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[Users] Error reading radius dictionary - Config Openser - radiusclient - freeradius
by Alberto Pacifico 22 Aug '06

22 Aug '06

I like to connect Openser-1.0.1 with Freeradiuus and radiusclient-ng. I'm following the instructions of: http://www.openser.org/docs/openser-radius-1.0.x.html But when I'm starting the Openser-1.0.1 and the modules (like acc) is being initialized, some errors appears related with " error reading radius dictionary". Thanks Alberto __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Users] loading modules PA and JABBER fails
by Chris Robson 22 Aug '06

22 Aug '06

No matter what version of openser I build the PA and Jabber modules will fail when loading up modules in Openser start up. Both modules compile without error. Interesting enough is both these modules do not get installed by default with "make install", so I have to install them by hand. Anyone now what I'm missing or what isnt working. loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.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/avpops.so" loadmodule "/usr/local/lib/openser/modules/dbtext.so" loadmodule "/usr/local/lib/openser/modules/diversion.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/enum.so" loadmodule "/usr/local/lib/openser/modules/flatstore.so" loadmodule "/usr/local/lib/openser/modules/gflags.so" loadmodule "/usr/local/lib/openser/modules/mangler.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/pa.so" loadmodule "/usr/local/lib/openser/modules/options.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" loadmodule "/usr/local/lib/openser/modules/speeddial.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/exec.so" loadmodule "/usr/local/lib/openser/modules/group.so" loadmodule "/usr/local/lib/openser/modules/msilo.so" loadmodule "/usr/local/lib/openser/modules/textops.so" #oadmodule "/usr/local/lib/openser/modules/jabber.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so"

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[Users] nat-rtpproxy.cfg
by Mark Kent 22 Aug '06

22 Aug '06

A few months ago I downloaded example config files, as a way to learning openser, but I'm not 100% sure where I got them. One of them has this header: # $Id: nat-rtpproxy.cfg 9 2005-08-19 15:30:55Z /CN=Greger V. Teigre/emailAddress=greger(a)onsip.org $ I just found (re-found?) at onsip.org->downloads->Getting Started, a later version of nat-rtpproxy.cfg with this header: # $Id: nat-rtpproxy.cfg 51 2006-01-31 13:28:04Z /CN=Paul Hazlett/emailAddress=paul(a)onsip.org $ and they appear different only in the loose_route handling: OLD: if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; NEW: if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; Now, who wants to take a stab at explaining why this was changed? Why was ACK checked for in the old version and not the new? (my guess: zero length message body messes with force_rtp_proxy) Why does that INVITE logic appear in the new loose_route handler? What case does that handle that the later "if (method == INVITE)" doesn't handle? Does this have something to do with re-INVITES? For convenience, the two files are here: http://a.mainstreet.net/nat-rtpproxy.5.0.cfg http://a.mainstreet.net/nat-rtpproxy-2006.cfg And do we openser people have our own set of sample config files somewhere on openser.org? Thanks, -mark

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[Users] openSER and SEMS: pb with voicemail when phone is busy or on no answer
by Christophe Irles 22 Aug '06

22 Aug '06

Hi, I want to have voicemail feature when phone is busy or on no answer My configuration file is like that: .. # route block used only to have voicemail if phone is busy or on no answer route[6] { append_branch(); avp_db_load("$ruri", "$avp(email)/$email_scheme"); if(!t_write_unix("/tmp/sems_socket", "voicemail/voicemail_headers")) { t_reply("500", "error contacting sems"); return; } route(5); route(4); route(1); } failure_route[1] { # flagset 26 <=> if phone busy we want a voicemail if ( (isflagset(26)) && (t_check_status("486")) ) { resetflag(26); route(6); return; } # flagset 27 <=> on no answer we want a voicemail if ( (isflagset(27)) && (t_check_status("408")) ){ resetflag(27); route(6); return; } end_media_session(); } Now if I use a "default.template" (used by the voicemail module) with the line: from:voicemail@%domain%, SEMS cannot send correctly a email: (3988) INFO: connect (AmSmtpClient.cpp:96): 127.0.0.1 welcomes us (3988) DEBUG: send_line (AmSmtpClient.cpp:173): SENDED: HELO 127.0.0.1 (3988) DEBUG: read_line (AmSmtpClient.cpp:145): RECEIVED: 250 christophe.rv.fr (3988) DEBUG: run (AmMail.cpp:119): Mail deamon starting its work (3988) DEBUG: send_line (AmSmtpClient.cpp:173): SENDED: mail from: <voicemail(a)172.20.250.1> (3988) DEBUG: read_line (AmSmtpClient.cpp:145): RECEIVED: 501 Bad address syntax SEMS is trying to send a mail from "voicemail(a)172.20.250.1" ! The SMTP server doesn't accept an IP and this IP is the IP of the callee SIP device . Strange. It's important to notice that "voicemail" works fine when is used on a phone not registered with the line "from:voicemail@%domain%" in file "default.template" . In this case the SEMS log file is: (4494) INFO: connect (AmSmtpClient.cpp:96): 127.0.0.1 welcomes us (4494) DEBUG: send_line (AmSmtpClient.cpp:173): SENDED: HELO 127.0.0.1 (4494) DEBUG: read_line (AmSmtpClient.cpp:145): RECEIVED: 250 christophe.rv.fr (4494) DEBUG: run (AmMail.cpp:119): Mail deamon starting its work (4494) DEBUG: send_line (AmSmtpClient.cpp:173): SENDED: mail from: <voicemail(a)christophe.rv.fr> (4494) DEBUG: read_line (AmSmtpClient.cpp:145): RECEIVED: 250 Ok So I suspect the "%domain%" in "default.template" file is not correctly read by the module "voicemail" after a "failure_route" or after a "append_branch". What I must change in my OpenSER configuration file in order to resolve this problem ? NOTE: I'm using sems-0.1.1 in revision 68 Thanks in advance, Christophe

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[Users] Sample config
by Pavel D. Kuzin 22 Aug '06

22 Aug '06

Hello! Can anyone provide sample config to proxy any calls. Thank you. -- Pavel D.Kuzin System Administrator Nodex ISP St. Petersburg, Russia pk(a)nodex.ru http://nodex.ru

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[Users] SMS Module (SIP to GSM)
by ade darmawan 22 Aug '06

22 Aug '06

Hi all, please help me.... Nowadays, i've been trying to implemented the SMS Module (SIP to GSM) using OpenSER 1.0.0, but never the lles, the tutorials that had given at www.openser.org are so poor... Could you please help me to find more added tutorial guide to implemented the SMS Module (SIP to GSM) ? I wanna know, what kind of facilities that needed to build this module and make it run correctly (such as DNS,etc), step by step to build this module? i hope your opinion? --------------------------------- Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.

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[Serusers] Is there a function for lookup contents in a given column of the subscriber table?
by Ricardo Carvalho 22 Aug '06

22 Aug '06

Hi, Is there any Ser function for lookup contents in a given column of the subscriber table, like the *is_user_in* function that looks up in group table? Thanks, Ricardo.

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[Users] default behaviour
by Mark Kent 22 Aug '06

22 Aug '06

As an educational tool, I'm trying to write a pure proxy using openser. This proxy doesn't have users or domains or gateways, it just shuffles messages that happen to land on it. I'm starting with REGISTER where I have simply this: t_on_reply("10"); t_on_failure("10"); t_relay(); The onreply and onfailure callbacks just log stuff. The openser cookbook says: The implicit action after execution of the main route block is to drop the SIP request. To send a reply or forward the request, explicit actions must be called inside the route block. To find what happens in other route blocks (onreply, etc.), I searched the cookbook for "implicit" and learned: implicit action for branch_route is to forward the request. implicit action for onreply_route is to send the reply upstream according to Via header And that's all there is. Is there more written somewhere else? What I noticed is that informational responses (1xx) are not relayed by openser, but Final (200) or Client Error (401) are relayed. Is this conclusion correct? This implicit behaviour seems a little odd to me; I would think that if I've got a t_on_reply out there then I'm saying "Hey, I'll handle it" but in fact I cannot handle it (t_relay is not allowed in an onreply block). So, I must be missing something in my understanding... I'ld appreciate it if someone could clarify for me. Thanks, -mark

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[Users] How do I only INVITE if I am registered?
by Mahesh Paolini-Subramanya 22 Aug '06

22 Aug '06

Consider the case where there are two fones, both registered as the same subscriber (foo(a)bar.com) I have max_contacts == 2, therefore the fones actually register I now add a *third* fone, also registered as foo(a)bar.com Since max_contacts == 2, this fone does *not* register. However, if a user makes a call from this fone, the call will go through. I wish to prevent this. What do I do? Something obvious that I am missing? Or something weird with twiddling with the RURI and using "registered(domain)"? Or heavy AVP mucking around? Cheers -- ******************************************* Mahesh Paolini-Subramanya (703) 386-1500 x9100 CTO mahesh(a)aptela.com Aptela, Inc. http://www.aptela.com "Aptela: How Business Answers The Call" *******************************************

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RE: [Users] Rewriting Contact HF port
by Justin Pearce 21 Aug '06

21 Aug '06

Thanks, Now that I understand what the received avp is for in the registrar module, that seems to make a lot more sense. I was unaware at the time how to go about doing that, and with the small number of replies I got, the only logical conclusion was to modify some functions that performed similarly that I knew worked 100% (all this because of a deadline). The information is helpful, since I will likely have to use it in a future implementation. Justin Pearce Information Technology/Programming Price Video Productions JustinP(a)PriceVideo.com 361-572-3810 800-733-3810 Fax: 361-572-3894 www.PriceVideo.com -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Monday, August 21, 2006 9:04 AM To: Justin Pearce Cc: users(a)openser.org Subject: Re: [Users] Rewriting Contact HF port Justin, just as an idea, you could upload the contact into an AVP (via pseudo variables) and then use a regexp to change the port part....so, you can do it only via script functions. regards, bogdan Justin Pearce wrote: >Hi Bogdan, > >I knew that Nathelper did some rewriting, but the circumstances required >that the thing record the private IP and a certain known port instead of >the public IP as a contact address (using fix_nated_register() to save >the other address). I wound up borrowing the code from the Nathelper >module and modifying it to accept a port argument and made it a separate >module. This way, it only rewrote the port and not left everything else >untouched. > >Thanks. > >Justin Pearce >Information Technology/Programming >Price Video Productions >JustinP(a)PriceVideo.com >361-572-3810 >800-733-3810 >Fax: 361-572-3894 >www.PriceVideo.com > >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: Monday, August 21, 2006 6:19 AM >To: Justin Pearce >Cc: Klaus Darilion; users(a)openser.org >Subject: Re: [Users] Rewriting Contact HF port > >Hi Justin, > >if you need save(location) to save as "received" a custom uri, you just >need to set an avp with the uri you need to be saved. See: > http://openser.org/docs/modules/1.1.x/registrar.html#AEN211 >It's exactly what nathelper does. > >regards, >bogdan > >Justin Pearce wrote: > > > >>Thanks, Klaus >> >>It looks like I'll have to make something similar to the nathelper >>module to do what I need. >> >>Justin Pearce >> >>-----Original Message----- >>From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] >>Sent: Monday, August 07, 2006 1:43 AM >>To: Justin Pearce >>Cc: users(a)openser.org >>Subject: Re: [Users] Rewriting Contact HF port >> >>Justin Pearce wrote: >> >> >> >> >>>I am wondering if there is a good way to rewrite the port in the >>> >>> >>> >>> >>contact >> >> >> >> >>>header in OpenSER 1.0.1 >>> >>> >>> >>> >>1. Why do you want to rewrite the port to fixed port? NAT problems? >> >> >> >> >> >>>before it is stored in the location table?(I'm using the MySQL >>> >>> >>> >>> >>database >> >> >> >> >>>backend). >>> >>>Thus far, I am using "subst(/^(Contact.+):[0-9]+(.+)$/\1:5060\2/ig);" >>>just before I do >>> >>>"save("location");", but that does not seem to be working. >>> >>> >>> >>> >>subst will be applied before the message is forwarded. Thus, save() >> >> >sees > > >>the original Contact header. You either loop the modified message to >> >> >the > > >>SIP proxy again or you write your own function (similar to the function >> >> > > > >>in nathelper module) >> >>regards >>klaus >> >> >> >> >> >>>Suggestions? >>> >>> >>> >>>Justin Pearce >>> >>>Information Technology/Programming >>> >>>Price Video Productions >>> >>>JustinP(a)PriceVideo.com <mailto:JustinP@PriceVideo.com> >>> >>>361-572-3810 >>> >>>800-733-3810 >>> >>>Fax: 361-572-3894 >>> >>>www.PriceVideo.com >>> >>> >>> >>> >>> >>> >>> >>> >>> >>---------------------------------------------------------------------- - >> >> >- > > >> >> >> >> >>>_______________________________________________ >>>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 >> >> >> >> >> > > >

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