voicemail
by ram 04 Jun '06

04 Jun '06

Hi Just started Learning OPENSER, installing at my office Now iam able to install and configured with my VoIP provider and iam able to make calls out, using Public IP, not tried yes NAT when i want to voice mail, openser, send email, But iam looking store and retrieve voice message, so i decided to install asterisk and send voice messages to Asterisk as per the site document i have configured iam running OPENSER 5060 port and Asterisk 5090 port on same server As guided document, maded Realtime config with my mysql, working Fine iam able to see both are contacting when i restart sip reload , so i able to see openser log, and console of asterisk Now the problem, when i call extention which is not registered I get fast busy tone, and i dont see any message in Asterisk console, but in the open ser log i can see User Not found can some one help me to achive this but when i dial *86, i can hear voice, saying that no messages, that means asterisk is working any help will be appriciated ram

1 0

[Users] Openser+Asterisk Voice mail
by ram 04 Jun '06

04 Jun '06

1 0

Re: [Users] Openser+Asterisk Voice mail
by Glenn Dalgliesh 04 Jun '06

04 Jun '06

Will look at it closer tomarrow sometime but a ngrep of the sip traffic would tell us if ser or asterisk giving user not found. --------------------- Sent with ChatterEmail True push email for the Treo Smartphone www.chatteremail.com -----Original Message----- From: "Glenn Dalgliesh" <glenn(a)routerboy.com> Date: Monday, Jun 5, 2006 7:36 pm Subject: Re: [Users] Openser+Asterisk Voice mail ok remove the route(1); and replace t_relay(); in the failure_route[2]; section as below failure_route[2] { if(!t_was_cancelled()) { revert_uri(); rewritehostport("asterisk-ip-voicemail:5090"); append_branch(); #PREVENT SOME CRAZY VOICEMAIL LOOP xlog("L_INFO", "INFO: CALL TO VOICEMAIL"); setflag(10); t_relay(); } ----- Original Message ----- From: ram To: Glenn Dalgliesh Cc: users(a)openser.org Sent: Monday, June 05, 2006 1:11 AM Subject: Re: [Users] Openser+Asterisk Voice mail Hi here is my config iam able to dial *86, i get voice message that no voice messages But the call rewriting when the user not available it should go to asterisks voice mail ram [root@sert openser]# more openser.cfg # # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) log_facility=LOG_LOCAL7 fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/group.so" loadmodule "/usr/lRe: [Users] Openser+Asterisk Voice mail"Glenn Dalgliesh" <glenn(a)routerboy.com>To: "ram" <talk2ram(a)gmail.com> Cc: users(a)openser.org

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