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37464 discussions

[Users] pap2 ATA problem
by Jayesh Nambiar 30 May '06

30 May '06

Hello everybody, I have implemented few features in my openser script. The features are like if a user presses *86 followed by any number, that number will be his forwarded number when someone calls him and he is busy. Similarly the code for forwarding on no-answer is *87 followed by the number to which it should be forwarded. I've tested these features using Eyebeam softphone and Audiocodoes MP series analog gateways. But while testing it with a linksys pap2 device, it does'nt work. The problem is that as soon as the user dials *86 the phone gets disconnected and his forwarded number becomes blank in the table where i store these numbers. I dont understand how does the device recognises the code which is written in my openser script. Its like if the user dials any feature enabling code, before he enters the number the call is disconnected. Can anyone please help me in this regard or can anyone please guide me if they worked on this kind of idea before. the snippet of cfg is as below: if(uri=~"sip:\*87[0-9]*@") { if(isflagset(9)) { #--check if the feature is allowed for user-- log(1,"Entered *87 condition"); avp_db_query("select attribute from usr_preferences where username='$fU' and attribute='fwd_noanswer'", "$avp(i:91)"); if(!avp_check("$avp(i:91)", "eq/fwd_noanswer/i")) { strip(3); avp_db_query("insert into usr_preferences values('', '$fU', '', 'fwd_noanswer', '', '$ruri', '')"); log(1,"Entered in DB"); sl_send_reply("200", "OK"); exit; }; strip(3); avp_db_query("update usr_preferences set value='$ruri' where username='$fU' and attribute='fwd_noanswer'"); sl_send_reply("200", "OK"); exit; }; sl_send_reply("403", "Forbidden"); exit; }; Waiting for reply, w/regards, Jayesh --------------------------------- Send free SMS to your Friends on Mobile from your Yahoo! Messenger Download now

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[Serusers] No. of users to register behind single NAT device.
by ravi reddy 30 May '06

30 May '06

HI all , My SER is running fine i made some test calls between two ,three soft phones who stayed in different places but , i tried to register some soft phones who are staying behind same NAt device but i can only register only two phones and remaining i cannot what is the reason for this one is it a serious problem ? Please help me .. Thanks in advance, Regards Ravi

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[Users] uac_auth cseq workaround - ANY ?
by G.Jacobsen 30 May '06

30 May '06

Hello, I am trying to use the uac_auth function against an asterisk box and receive 488 not acceptable here. It appears that this is not due to a media problem since the client which is routed through openser can issue an authenticated invite without problems when registered directly with the asterisk box - with exactly the the same media settings. So I supect that this 488 message is due to the fact that openser does not increase the cseq during authentication causing asterisk to issue a 488 after the correct response to the challenge. What are my workaround options to authenticate openser against asterisk (or any other RFC compliant proxy) ? ANY practical hints would be highly appreciated. TIA for your help. Gerry

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[Users] Unknown value error
by Giuseppe Parlato 30 May '06

30 May '06

Hello is there a way to ignore an attribute value when I make an authorization request to MsSQL database? that value is not important I think and it creates an error: ERROR: radius_xlat: '122(a)domain.com' rlm_sql (sql): sql_set_user escaped user --> '122(a)domain.com' radius_xlat: 'VoIP_Radius '122(a)domain.com'' rlm_sql (sql): Reserving sql socket id: 16 radius_xlat: '' radius_xlat: 'VoIP_Radius '122(a)domain.com','', '','192.168.1.1',''' rlm_sql: Failed to create the pair: Unknown value VoIP for attribute Service-Type rlm_sql (sql): Error getting data from database radius_xlat: '' MsSQL authorization queries: authorize_check_query = "VoIP_Radius '%{User-Name}'" authorize_reply_query = "VoIP_Radius '%{User-Name}','%{Calling-Station-Id}', \ '%{Called-Station-Id}','%{NAS-IP-Address}','%{h323-conf-id}','VoIP'" I tried to remove value VoIP from authorization query but the problem doesn't expire. Giuseppe Thank you

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[Users] CVS Head breaks Serweb
by G.Jacobsen 30 May '06

30 May '06

CVS head seems to break in several places serweb: 0. no login possible into user menues. Authentication seems to work but then serweb log shows problems with apu_aliases and apu_acl application layers. Admin functions: 1. Accounting details dont show - "caller_deleted" field missing in acc table 2. user account details cannot be accessed in admin - I suspect problem with key index since user_preference tables is indexed by uuid whereas in every other table username is used. 3. Aliases cannot be added - serweb doesnt throw error but hangs Are these things which should be ammended in openser or rather in serweb ? I suspect that problems with serweb will increase while openser develops independently from SER. Perhaps the openser maintainers might like to consider to fork also a serweb version and put it on the openser cvs system ? Thank you for your interest in the matter. Gerry Send instant messages to your online friends http://uk.messenger.yahoo.com

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[Serusers] help!!!
by Nhadie 30 May '06

30 May '06

Hi All, How can I make sure that an extension would only log-in once? Just want to make sure no one is usingmy extensios only my clients. Please help! Thank You Regards, Ron ________________________________________________ Message sent using UebiMiau 2.7

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Re: [Users] libradiusclient-ng-so.2 cann't share
by raviprakash sunkara 30 May '06

30 May '06

Hi Olivier, Thanks for replying, In installed and search the libradiusclient-ng.s0.2 in /usr/local/lib/ is there ? I think problen in env . variable . I also set the env variable LB_LIBRARY_PATH ALSO...... WHAT IS THE PROBLEM is solve this problem. Bye. Cheers On 5/28/06, olivier.taylor <olivier.taylor(a)gmail.com> wrote: > > not "can't share" but "can't open shared odject file : no file or > directory" > > Will it means radiusclient-ng is not installed? > > Olivier > > raviprakash sunkara a écrit : > > Hi Users.. > > I have problem in openser while I doing the accounting module. > > In that while I started the openser server its gives as > > " load_module : can't open module < /usr/local/lib/openser/module/acc.so> > : libradiusclient-ng.so.2: can't open shared odject file : no file or > directory " > > And " parse error fail to load " > > > And also in enter the LD_LIBRARY_DIR in env. varible . > > please solve my issue > > > bye.. > > -- > Thanks and Regards with cheers > Sunkara Ravi Prakash (Voip Developer) > Hyperion Technology > www.hyperion-tech.com > <b> > > ------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com <b>

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[Users] IAS and Openser
by Artur Hayne 30 May '06

30 May '06

Hi, I have a big trouble. I have a voip server Openser (Debian) and a database Active Directory with IAS (Windows 2000 server). I have configured the openser to authenticated in IAS, but the openser dont authenticate. When I snif the information only appear lot of packets of Radius Request, a Authentication Request. Should I modify any field in the dictionary of radiusclient, because Im authenticatting in windows? The IAS shows me the mensage: "malformed packet". The IAS support Openser? --------------------------------- Yahoo! doce lar. Faça do Yahoo! sua homepage.

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[Users] pls help on installing and maintaining openser from cvs
by Padmaja RV 30 May '06

30 May '06

Hi all! Is anybody running openser through cvs? If yes, pls let me know the steps on how to start openser thru the init.d script. I am new to openser and I have gone thru the link in openser site for making the init.d script work with the monit daemon. now when i start openser with /etc/init.d/openser start or /etc/rc.d/init.d/openser start, the monit daemon takes over and tries to start openser. but apparently it is failing because in the web interface of monit, i see that the starting of openser failed. and there is no openser.pid file created in /var/run/. Pls help me on how to start openser through monit daemon or is there a work around to do this without involving the monit daemon? I am stuck at this point and unable to make any progress..... Thanks and Regards, Padmaja ----- Original Message ----- From: <users-request(a)openser.org> To: <users(a)openser.org> Sent: Tuesday, May 30, 2006 4:02 AM Subject: Users Digest, Vol 12, Issue 78 > Send Users mailing list submissions to > users(a)openser.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://openser.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request(a)openser.org > > You can reach the person managing the list at > users-owner(a)openser.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Re: callback type TMCB_REQUEST_IN (Bogdan-Andrei Iancu) > 2. Mixing uri_db and auth_db (Edgar Barbosa) > 3. RE:Problems starting openser (BrOtHeR StRaiN) > 4. uac_auth cseq workaround - ANY ? (G.Jacobsen) > 5. dispatcher module and rewritehost vs forward (Script Head) > 6. dispatcher module and rewritehost vs forward (Script Head) > 7. Unknown value error (Giuseppe Parlato) > 8. Problem : Forwarding Routine to a NAT'd endpoint. > (Ricardo Martinez) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 29 May 2006 16:04:26 +0300 > From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> > Subject: Re: [Users] callback type TMCB_REQUEST_IN > To: raviprakash sunkara <sunkara.raviprakash.feb14(a)gmail.com> > Cc: Users(a)openser.org > Message-ID: <447AF15A.9020901(a)voice-system.ro> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi, > > as the message says, you need to load TM module before the ACC one. > > regards, > bogdan > > raviprakash sunkara wrote: > >> Hi Users and lister... >> How R u Guys, >> I need help from U , The problem in openser when doing the accounting >> module. >> it give error as >> >> >> find_export:found <load_tm> in module tm >> [/usr/local/lib/openser.modules/tm.so] >> ERRoR:tm:register_tmcb: callback type TMCB_request_in register >> attempt before tm module initializaton >> ERROR :acc:mod_init;cannot register TMCB_REQUEST_IN callback >> init_mod() : Error while inittialization module acc >> >> >> what is means, What i need to do, And give any samle openser.cfg in >> accing with radius and mysql >> >> Bye.. >> -- >> Thanks and Regards with cheers >> Sunkara Ravi Prakash (Voip Developer) >> Hyperion Technology >> www.hyperion-tech.com <http://www.hyperion-tech.com> >> <b> >> >>------------------------------------------------------------------------ >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > > > > ------------------------------ > > Message: 2 > Date: Mon, 29 May 2006 16:16:41 +0100 > From: "Edgar Barbosa" <edgar.barbosa(a)madetowork.com> > Subject: [Users] Mixing uri_db and auth_db > To: <Users(a)openser.org> > Message-ID: <20060529151635.EA0816800E1(a)gw.madetowork.com> > Content-Type: text/plain; charset="us-ascii" > > Hi all, > > I'm having a weird behaviour when mixing auth_db and uri_db. > > With the following openser.cfg, the xlog("L_INFO", "TEST\n") is never > executed... seems that openser just hangs up... > > If I remove the uri_db it works... > If I remove the auth_db it works too... > The problem seems to happen when I mix both... > > Any idea of what may be wrong? > I'm using the latest version (openser 1.0.1). > > Thanks > > > > > debug=3 # debug level (cmd line: -dddddddddd) > fork=yes > log_stderror=yes > > reply_to_via=no > check_via=no # (cmd. line: -v) > dns=no # (cmd. line: -r) > rev_dns=no # (cmd. line: -R) > port=5060 > children=4 > fifo="/tmp/ser_fifo" > > > loadmodule "/usr/lib/openser/modules/mysql.so" > loadmodule "/usr/lib/openser/modules/sl.so" > loadmodule "/usr/lib/openser/modules/auth.so" > loadmodule "/usr/lib/openser/modules/auth_db.so" > loadmodule "/usr/lib/openser/modules/uri_db.so" > loadmodule "/usr/lib/openser/modules/xlog.so" > > modparam("uri_db", "db_url", "mysql://root@192.168.1.104/openser") > modparam("auth_db", "db_url", "mysql://root@192.168.1.104/openser") > > route { > > xlog("L_INFO", "TEST\n"); > > } > > > > > ------------------------------ > > Message: 3 > Date: Mon, 29 May 2006 10:13:05 -0700 (PDT) > From: BrOtHeR StRaiN <strain15(a)yahoo.com> > Subject: RE:[Users] Problems starting openser > To: users(a)openser.org > Message-ID: <20060529171305.6804.qmail(a)web38801.mail.mud.yahoo.com> > Content-Type: text/plain; charset="iso-8859-1" > > Thank you. Finally that configuration worked. However, the quality of the > call was very bad. I couldn't make out a word of what was coming through. > > > --------------------------------- > Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great > rates starting at 1¢/min. >

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[Users] Problem : Forwarding Routine to a NAT'd endpoint.
by Ricardo Martinez 30 May '06

30 May '06

Hello List I have a problem forwarding a number for a NAT'd endpoint. The problem presents as follow. For incoming calls to my OPENSER users i load via avp_load_radius("callee") the "forward options" for my users. For example if a user has set the Call Forward ON and the set the forwarded number, these options are loaded via avp_radius. So for a incoming call to a user with Call Forward ON, the call is then redirected to a Asterisk Box to handle the call. I have no problem with this if the user is not NAT'd, even if is NAT'd but the user is not registered i can forward the call to my Asterisk Box via a special route (route[8]) without any problem. But if the NAT'd endpoint is REGISTERED in my ser BOX like this : domain: 'location' aor : '5507515800' ~~~Contact(0x40502de0)~~~ domain : 'location' aor : '5507515800' Contact : 'sip:5507515800@192.168.1.100:5061' Expires : 560 q : Call-ID : '66e2643a-e646955e(a)192.168.1.100' CSeq : 2 User-Agent: 'Linksys/PAP2-3.1.3(LS)' received : 'sip:64.76.148.248:46139' State : CS_NEW Flags : 1 Sock : 64.76.148.246:5060 (0x8132c80) next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... I'm even using rewritehostport (as indicated in my ser.cfg snipet) route[8] { xlog("L_INFO", "[%Tf] : Ruta [8] Enviando llamada a asterisk\n"); rewritehostport("111.76.154.101:5060"); route(4); route(1); } the call is re-routed to the "received" parameter indicated in the register RECORD. Please check the next debug taken in my ser box. 111.76.148.237 : Caller Endpoint 111.76.148.246 : OPENSER Box 111.76.148.101 : Asterisk Box 111.76.148.248 : Called Enpoint (NAT'd) U 111.76.148.237:5060 -> 111.76.148.246:5060 INVITE sip:5507515800@sipvoiss.desa.redvoiss.net:5060 SIP/2.0. Via: SIP/2.0/UDP 111.76.148.237:5060;branch=z9hG4bKf50b1f9ca. Max-Forwards: 70. Content-Length: 304. To: sip:5507515800@sipvoiss.desa.redvoiss.net:5060. From: sip:2408100@111.76.148.237:5060;tag=acee04f44a858c7. Call-ID: 12524b29b2c08e262bc8daef67f6dcf0(a)111.76.148.237. CSeq: 1032746828 INVITE. Supported: timer. Content-Type: application/sdp. Contact: sip:2408100@111.76.148.237:5060. Supported: replaces. . v=0. o=MxSIP 0 26 IN IP4 111.76.148.237. s=SIP Call. c=IN IP4 111.76.148.237. t=0 0. m=audio 4978 RTP/AVP 18 4 4 101. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:4 G723/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:18 annexb=no. a=fmtp:101 0-15. a=sendrecv. m=image 4980 udptl t38. a=sendrecv. U 111.76.148.246:5060 -> 111.76.148.237:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 111.76.148.237:5060;branch=z9hG4bKf50b1f9ca. To: sip:5507515800@sipvoiss.desa.redvoiss.net:5060. From: sip:2408100@111.76.148.237:5060;tag=acee04f44a858c7. Call-ID: 12524b29b2c08e262bc8daef67f6dcf0(a)111.76.148.237. CSeq: 1032746828 INVITE. Server: Sip EXpress router (0.9.3 (i386/linux)). Content-Length: 0. Warning: 392 111.76.148.246:5060 "Noisy feedback tells: pid=22321 req_src_ip=111.76.148.237 req_src_port=5060 in_uri=sip:5507515800@sipvoiss.desa.redvoiss.net:5060 out_uri=sip:75755507515800@111.76.154.101:5060 via_cnt==1". . U 111.76.148.246:5060 -> 111.76.148.248:46139 INVITE sip:75755507515800@111.76.154.101:5060 SIP/2.0. Record-Route: <sip:111.76.148.246;ftag=acee04f44a858c7;lr=on>. Via: SIP/2.0/UDP 111.76.148.246;branch=z9hG4bK38bc.f157bcd5.0. Via: SIP/2.0/UDP 111.76.148.237:5060;branch=z9hG4bKf50b1f9ca. Max-Forwards: 16. Content-Length: 306. To: sip:5507515800@sipvoiss.desa.redvoiss.net:5060. From: sip:2408100@111.76.148.237:5060;tag=acee04f44a858c7. Call-ID: 12524b29b2c08e262bc8daef67f6dcf0(a)111.76.148.237. CSeq: 1032746828 INVITE. Supported: timer. Content-Type: application/sdp. Contact: sip:2408100@111.76.148.237:5060. Supported: replaces. . v=0. o=MxSIP 0 26 IN IP4 111.76.148.237. s=SIP Call. c=IN IP4 111.76.148.246. t=0 0. m=audio 35068 RTP/AVP 18 4 4 101. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:4 G723/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:18 annexb=no. a=fmtp:101 0-15. a=sendrecv. m=image 35070 udptl t38. a=sendrecv. Why openser is routing the call to the "received" ip address insted of the ip address indicated in my route[8] by the rewritehostport("111.76.154.101:5060"); ??? Hope that someone can help me here Is there a way to avoid this ? Thanks ! Ricardo Martinez Ricardo Javier Martinez Ogalde Ingeniero de Desarrollo VOISS NET S.A. Cisco Certified Network Associate ,CCNA (CSCO 10643101) * : (56 2) 240 81 96 * : (56 2) 245 74 95 * : rmartinez(a)redvoiss.net

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