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37456 discussions

Re: [Users] Radius
by Cseke Tamas 07 Jul '06

07 Jul '06

Hi, I ve added /etc/openser/dictionary.radius to /etc/radiusclient-ng/dictionary and to BSDradius dictionary file too Largo wrote: > Hi > > Try to use the file provided in the folder "/usr/local/etc/openser" > called "dictionary.radius" as a dictionary. > > Bye > > Cseke Tamas schrieb: > >> Hello, >> >> We 'd like to use openser with radius authentication, accounting >> >> We have installed openser from debian package on an ubuntu system: >> deb http://www.openser.org/debian sarge main >> >> But radius authentication failed, and we need avp_radius module too, >> wich is not included in this package i guess >> >> error message in the log: >> ERROR:auth_radius:radius_authorize_sterman: rc_auth failed >> >> on http://openser.org/docs/openser-radius-1.0.x.html i ve 'read that >> openser should be compiled (with some macros defined in makefile) to >> work with >> radius. >> Is there any possibility to install some deb packages instead of >> compilation? >> Which package should we install? >> >> Thanks, >> Tamas >> >> >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users > > >

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[Serusers] How to do REGEXP operation of mysql with db-api?
by wu yaqian 07 Jul '06

07 Jul '06

Hello,erery one, I want to use mysql's regexp operator in ser,but with db-api ,I can't do this.Because the patten is stored in the mysql table.The db-api can only do key op value.I want to do the value REGEXP key operation, Any one has advices to me? Thanks very much.

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Re: [Serusers] IP address spoofing
by Greger V. Teigre 07 Jul '06

07 Jul '06

>Hi Greger. SER and Asterisk can be configured to use TCP for SIP/SDP >messages? > Try this before forwarding the INVITE to Asterisk: if(!uri_param("transport")) { add_uri_param("transport=tcp"); } >The reason I said that the caller wouldn't receive audio is because the >callee's RTP stream would be directed to SER, not the caller. >-- Nick How come? I don't see rtpproxy or mediaproxy in your setup. So, unless you force an RTP proxy in your SER, I would expect Asterisk and caller to communicate directly on RTP. g-) Nick Hoffman wrote: >> Nick Hoffman wrote: >> >>> Hi guys. Say you have this setup, with an account for the caller on >>> both Asterisk and SER: >>> Caller -> SER -> Asterisk -> VoIP Provider -> Callee >>> >>> If the caller were to spoof SER's IP address and place a call directly >>> to Asterisk (thus circumventing SER), what would happen? >>> >>> If the call was in fact setup, obviously the caller would not receive >>> any audio from the callee. However, would the call be setup? When >>> Asterisk responds to the caller's request and sends SIP packets back >>> (to SER), would SER say "I don't know anything about this call! >>> Asterisk, kill this call please."? >>> >>> Thanks for your input! >>> -- Nick >>> e: nick.hoffman(a)altcall.com >>> p: +61 7 5591 3588 >>> f: +61 7 5591 6588 >>> > > > On Wed July 5 2006 17:58, "Greger V. Teigre" <greger(a)teigre.com> wrote: > >> Depends on the config and what type of message the caller managed to >> make asterisk create... You could (and probably should) put asterisk on >> a private routable network (i.e. NATed behind a firewall). The best >> would be to put ser and asterisk on the same network and only allow >> outside world to contact ser and let ser contact asterisk using the >> private address of asterisk. Alternatively you could use tcp to >> asterisk and stop udp traffic. >> Why the caller wouldn't receive audio, I don't understand... >> g-) >> > > > Hi Greger. SER and Asterisk can be configured to use TCP for SIP/SDP > messages? > > The reason I said that the caller wouldn't receive audio is because the > callee's RTP stream would be directed to SER, not the caller. > -- Nick > e: nick.hoffman(a)altcall.com > p: +61 7 5591 3588 > f: +61 7 5591 6588 > > If you receive this email by mistake, please notify us and do not make any > use of the email. We do not waive any privilege, confidentiality or > copyright associated with it. > > >

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[Users] rtpproxy - fax
by Hakan YASTI 07 Jul '06

07 Jul '06

Dear All, I use openser 1.0.1 and use rtp proxy for some cases other than nat option. At voice calls I do not have any problem. But When I try to send fax,it is unsuccesful. I investigated the logs and saw that, after the reinvites ( to take the session from g729 to G711u bypass mode ), the last OK message's SDP that is sending from openser to user agent is not correct.Etheral logs are attached. Is it a bug of rtpproxy ? Any suggestion. Best Regards, Hakan.

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[Users] NEED HELP
by Tanzeel serfaraz 07 Jul '06

07 Jul '06

Hi users, i have the task to implement "IP authentication for OPENSER using the database"IP are trusted IP's, I have the idea that it would be done through AVPOPS module but for this i need your help . if any one implment AVPOPS module please send me OPENSER.cnf file (if possible) so that i will take idea about implementing the module thanks a lot for your help, TANZEEL --------------------------------- Want to be your own boss? Learn how on Yahoo! Small Business.

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[Users] ERROR: extract_body: message body has length zero
by Kenny Chua 07 Jul '06

07 Jul '06

I'm using openser 1.0.1 with rtpproxy and nathelper. Calls are goin through with good audio. Hanging up is fine too. However I've noticed I've been getting errors in my logs: ERROR: extract_body: message body has length zero ERROR: force_rtp_proxy2: can't extract body from the message I'm assuming I'm doin something wrong in my config but I dont know where. Can someone please point out my mistakes? Thank you # start RTPProxy with: rtpproxy -l your_public_ip -s udp:localhost:8899 # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ port=5060 children=4 listen=68.xxx.xxx.xxx alias=test.com check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" # --- module loading 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/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" # --- setting module parameters modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") # -- usrloc params -- modparam("usrloc", "db_mode", 2) # -- registrar params -- modparam("registrar", "nat_flag", 6) modparam("registrar", "sip_natping_flag", 7) # -- auth params -- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") # -- rr params -- modparam("rr", "enable_full_lr", 1) # -- nathelper -- modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_from", "sip:pinger@test.com") # -- tm params -- modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") # -- permissions param -- modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") # --- main routing logic route{ # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- 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; }; # ----------------------------------------------------------------- # NAT detection # ----------------------------------------------------------------- route(2); # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (!method=="REGISTER") record_route(); # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); return; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); }; route(1); }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbidden"); }; }; if (method=="ACK") { route(1); return; } else if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { route(5); return; } else if (method=="REGISTER") { route(4); return; }; lookup("aliases"); if (!is_uri_host_local()) { route(4); route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { if (subst_uri('/(sip:.*);nat=yes/\1/')){ setflag(6); }; if (isflagset(5)||isflagset(6)) { route(3); } if (!t_relay()) { sl_reply_error(); }; exit; } route[2]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; } route[3] { if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); t_on_failure("1"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); } route[4] { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; }; if (isflagset(5)) { setflag(6); # if you want OPTIONS natpings uncomment next # setflag(7); }; save("location"); if (!lookup("location")) { sl_reply_error(); }; append_hf("P-hint: usrloc applied\r\n"); } route[5] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; }; lookup("aliases"); if (!is_uri_host_local()) { route(1); return; }; if (uri=~"^sip:[0-9]{3}@") { lookup("aliases"); lookup("location"); route(1); return; }; if(!lookup("location")){ if (uri=~"^sip:[0-9]*@") { route(6); return; }; sl_send_reply("404", "User Not Found"); return; }; route(1); } route[6] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehostport("pstn.gateway:5060"); avp_write("i:45", "inv_timeout"); route(1); } failure_route[1] { if (isflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } } onreply_route[1] { if ((isflagset(5) || isflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); if (isflagset(6)) { fix_nated_contact(); } exit; } --------------------------------- Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail Beta.

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[Users] P-D installation problem
by testcis 07 Jul '06

07 Jul '06

Hello! So glad to join your community. We've desided to use our server like a sipproxy server, but I have a priblem during installation of OpenSER. It looks like there are some troubles with Makefile. I have Pentium-D 3200 Suse server and whenever I try to compile OpenSER I have the following message: linksysserver:/openser-1.0.1 # make all Compiling action.c gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=x86_64 -DNAME='"openser"' -DVERSION='"1.0.1"' -DARCH='"x86_64"' -DOS='"linux"' -DCOMPILER='"gcc 4.1.0"' -D__CPU_x86_64 -D__OS_linux -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c action.c -o action.o action.c:1: error: bad value (x86_64) for -mtune= switch make: *** [action.o] Error 1 Do you have any ideas on how to solve this problem?

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[Users] need help progressing from newb status.
by Marc Rys 07 Jul '06

07 Jul '06

Hello all. Over the last week, I’ve installed OpenSER on my Debian Sarge box with some degree of success. I’ve succeeded adding users to OpenSER, and I’ve stumbled my way through the openserctl commands to add my VOIP capable Ascend Max as a digital PSTN gateway. At this point I can only get PSTN calls to terminate from the PSTN to my un-NAT’d VOIP UA’s(grandstream HT-486’s). I’m still on my search to find the proper openserctl command to route all PSTN calls out my gateway. (any help here would be appreciated) Also, could somebody point me in the right direction on where to find complete documentation on how to setup mediaproxy on OpenSER? I’ve followed the install instructions included with mediaproxy-1.7.2.tar.gz. but when I I attempt to start mediaproxy I get the following error…. sipproxy1:~# /etc/init.d/mediaproxy start Starting SER MediaProxy server: mediaproxy/usr/bin/env: python: No such file or directory /usr/bin/env: python: No such file or directory My main goal is to get these HT486’s further out on the Internet and of course behind some kind of NAT device. Thanks in advance for your help. Marc -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.8/381 - Release Date: 7/3/2006

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[Users] RTP , NAT problem,
by raviprakash sunkara 07 Jul '06

07 Jul '06

Hi Users, I install the openser 1.0.1 and rtp proxy 0.3 in same Linux System openser server is located with public id xx.xxx.xxx.xx of 192.168.2.2 , And UAC are outside the NAT, When one UAC call to other UAC( are both in outside the NAT where openser server), after the INVITE method get request by server, after 32 second its hung up automatically, Voice is ok , and callee is hung upping, not caller, UAC ( inside the nAT , openser server ) in not hung uping and voice is not ok.... Where is the problem, in NAt with rtp or networking, route{ 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; }; # NAT detection route(2); if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("xx.xxx.xxx.xxx", "subscriber")) { www_challenge("xx.xxx.xxx.xxx", "0"); exit; }; if (isflagset(5)) { setflag(6); # if you want OPTIONS natpings uncomment next # setflag(7); }; save("location"); exit; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { if (subst_uri('/(sip:.*);nat=yes/\1/')){ setflag(6); }; if (isflagset(5)||isflagset(6)) { route(3); } if (!t_relay()) { sl_reply_error(); }; exit; } route[2]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; } route[3] { if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); t_on_failure("1"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); } failure_route[1] { if (isflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } } onreply_route[1] { if ((isflagset(5) || isflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); if (isflagset(6)) { fix_nated_contact(); } exit; } -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology Kondapur, Hi-tech city, Hyderabad. www.hyperion-tech.com +91-9985077535

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[Users] Sip infrastructure questions
by dam＠cosinux.org 07 Jul '06

07 Jul '06

Hello, First, thanks to the developer of the openser software for their work. I will appreciate it when my configuration will be working ;) I would like to configure sip on my server but I'm a bit lost. I've installed openser on it but I cannot have it to do what I want to do. So this is what I want to do: - I own a virtual server which host my domain cosinux.org . I would like to create myself a sip address dam(a)cosinux.org. - I would like to be able to send/receive sip call at that address from work. My work network is firewalled, however, I have a vtun tunnel with my server. - I would like to be able to send/receive sip call at that address from home. I have dsl at home and I can configure nat as I want to (port forwarding for example). >From what I understand, openser is used as registrar. It means my sip client connects to it, tells him that dam(a)cosinux.org is online and openser then redirects call to my client. This will work from my home network if I configure nat correctly. For my work network, I need a sip proxy so that each packet is routed by my server to my client, because others sip user agent cannot see my IP address. If I am right, what I have to do is - install openser on my server and tell him to handle cosinux.org sip address. - install mediaproxy to relay the sip connection from my work. So, does this seem sensible ? Am I doing something wrong or missing something ? Please CC me as I'm not suscribed to the list. Thank you very much, dam -- Damien MERENNE <dam(a)cosinux.org> http://www.cosinux.org/blogs/dam/ CNN was originally created as the "Chuck Norris Network" to update Americans with on-the-spot ass kicking in real-time.

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