- sr-users - kamailio.org

[Serusers] SER Children Misbehaving
by Andres 26 Nov '05

26 Nov '05

Today we had an incident where SER (0.9.4) children drained all the CPUs of one of our servers. Top Showed: PID USER PRI NI SIZE RSS SHARE STAT %CPU %MEM TIME CPU COMMAND 17925 root 25 0 5644 5644 3888 R 25.5 0.2 6:26 1 ser 17929 root 25 0 5672 5672 3880 R 24.7 0.2 6:48 0 ser 17928 root 25 0 5688 5688 3872 R 24.3 0.2 6:25 1 ser 17933 root 25 0 4540 4540 3740 R 22.8 0.2 6:00 0 ser And .. # ps -Al | grep ser 1 S 0 17901 1 0 85 0 - 14200 pause ? 00:00:00 ser 1 S 0 17916 17901 0 75 0 - 14200 pipe_w ? 00:00:00 ser 1 S 0 17917 17901 0 75 0 - 14418 schedu ? 00:00:22 ser 1 S 0 17918 17901 0 75 0 - 14422 schedu ? 00:00:23 ser 1 S 0 17919 17901 0 75 0 - 14423 schedu ? 00:00:24 ser 1 S 0 17920 17901 0 75 0 - 14447 schedu ? 00:00:22 ser 1 S 0 17921 17901 0 75 0 - 14421 schedu ? 00:00:22 ser 1 S 0 17922 17901 0 75 0 - 14424 schedu ? 00:00:22 ser 1 S 0 17923 17901 0 75 0 - 14428 schedu ? 00:00:21 ser 1 S 0 17924 17901 0 75 0 - 14424 schedu ? 00:00:22 ser 1 R 0 17925 17901 0 85 0 - 14448 - ? 00:06:22 ser 1 S 0 17926 17901 0 75 0 - 14457 schedu ? 00:00:49 ser 1 S 0 17927 17901 0 75 0 - 14453 schedu ? 00:00:50 ser 1 R 0 17928 17901 0 85 0 - 14477 - ? 00:06:20 ser 1 R 0 17929 17901 0 85 0 - 14455 - ? 00:06:44 ser 1 S 0 17930 17901 0 75 0 - 14452 schedu ? 00:00:50 ser 1 S 0 17931 17901 0 75 0 - 14448 schedu ? 00:00:50 ser 1 S 0 17932 17901 0 76 0 - 14448 schedu ? 00:00:49 ser 1 R 0 17933 17901 0 85 0 - 14235 - ? 00:05:55 ser As you can see it looks like 4 children dropped out of the scheduler. The only thing suspicious is that RTPProxy became non-responsive around that time. At least thats the only thing the log shows: Nov 22 15:56:17 /usr/local/sbin/ser[17931]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Any idea why these 4 children dropped out? Any hints on how to troubleshoot this? Thanks, -- Andres Network Admin http://www.telesip.net

2 5

RE: [Users] Setting t_relay() timeout in lcr
by Douglas Garstang 26 Nov '05

26 Nov '05

Yes. I have read sections 1.3.17 and 1.3.18 of the module documentation for lcr. That's why I made the post. Section 1.3.17 says: 'Sets the value of the fist INVITE's Final Response timeout to be used during sequential forwarding:" Section 1.3.18 says: "Sets the value of the next INVITE's Final Response timeouts to be used during sequential fowarding:" So, I went and set: modparam("lcr","fr_inv_timer",2) modparam("lcr","fr_inv_timer_next",2) and it had no effect on the timeout that t_relay() used. However, when I set: modparam("tm", "fr_timer", 1) it _DID_ have an effect in the timeout that t_relay() used. I _have_ experimed with the variables and functions WAY more than I would have if the documentation was complete. Experimenting with the functions showed me that t_relay() is always returning 0, even when there is a failure. My question therefore stands. Why does setting fr_inv_timer and fr_inv_timer_next have no effect? What is the relationship between the settings in the tm module and the lcr module? Why are there two settings? Which should I use? Why does t_relay() always return 0 even on failure? I could be doing things completely wrong, as there is a lack of good examples accompanying the documentation, and it's easy to see the documentation was written by the same people who wrote the code - there's an assumed level of knowledge when reading it that most people don't have. Douglas. -----Original Message----- From: Juha Heinanen [mailto:jh@tutpro.com] Sent: Fri 11/25/2005 6:32 PM To: Douglas Garstang Cc: users(a)openser.org Subject: RE: [Users] Setting t_relay() timeout in lcr Douglas Garstang writes: > I just managed to set the timeout with fr_inv_timer from the tm > module, and it seems to have the right effect. So, what are > fr_inv_timer and fr_inv_timer_next in lcr module for then??? before you sent this message, have you checked sections 1.3.17 and 1.3.18 of lcr README file? you should read all existing documentation, experiment with the variables and the functions and if you after that still have problems, you post your message to the list. -- juha

1 0

RE: [Users] Setting t_relay() timeout in lcr
by Douglas Garstang 26 Nov '05

26 Nov '05

I just managed to set the timeout with fr_inv_timer from the tm module, and it seems to have the right effect. So, what are fr_inv_timer and fr_inv_timer_next in lcr module for then??? -----Original Message----- From: Douglas Garstang Sent: Fri 11/25/2005 5:07 PM To: users(a)openser.org Cc: Subject: [Users] Setting t_relay() timeout in lcr How can I set the t_relay() INVITE timeout in the lcr module? Setting fr_inv_timer and fr_inv_timer_next with modparam seems to have no effect. What's the significance of fr_inv_timer_avp? Do I need to set that instead? And just what is an 'rpid'??? I can't find this documented anywhere. Thanks

2 1

RE: [Users] Setting t_relay() timeout in lcr
by Douglas Garstang 26 Nov '05

26 Nov '05

When my t_relay() times out, it returns 0. Why doesn't it return -1? Before calling t_relay(), I call t_on_failure(). If t_relay() always returns 0, why does my failure block execute? This makes no sense to me. I must not be reading in between the lines of the docs enough, because theres a lot of things here that just don't add up. Yes, I'm asking a lot of questions. That's because the docs don't have all the information. Doug. -----Original Message----- From: Douglas Garstang Sent: Fri 11/25/2005 5:16 PM To: users(a)openser.org Cc: Subject: RE: [Users] Setting t_relay() timeout in lcr I just managed to set the timeout with fr_inv_timer from the tm module, and it seems to have the right effect. So, what are fr_inv_timer and fr_inv_timer_next in lcr module for then??? -----Original Message----- From: Douglas Garstang Sent: Fri 11/25/2005 5:07 PM To: users(a)openser.org Cc: Subject: [Users] Setting t_relay() timeout in lcr How can I set the t_relay() INVITE timeout in the lcr module? Setting fr_inv_timer and fr_inv_timer_next with modparam seems to have no effect. What's the significance of fr_inv_timer_avp? Do I need to set that instead? And just what is an 'rpid'??? I can't find this documented anywhere. Thanks

1 0

[Users] Setting t_relay() timeout in lcr
by Douglas Garstang 26 Nov '05

26 Nov '05

How can I set the t_relay() INVITE timeout in the lcr module? Setting fr_inv_timer and fr_inv_timer_next with modparam seems to have no effect. What's the significance of fr_inv_timer_avp? Do I need to set that instead? And just what is an 'rpid'??? I can't find this documented anywhere. Thanks

1 0

[Serusers] NAT again
by bosinclar＠laposte.net 25 Nov '05

25 Nov '05

I need to add some functions in my ser.cfg for nat support. I use my open(ser) as outbound proxy with two interfaces ppp0 and eth0. sip agents send requests to eth0. I want my packets passing through the register and invite blocks by fixing contact with a public ip. All packets from private network must be sent to eth0. Any suggestions Bob Accédez au courrier électronique de La Poste : www.laposte.net ; 3615 LAPOSTENET (0,34/mn) ; tél : 08 92 68 13 50 (0,34/mn)

3 2

[Users] writing URI lists from headers into AVPs
by JF 25 Nov '05

25 Nov '05

Hi, Is there a way (from the script) to write a URI list from a SIP header (like the ones in Route or Record-Route headers) into one or more AVPs? Thanks in advance! JF

2 1

RE: [Users] Multiple IP in one box
by Fernando Rodriguez 25 Nov '05

25 Nov '05

Iqbal, Actualy the conection between the atas is made the problem is that only one side can hear the other so the voice is not going tru. Of I run ser/openser with out forking I see it listening to all the ip addresses, I think is more an issue of using media proxy or rtpproxy. The 10.xx and 192.169.xx networks can ping and do anything to the internet the problem is from the internet to the inside .. -----Mensaje original----- De: Iqbal [mailto:iqbal@gigo.co.uk] Enviado el: Martes, 22 de Noviembre de 2005 10:38 a.m. Para: Fernando Rodriguez Asunto: Re: [Users] Multiple IP in one box Hi You have 3 IP to which you want ser to listen to, to allow ATA to make calls, why not just tell the ATA the sip proxy addess, instead and let it deal with the NAT issue and henceonly listen to one IP address, I mean can your ATA on the 10.254 network see the public IP address vis NA translation. If not get SER to listen to all the IP address, by using the listen command iqbal Fernando Rodriguez wrote: > Im trying to use openser as a Proxy for different interfaces in one > box I have tried almos averithing to make it work properly but for > some reason I cant. > > > > Here is my setup and I hope someone can helpme or guide me in the > right way. > > > > > > > > > > One Linux-OpenSer box with 4 netcards > > > > 1 Has a public ip > > 2 Has a 192.168.23.10 ip > > 3 Has a 10.254.10.240 ip > > 4 Is not being used > > > > I want to be able to make calls from the ATAs in all the different > networks, but the fact is that as it is now I cant, I want to know if > I have to set up multiple media proxies or rtpproxy if I have to use a > proxy dispatcher or not. > > > > This is my configuration file > > > > > > > > > > # ----------- global configuration parameters ------------------------ > > > > #debug=9 # 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 > > */ > > > > 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" > > fifo_db_url="mysql://openser:openserrw@localhost/openser" > > > > alias="test.net" > > # ------------------ module loading ---------------------------------- > > > > #loadmodule "/usr/lib/openser/modules/acc.so" > > loadmodule "/usr/lib/openser/modules/mysql.so" > > loadmodule "/usr/lib/openser/modules/sl.so" > > loadmodule "/usr/lib/openser/modules/tm.so" > > loadmodule "/usr/lib/openser/modules/rr.so" > > loadmodule "/usr/lib/openser/modules/maxfwd.so" > > loadmodule "/usr/lib/openser/modules/usrloc.so" > > loadmodule "/usr/lib/openser/modules/registrar.so" > > loadmodule "/usr/lib/openser/modules/auth.so" > > loadmodule "/usr/lib/openser/modules/auth_db.so" > > loadmodule "/usr/lib/openser/modules/uri.so" > > loadmodule "/usr/lib/openser/modules/uri_db.so" > > loadmodule "/usr/lib/openser/modules/domain.so" > > loadmodule "/usr/lib/openser/modules/mediaproxy.so" > > loadmodule "/usr/lib/openser/modules/nathelper.so" > > loadmodule "/usr/lib/openser/modules/textops.so" > > > > > > # ----------------- setting module-specific parameters --------------- > > # -- usrloc params -- > > > > modparam("usrloc", "db_mode", 2) > > modparam("usrloc", "timer_interval", 10) > > modparam("usrloc", "db_url", > "mysql://openser:openserrw@localhost/openser") > > > > # -- auth params -- > > modparam("auth_db", "db_url", > "mysql://openser:openserrw@localhost/openser") > > modparam("auth_db", "calculate_ha1", yes) > > 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) > > > > > > # -- tm Params -- > > modparam("tm", "fr_inv_timer", 45 ) > > modparam("tm", "fr_timer", 12 ) > > > > modparam("registrar", "nat_flag", 6) > > > > > > modparam("nathelper", "rtpproxy_disable", 1) > > #modparam("nathelper", "natping_interval", 30) > > #modparam("nathelper", "sipping_from", "sip:ping@sipserver.net") > > > > > > #modparam("mediaproxy", "ping_nated_only", 0) # Ping only clients > behind NAT > > modparam("mediaproxy", "natping_interval", 20) > > modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") > > > > modparam("registrar", "nat_flag", 6) > > > > > > # ------------------------- request routing logic ------------------- > > > > # main routing logic > > > > route { > > > > # > ----------------------------------------------------------------- > > # Procesos de Limpieza > > # > ----------------------------------------------------------------- > > if (!mf_process_maxfwd_header("10")) { > > sl_send_reply("483", "Too Many Hops"); > > break; > > }; > > > > if (msg:len > max_len) { > > sl_send_reply("513", "Message Overflow"); > > break; > > }; > > > > # > ----------------------------------------------------------------- > > # Registro > > # > ----------------------------------------------------------------- > > if (method!="REGISTER") { > > log(1, "Area de Registro\n"); > > > > record_route(); > > }; > > > > > > # > ----------------------------------------------------------------- > > # Call Tear Down Section > > # > ----------------------------------------------------------------- > > if (method=="BYE" || method=="CANCEL") { > > log(1, "Proceso de fin de llamada\n"); > > end_media_session(); > > }; > > # > ----------------------------------------------------------------- > > # Loose Route Section > > # > ----------------------------------------------------------------- > > if (loose_route()) { > > > > if (has_totag() && (method=="INVITE" || method=="ACK")) { > > > > if (client_nat_test("3") || > search("^Route:.*;nat=yes")) { > > setflag(6); > > use_media_proxy(); > > }; > > }; > > > > route(1); > > break; > > }; > > > > # > ----------------------------------------------------------------- > > # Call Type Processing Section > > # > ----------------------------------------------------------------- > > > > if (uri!=myself) { > > route(1); > > break; > > }; > > > > if (uri==myself) { > > > > if (method=="CANCEL") { > > route(3); > > break; > > } else if (method=="INVITE") { > > route(3); > > break; > > } else if (method=="REGISTER") { > > route(2); > > break; > > }; > > > > lookup("aliases"); > > if (uri!=myself) { > > route(1); > > break; > > }; > > > > if (!lookup("location")) { > > sl_send_reply("404", "User Not Found"); > > break; > > }; > > }; > > > > route(1); > > } > > > > route[1] { > > > > # > ----------------------------------------------------------------- > > # Default Message Handler > > # > ----------------------------------------------------------------- > > > > t_on_reply("1"); > > > > if (!t_relay()) { > > > > if (method=="INVITE" || method=="ACK") { > > end_media_session(); > > }; > > > > sl_reply_error(); > > }; > > } > > > > route[2] { > > > > # > ----------------------------------------------------------------- > > # REGISTER Message Handler > > # ---------------------------------------------------------------- > > > > sl_send_reply("100", "Trying"); > > > > # if(dst_ip==192.168.1.1) { > > # log("message received on loopback interface\n"); > > # setflag(7); > > # fix_nated_register(); > > # force_rport(); > > # }; > > # if(dst_ip==192.168.108.253) { > > # log("message received on loopback interface\n"); > > # setflag(6); > > # fix_nated_register(); > > # force_rport(); > > # }; > > > > if (!www_authorize("test.net","subscriber")) { > > www_challenge("test.net","0"); > > break; > > }; > > > > if (!check_to()) { > > sl_send_reply("401", "Unauthorized"); > > break; > > }; > > > > consume_credentials(); > > > > if (!save("location")) { > > sl_reply_error(); > > }; > > fix_nated_register(); > > } > > > > route[3] { > > > > # > ----------------------------------------------------------------- > > # CANCEL and INVITE Message Handler > > # > ----------------------------------------------------------------- > > > > lookup("aliases"); > > if (uri!=myself) { > > route(1); > > break; > > }; > > > > if (!lookup("location")) { > > sl_send_reply("404", "User Not Found"); > > break; > > }; > > > > if (method=="CANCEL") { > > route(1); > > break; > > }; > > > > if (!proxy_authorize("","subscriber")) { > > proxy_challenge("","0"); > > break; > > } else if (!check_from()) { > > sl_send_reply("403", "Use From=ID"); > > break; > > }; > > > > consume_credentials(); > > > > # > ------------------------------------------------------------------------ > > # Reglas de LLamada > > # > ------------------------------------------------------------------------ > > > > if(uri=~"(^sip:1[0-9]*@.*)" && > from_uri=~"(^sip:2[0-9]*@.*)") { > > log(1, "Llamada de extenciones 1xxx a 2xxx\n"); > > use_media_proxy(); > > }; > > > > if(uri=~"(^sip:1[0-9]*@.*)" && > from_uri=~"(^sip:3[0-9]*@.*)") { > > log(1, "Llamada de extenciones 1xxx a 3xxx\n"); > > fix_contact(); > > use_media_proxy(); > > }; > > > > if(uri=~"(^sip:1[0-9]*@.*)" && > from_uri=~"(^sip:4[0-9]*@.*)") { > > log(1, "Llamada de extenciones 1xxx a 4xxx\n"); > > fix_contact(); > > use_media_proxy(); > > }; > > > > if(uri=~"(^sip:2[0-9]*@.*)" && > from_uri=~"(^sip:1[0-9]*@.*)") { > > log(1, "Llamada de extenciones 2xxx a 1xxx\n"); > > fix_contact(); > > use_media_proxy(); > > }; > > > > if(uri=~"(^sip:2[0-9]*@.*)" && > from_uri=~"(^sip:3[0-9]*@.*)") { > > log(1, "Llamada de extenciones 2xxx a 3xxx\n"); > > fix_contact(); > > use_media_proxy(); > > }; > > > > if(uri=~"(^sip:2[0-9]*@.*)" && > from_uri=~"(^sip:4[0-9]*@.*)") { > > log(1, "Llamada de extenciones 2xxx a 4xxx\n"); > > fix_contact(); > > use_media_proxy(); > > }; > > > > if(uri=~"(^sip:3[0-9]*@.*)" && > from_uri=~"(^sip:1[0-9]*@.*)") { > > log(1, "Llamada de extenciones 3xxx a 1xxx\n"); > > fix_contact(); > > use_media_proxy(); > > }; > > > > if(uri=~"(^sip:3[0-9]*@.*)" && > from_uri=~"(^sip:2[0-9]*@.*)") { > > log(1, "Llamada de extenciones 3xxx a 2xxx\n"); > > fix_contact(); > > use_media_proxy(); > > }; > > > > if(uri=~"(^sip:3[0-9]*@.*)" && > from_uri=~"(^sip:4[0-9]*@.*)") { > > log(1, "Llamada de extenciones 3xxx a 4xxx\n"); > > fix_contact(); > > use_media_proxy(); > > }; > > > > if(uri=~"(^sip:4[0-9]*@.*)" && > from_uri=~"(^sip:1[0-9]*@.*)") { > > log(1, "Llamada de extenciones 4xxx a 1xxx\n"); > > fix_contact(); > > use_media_proxy(); > > }; > > > > if(uri=~"(^sip:4[0-9]*@.*)" && > from_uri=~"(^sip:2[0-9]*@.*)") { > > log(1, "Llamada de extenciones 4xxx a 2xxx\n"); > > fix_contact(); > > use_media_proxy(); > > }; > > > > if(uri=~"(^sip:4[0-9]*@.*)" && > from_uri=~"(^sip:3[0-9]*@.*)") { > > log(1, "Llamada de extenciones 4xxx a 3xxx\n"); > > fix_contact(); > > use_media_proxy(); > > }; > > > > route(1); > > } > > > > onreply_route[1] { > > > > #if ((isflagset(6) || isflagset(7)) && > (status=~"(180)|(183)|2[0-9][0-9]")) { > > > > # if (!search("^Content-Length:\ +0")) { > > log(1, "Mensaje de prueba en reply route\n"); > > fix_contact(); > > use_media_proxy(); > > # }; > > # }; > > > > } > > > > > > As you can see I use different starting numbers for the ATA for each > of the interfaces, I tested using only one mediaproxy listening to > 127.0.0.1 but I have tried a lot of configurations, all the clients > register perfectly but cant seem to call one another exept the ones > that are on the same interface. > > > > Im using fedora core 3 with openser 0.95 mediaproxy 1.42 > > > > Hope any one can help me or point me the way to do this of if its not > viable or cant be done. > > > > Thanks a lot for your kind help > > > > Yours truly > > > > > > Fernando Rodriguez V. > > frod(a)aitelecom.net <mailto:frod@aitelecom.net> > > AITelecom S.A. de C.V. > > http://www.aitelecom.net > > > > > > > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > >

3 3

[Serusers] Noisy feedback tells & radius_authorize_sterman(): Failure
by Kamran Ahmad 25 Nov '05

25 Nov '05

Hi all I am using SER-0.9.3 with radiusclient-ng-0.5.1 but unable to get thrrough radius Digest Authentication i am getting Noisy feedback tells and no request to Radius ---------------------------------------------------- 192.168.0.1:10411 -> 192.168.0.3:5060 REGISTER sip:192.168.0.3 SIP/2.0..To: 2000<sip:2000@192.168.0.3>..From: 2000<sip:2000@212. 36.71.78>;tag=68a232bb..Via: SIP/2.0/UDP 192.168.1.236:10411;branch=z9hG4bK-d87543-788469354 -1--d87543-;rport..Call-ID: 726b328158a24912..CSeq: 2 REGISTER..Contact: <sip:2000@192.168.1 .236:10411>..Expires: 60..Max-Forwards: 70..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: eyeBeam release 3005x stamp 17288..Authorizat ion: Digest username="2000",realm="192.168.0.3",nonce="438142b95bb202f9f7b35267264fa3cdf240 c34c",uri="sip:192.168.0.3",response="587553ab1fdd8979c16dd5619ea1f3f1",algorithm=MD5..Cont ent-Length: 0.... # U 192.168.0.3:5060 -> 192.168.0.1:10411 SIP/2.0 100 Trying..To: 2000<sip:2000@192.168.0.3>..From: 2000<sip:2000@192.168.0.3>;tag=6 8a232bb..Via: SIP/2.0/UDP 192.168.1.236:10411;branch=z9hG4bK-d87543-788469354-1--d87543-;rpo rt=10411;received=192.168.0.1..Call-ID: 726b328158a24912..CSeq: 2 REGISTER..Server: OpenSer (0.9.5 (i386/linux))..Content-Length: 0..Warning: 392 192.168.0.3:5060 "Noisy feedback tel ls: pid=27853 req_src_ip=192.168.0.1 req_src_port=10411 in_uri=sip:192.168.0.3 out_uri=si p:192.168.0.3 via_cnt==1".... # U 192.168.0.3:5060 -> 192.168.0.1:10411 SIP/2.0 401 Unauthorized..To: 2000<sip:2000@192.168.0.3>;tag=236419814c1d6cbca250c821be3316 c3.97c8..From: 2000<sip:2000@192.168.0.3>;tag=68a232bb..Via: SIP/2.0/UDP 192.168.1.236:1041 1;branch=z9hG4bK-d87543-788469354-1--d87543-;rport=10411;received=192.168.0.1..Call-ID: 726 b328158a24912..CSeq: 2 REGISTER..WWW-Authenticate: Digest realm="192.168.0.3", nonce="7264fa3cdf240c34c"..Server: OpenSer (0.9.5 (i386/linux))..Content-Length: 0..Warning: 392 192.168.0.3:5060 "Noisy feedback tells: pid=27853 req_src_ip=192.168.0.1 req_src_port=10411 in_uri=sip:192.168.0.3 out_uri=sip:192.168.0.3 via_cnt==1".... ------------------------------------------------------------------------ why i am getting these ------------------------------------------------------- 0(27853) radius_authorize_sterman(): Failure 0(27853) build_auth_hf(): 'WWW-Authenticate: Digest realm="192.168.0.3, nonce="43814271015d7812e9f068467d0c352ceb679d70" ' 0(27853) parse_headers: flags=-1 0(27853) check_via_address(192.168.0.1, 192.168.0.1, 0) 0(27853) DEBUG:destroy_avp_list: destroying list 0x4043dee0 0(27853) receive_msg: cleaning up ser.cfg ----------------------- loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/uri.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" modparam("usrloc", "db_mode", 0) modparam("auth_radius","radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("auth_radius","service_type",15) # 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) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") #modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) ############################################################################# modparam("tm", "fr_timer", 40) modparam("tm", "fr_inv_timer", 35) modparam("tm", "wt_timer", 5) modparam("tm", "fr_inv_timer_avp", "inv_timeout") ########################################################################### # ------------------------- request routing logic ------------------- # main routing logic route{ # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; if (!method=="REGISTER") { record_route(); } if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } 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"); force_rtp_proxy(); }; route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command if (uri==myself) { if (method=="INVITE") { route(3); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:\ +\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!radius_www_authorize("")) { www_challenge("","0"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

1 0

RE: [Users] Dispatcher module - Does it actually work?
by Douglas Garstang 25 Nov '05

25 Nov '05

So, will the lcr module let me timeout a gateway, and try another if it's unavailable, as someone else suggested? As soon as I call t_relay(), I hit the same problem.... it never times out and tries to reach the first gateway forever. -----Original Message----- From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] Sent: Fri 11/25/2005 2:56 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Dispatcher module - Does it actually work? On 11/24/05 22:42, Douglas Garstang wrote: > Hashing over call-id may work to some degree, and I'll investigate it more, but hashing over the From: means that the same caller will always get the same Asterisk box, and your redundancy goes down the toilet. > The redundancy you talk about is only in your mind. I never said that dispatcher supports redundancy. Please read carefully the messages and have an adequate language. Besides that, the topic was not about the redundancy, but about traffic dispatching. To get the right answer put the proper question. As you said, "The request URI from the phones is always going to be the same." I suggested to use either call-id or from uri hash algorithm. Yes, for same From uri the destination will be the same, and also for same call-id the destination will be the same. Moreover, for different values, the destination can be the same, since there is hash computation and some results will overlap. Cheers, Daniel > > Actually I guess it's down the toilet anyway. The bottom line seems to be that it doesn't matter whether you use the dispatcher or SRV records to distribute load, OpenSER will never try to connect to anything else on failure and at least for us, that makes it use in a production system, not an option. > > > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Sent: Thu 11/24/2005 12:19 PM > To: Douglas Garstang > Cc: users(a)openser.org > Subject: Re: [Users] Dispatcher module - Does it actually work? > > > > On 11/24/05 19:53, Douglas Garstang wrote: > > The request URI from the phones is always going to be the same. Doesn't that mean that OpenSER will always just use the same Asterisk box then? > if you do hash over r-uri, then yes, all messages will go to same > destination. > > I'm trying to round-robin as evenly (as possible) and distribute call load amonst multiple Asterisk systems. > > > In this case hash over call-id or from is more indicated. > > Cheers, > Daniel > > > -----Original Message----- > > From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > > Sent: Thu 11/24/2005 10:25 AM > > To: Douglas Garstang > > Cc: users(a)openser.org > > Subject: Re: [Users] Dispatcher module - Does it actually work? > > > > > > > > On 11/24/05 01:37, Douglas Garstang wrote: > > > After many many fruitless hours spent on trying to get OpenSER to work with DNS SRV records (I made a post to the group earlier today on this), I stumbled across the dispatcher module, and have been trying to get it to work instead. > > > > > > The module was designed to load balance, right? It doesn't seem to have been designed very well, unless I am missing something. > > > > > If you read carefully the documentation of the module, you will see the > > scope and limitations of the module: > > http://openser.org/docs/modules/1.0.x/dispatcher.html > > > > If something is not according to the documentation, then it is a bug. > > Stateless means that the module does not keep any state, so it cannot > > select the same destination using algorithm 4 (round robin) after a > > authentication request, choose another algorithm (hashing r-uri, > > callid). Also, being stateless, it cannot try the next address if the > > selected destination fails. > > > > The module was not designed to be load balancer, but a request > > dispatcher. Load balancing is something more complex. > > > > In the last weeks I added destination failover support in dispatcher > > (serial forking within the destination set, with auto-detection of > > failed destinations). It will be on cvs very soon, I have to update the > > documentation and do some testing. > > > > Cheers, > > Daniel > > > > > > > > If you set the algorithm with ds_select_dst() to 4, round robin, it doesn't work! Here's what happens. > > > > > > 1. UA sends INVITE to OpenSER > > > 2. OpenSER forwards INVITE to the first proxy in the dispatcher list. > > > 3. Proxy sends back a digest challenge for authorisation credentials to OpenSER. > > > 4. OpenSER forwards the challenge back to the UA. > > > 5. The UA sends the INVITE again to OpenSER, this time with the requested credentials. > > > 6. OpenSER forwards the new INVITE to the _OTHER_ proxy in the dispatcher list. > > > > > > This is where things start to fall over. This proxy has not issued a challenge request yet so it _again_ sends a digest challenge back to the UA through OpenSER. The UA does not like to receive a second challenge to what it's already sent and gives up. > > > > > > If you set the algorithm to something else besides 4, say 0, to hash to the callid, then OpenSER tries to use an arbitrary proxy from the list. Given that it's hashed against the callid it doesn't switch proxies like it does above in mid call. However, if that proxy is down, it _NEVER_ tries another proxy, thus rending the dispatcher module as a load balancer completely useless. > > > > > > If this module was designed to be a load balancer, how have people gotten it to work? I find it incredibly hard to believe that the dispatcher module was not designed with the above scenarios in mind. I'm so frustrated because what I am trying to do is so simple (had issues with SRV behaving in a similar way). Send requests from OpenSER to another proxy/pbx in a redundant and load balanced fashion. If a system is down, simply go to the next! It's that simple! > > > > > > Btw, the 'proxy' that OpenSER is forwarding too is Asterisk (yeah ok so it isn't a proxy) and the UA's are Polycom 301/501/601 phones. > > > > > > Help very much appreciated! > > > > > > Thanks. > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > 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 > > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

1 0

[Serusers] NAT type detection
by Ladislav Andel 25 Nov '05

25 Nov '05

Hello, I've been using Asus WL-500G as NAT/Router device for my testing purposes with SER. I implemented MyStun server which is running on the same OS as SER on 192.168.1.50. I also have two IP addresses on the server which are 192.168.1.50-51 for MyStun functionality. When I start MyStun it listens on default ports 3478 an 3479. UA behind the NAT is Xlite so the STUN implemetation of this softphone should be ok. Question 1: Where should I be able to determine/see what type of NAT I'm behind with X-lite? I'm trying to combine NAThelper/RTPproxy with STUN and use STUN if there is no symmetric NAT. Question 2: Could you advice me what tools I should use for debugging functionality of STUN, please? Hopefully, everything is clear. Ladislav

2 1

RE: [Users] Dispatcher module - Does it actually work?
by Douglas Garstang 25 Nov '05

25 Nov '05

Correct me if I am wrong here, but the sketchy docs at openser.org seem to imply that I am going to need to have multiple rows in my mysql table 'gw' for every single user who places calls? Is that correct? Is there some way to trim this down? -----Original Message----- From: Andreas Granig [mailto:andreas.granig@inode.info] Sent: Thu 11/24/2005 5:24 PM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Dispatcher module - Does it actually work? Douglas Garstang wrote: > Andy, no. No one has suggested using the lcr module, and I wasn't even aware of it until you mentioned it just now. I took a look at the module docs, they're way beyond my current understanding. it. I'll have to do some reading. Does it allow you to try another route when one is not responding? Yes. You have to define a set of destinations which you can load in the script using load_gws(), and every call to next_gw() sets the ip/port of another destination as ruri, until no more destinations are available. You can find an example here: http://linguin.org/ser/lcr_cap/lcr_config_example.txt It's for a lcr-version which I have extended a little to be able to group destinations according to their capabilities (http://linguin.org/ser.php#lcr-cap), but you should at least get the idea on how to do it with the vanilla lcr module (just leave out the argument in load_gws()). Andy

2 1

[Serusers] MediaProxy exception (text version)
by Adrian Georgescu 25 Nov '05

25 Nov '05

Hello Patrick, The issue can be fixed by the attached patch (it will be also fixed in the next release). Alternatively use python2.3 or newer which works even with the existing code. Index: dispatcher.py =================================================================== RCS file: /home/cvs/cvsroot/mediaproxy/modules/dispatcher.py,v retrieving revision 1.28 diff -u -r1.28 dispatcher.py --- dispatcher.py 25 Jul 2005 11:18:17 -0000 1.28 +++ dispatcher.py 25 Nov 2005 05:03:47 -0000 @@ -579,7 +579,7 @@ else: self.address = MediaProxyAddress(url) # implement this def __setattr__(self, name, value): - readonly = ('address') + readonly = ('address',) if name in readonly and self.__dict__.has_key(name): raise AttributeError, "'%s' is a readonly attribute" % name self.__dict__[name] = value >>>>>>>>>> Patrick Jordan-Smith pjs at quicksilver.co.nz Fri Nov 18 02:46:18 CET 2005 Hi, When I have a outgoing INVITE my mediaproxy crashes :( See the following: Nov 18 14:29:00 sip1 proxydispatcher[23201]: command request b566cc84-cd467d92-da2c547b at 202.89.130.40 202.89.130.40:2252:audio 202.89.130.40 sip.qsi.net.nz remote eth0.1 remote PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054 info=from:728728 at sip.qsi.net.nz,to:8378 at sip.qsi.net.nz,fromtag:60DCA320-5F6D23C9,totag: Nov 18 14:29:00 sip1 proxydispatcher[23201]: ---------------------------------------- Nov 18 14:29:00 sip1 proxydispatcher[23201]: Exception happened during processing of request from Nov 18 14:29:00 sip1 proxydispatcher[23201]: Traceback (most recent call last): Nov 18 14:29:00 sip1 proxydispatcher[23201]: File "/usr/lib/python2.2/SocketServer.py", line 458, in process_request_thread Nov 18 14:29:00 sip1 proxydispatcher[23201]: self.finish_request(request, client_address) Nov 18 14:29:00 sip1 proxydispatcher[23201]: File "/usr/lib/python2.2/SocketServer.py", line 253, in finish_request Nov 18 14:29:00 sip1 proxydispatcher[23201]: self.RequestHandlerClass(request, client_address, self)

1 0

[Serusers] MediaProxy exception
by Adrian Georgescu 25 Nov '05

25 Nov '05

Hello Patrick, The issue can be fixed by the attached patch (it will be also fixed in the next release). Alternatively use python2.3 or newer which works even with the existing code. >>>>>>>>>> Patrick Jordan-Smith pjs at quicksilver.co.nz Fri Nov 18 02:46:18 CET 2005 Hi, When I have a outgoing INVITE my mediaproxy crashes :( See the following: Nov 18 14:29:00 sip1 proxydispatcher[23201]: command request b566cc84-cd467d92-da2c547b at 202.89.130.40 202.89.130.40:2252:audio 202.89.130.40 sip.qsi.net.nz remote eth0.1 remote PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054 info=from:728728 at sip.qsi.net.nz,to:8378 at sip.qsi.net.nz,fromtag:60DCA320-5F6D23C9,totag: Nov 18 14:29:00 sip1 proxydispatcher[23201]: ---------------------------------------- Nov 18 14:29:00 sip1 proxydispatcher[23201]: Exception happened during processing of request from Nov 18 14:29:00 sip1 proxydispatcher[23201]: Traceback (most recent call last): Nov 18 14:29:00 sip1 proxydispatcher[23201]: File "/usr/lib/python2.2/SocketServer.py", line 458, in process_request_thread Nov 18 14:29:00 sip1 proxydispatcher[23201]: self.finish_request(request, client_address) Nov 18 14:29:00 sip1 proxydispatcher[23201]: File "/usr/lib/python2.2/SocketServer.py", line 253, in finish_request Nov 18 14:29:00 sip1 proxydispatcher[23201]: self.RequestHandlerClass(request, client_address, self)

1 0

RE: [Users] Dispatcher module - Does it actually work?
by Douglas Garstang 25 Nov '05

25 Nov '05

Hashing over call-id may work to some degree, and I'll investigate it more, but hashing over the From: means that the same caller will always get the same Asterisk box, and your redundancy goes down the toilet. Actually I guess it's down the toilet anyway. The bottom line seems to be that it doesn't matter whether you use the dispatcher or SRV records to distribute load, OpenSER will never try to connect to anything else on failure and at least for us, that makes it use in a production system, not an option. -----Original Message----- From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] Sent: Thu 11/24/2005 12:19 PM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Dispatcher module - Does it actually work? On 11/24/05 19:53, Douglas Garstang wrote: > The request URI from the phones is always going to be the same. Doesn't that mean that OpenSER will always just use the same Asterisk box then? if you do hash over r-uri, then yes, all messages will go to same destination. > I'm trying to round-robin as evenly (as possible) and distribute call load amonst multiple Asterisk systems. > In this case hash over call-id or from is more indicated. Cheers, Daniel > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Sent: Thu 11/24/2005 10:25 AM > To: Douglas Garstang > Cc: users(a)openser.org > Subject: Re: [Users] Dispatcher module - Does it actually work? > > > > On 11/24/05 01:37, Douglas Garstang wrote: > > After many many fruitless hours spent on trying to get OpenSER to work with DNS SRV records (I made a post to the group earlier today on this), I stumbled across the dispatcher module, and have been trying to get it to work instead. > > > > The module was designed to load balance, right? It doesn't seem to have been designed very well, unless I am missing something. > > > If you read carefully the documentation of the module, you will see the > scope and limitations of the module: > http://openser.org/docs/modules/1.0.x/dispatcher.html > > If something is not according to the documentation, then it is a bug. > Stateless means that the module does not keep any state, so it cannot > select the same destination using algorithm 4 (round robin) after a > authentication request, choose another algorithm (hashing r-uri, > callid). Also, being stateless, it cannot try the next address if the > selected destination fails. > > The module was not designed to be load balancer, but a request > dispatcher. Load balancing is something more complex. > > In the last weeks I added destination failover support in dispatcher > (serial forking within the destination set, with auto-detection of > failed destinations). It will be on cvs very soon, I have to update the > documentation and do some testing. > > Cheers, > Daniel > > > > > If you set the algorithm with ds_select_dst() to 4, round robin, it doesn't work! Here's what happens. > > > > 1. UA sends INVITE to OpenSER > > 2. OpenSER forwards INVITE to the first proxy in the dispatcher list. > > 3. Proxy sends back a digest challenge for authorisation credentials to OpenSER. > > 4. OpenSER forwards the challenge back to the UA. > > 5. The UA sends the INVITE again to OpenSER, this time with the requested credentials. > > 6. OpenSER forwards the new INVITE to the _OTHER_ proxy in the dispatcher list. > > > > This is where things start to fall over. This proxy has not issued a challenge request yet so it _again_ sends a digest challenge back to the UA through OpenSER. The UA does not like to receive a second challenge to what it's already sent and gives up. > > > > If you set the algorithm to something else besides 4, say 0, to hash to the callid, then OpenSER tries to use an arbitrary proxy from the list. Given that it's hashed against the callid it doesn't switch proxies like it does above in mid call. However, if that proxy is down, it _NEVER_ tries another proxy, thus rending the dispatcher module as a load balancer completely useless. > > > > If this module was designed to be a load balancer, how have people gotten it to work? I find it incredibly hard to believe that the dispatcher module was not designed with the above scenarios in mind. I'm so frustrated because what I am trying to do is so simple (had issues with SRV behaving in a similar way). Send requests from OpenSER to another proxy/pbx in a redundant and load balanced fashion. If a system is down, simply go to the next! It's that simple! > > > > Btw, the 'proxy' that OpenSER is forwarding too is Asterisk (yeah ok so it isn't a proxy) and the UA's are Polycom 301/501/601 phones. > > > > Help very much appreciated! > > > > Thanks. > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > 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 >

3 2

[Serusers] ser and TCP/TLS experience
by Klaus Darilion 25 Nov '05

25 Nov '05

Hi all! Regarding SIP and TCP/TLS there are often discussions how many concurrently TCP/TLS connections a server can handle. Does some one of have experience with long living TCP connections? Are there performance issues? I also want to do some tests, loading ser with lots of TCP connections. Do you know any test tools for this? regards klaus

2 1

RE: [Users] Dispatcher module - Does it actually work?
by Douglas Garstang 25 Nov '05

25 Nov '05

Well, after 12 hours at this, my brain hurts. Andreas, I've been trying to get the lcr module to work. I'd really appreciate some pointers or help. :) OpenSER is not attempting to contact the second gw. It just sits there forever trying to connect to the first. I even tried setting fr_inv_timer, but it has no effect. The openser.cfg is probably full of errors too, but I can definitely see it trying to connect to the first gw, and it never tries the second. I purposely shut Asterisk down on the first. Here's my mysql database tables: mysql> select * from lcr; +--------+----------+--------+----------+ | prefix | from_uri | grp_id | priority | +--------+----------+--------+----------+ | | % | 1 | 1 | | | % | 2 | 2 | +--------+----------+--------+----------+ 2 rows in set (0.00 sec) mysql> select * from gw; +-----------+--------+-----------+------+------------+-----------+--------+ | gw_name | grp_id | ip_addr | port | uri_scheme | transport | prefix | +-----------+--------+-----------+------+------------+-----------+--------+ | asterisk1 | 1 | 369797312 | 5060 | 1 | 1 | | | asterisk2 | 2 | 386574528 | 5060 | 1 | 1 | | +-----------+--------+-----------+------+------------+-----------+--------+ 2 rows in set (0.00 sec) And here's the relevant snippets from openser.cfg: modparam("lcr","fr_inv_timer",1) # Standard openser.cfg down to here lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # I added the line below route(2); # More standard openser.cfg lines # 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"); #I added the route[2] block route[2] { if (!load_gws()) { sl_send_reply("500", "Server Internal Error - Cannot load gateways"); xlog ("L_INFO","Not found"); return; } else { xlog ("L_INFO","Gateways loaded"); } if (!next_gw()) { sl_send_reply("503", "Service not available, no gateways found"); xlog ("L_INFO","No more gateways"); return; } else { xlog ("L_INFO","Found a gateway"); } xlog ("L_INFO","Calling t_relay"); t_on_failure("2"); t_relay(); exit; } failure_route[2] { if(method == "INVITE" && t_check_status("408|500|503")) { if (!next_gw()) { t_reply("503", "Service not available, no more gateways"); return; } t_on_failure("2"); t_relay(); } } -----Original Message----- From: Andreas Granig [mailto:andreas.granig@inode.info] Sent: Thu 11/24/2005 5:24 PM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Dispatcher module - Does it actually work? Douglas Garstang wrote: > Andy, no. No one has suggested using the lcr module, and I wasn't even aware of it until you mentioned it just now. I took a look at the module docs, they're way beyond my current understanding. it. I'll have to do some reading. Does it allow you to try another route when one is not responding? Yes. You have to define a set of destinations which you can load in the script using load_gws(), and every call to next_gw() sets the ip/port of another destination as ruri, until no more destinations are available. You can find an example here: http://linguin.org/ser/lcr_cap/lcr_config_example.txt It's for a lcr-version which I have extended a little to be able to group destinations according to their capabilities (http://linguin.org/ser.php#lcr-cap), but you should at least get the idea on how to do it with the vanilla lcr module (just leave out the argument in load_gws()). Andy

1 0

RE: [Users] Dispatcher module - Does it actually work?
by Douglas Garstang 25 Nov '05

25 Nov '05

Andy (or anyone who cares to jump in here), From the openser docs: "A gateway matches a request if user part of Request-URI and "From" URI match the prefix and From pattern of the gateway. " Does that mean that I need to have a gw table entry for every single possible combination of every user making calls and every number in the universe they may dial? That's insane! I must be missing something here. Oh, and what is an 'RPID URI'???? Never heard of it and I can't find it mentioned anywhere outside the lcr module. -----Original Message----- From: Douglas Garstang Sent: Thu 11/24/2005 7:45 PM To: users(a)openser.org Cc: Subject: RE: [Users] Dispatcher module - Does it actually work? Correct me if I am wrong here, but the sketchy docs at openser.org seem to imply that I am going to need to have multiple rows in my mysql table 'gw' for every single user who places calls? Is that correct? Is there some way to trim this down? -----Original Message----- From: Andreas Granig [mailto:andreas.granig@inode.info] Sent: Thu 11/24/2005 5:24 PM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Dispatcher module - Does it actually work? Douglas Garstang wrote: > Andy, no. No one has suggested using the lcr module, and I wasn't even aware of it until you mentioned it just now. I took a look at the module docs, they're way beyond my current understanding. it. I'll have to do some reading. Does it allow you to try another route when one is not responding? Yes. You have to define a set of destinations which you can load in the script using load_gws(), and every call to next_gw() sets the ip/port of another destination as ruri, until no more destinations are available. You can find an example here: http://linguin.org/ser/lcr_cap/lcr_config_example.txt It's for a lcr-version which I have extended a little to be able to group destinations according to their capabilities (http://linguin.org/ser.php#lcr-cap), but you should at least get the idea on how to do it with the vanilla lcr module (just leave out the argument in load_gws()). Andy

1 0

RE: [Users] Dispatcher module - Does it actually work?
by Douglas Garstang 25 Nov '05

25 Nov '05

Thanks Andy. I'll take a look. Considering I can't even find documentation for what a route() block is, I'm in for a lot of reading. *sigh* Docs.... docs.... docs.... (or lack thereof) -----Original Message----- From: Andreas Granig [mailto:andreas.granig@inode.info] Sent: Thu 11/24/2005 5:24 PM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Dispatcher module - Does it actually work? Douglas Garstang wrote: > Andy, no. No one has suggested using the lcr module, and I wasn't even aware of it until you mentioned it just now. I took a look at the module docs, they're way beyond my current understanding. it. I'll have to do some reading. Does it allow you to try another route when one is not responding? Yes. You have to define a set of destinations which you can load in the script using load_gws(), and every call to next_gw() sets the ip/port of another destination as ruri, until no more destinations are available. You can find an example here: http://linguin.org/ser/lcr_cap/lcr_config_example.txt It's for a lcr-version which I have extended a little to be able to group destinations according to their capabilities (http://linguin.org/ser.php#lcr-cap), but you should at least get the idea on how to do it with the vanilla lcr module (just leave out the argument in load_gws()). Andy

1 0

[Serusers] How to control a call PSTN in SER ?
by Truong Chi Thanh 25 Nov '05

25 Nov '05

3 2

RE: [Users] Dispatcher module - Does it actually work?
by Douglas Garstang 25 Nov '05

25 Nov '05

Andy, no. No one has suggested using the lcr module, and I wasn't even aware of it until you mentioned it just now. I took a look at the module docs, they're way beyond my current understanding. it. I'll have to do some reading. Does it allow you to try another route when one is not responding? Doug. -----Original Message----- From: Andreas Granig [mailto:andreas.granig@inode.info] Sent: Thu 11/24/2005 2:25 PM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Dispatcher module - Does it actually work? Douglas Garstang wrote: > Hashing over call-id may work to some degree, and I'll investigate it more, but hashing over the From: means that the same caller will always get the same Asterisk box, and your redundancy goes down the toilet. > > Actually I guess it's down the toilet anyway. The bottom line seems to be that it doesn't matter whether you use the dispatcher or SRV records to distribute load, OpenSER will never try to connect to anything else on failure and at least for us, that makes it use in a production system, not an option. I didn't follow the whole thread, so maybe this has been suggested before, but have you ever thought about using the lcr module for load balancing? In combination with failure routes for failover this works very well for me. Andy

2 1

[Serusers] harry's project
by bosinclar＠laposte.net 25 Nov '05

25 Nov '05

Hello, here is an other diagram for people who don't yet understand what i expect to do. Look at sip_call_flow.png file i wish to substitute ondo sip server with ser and ondo pbx with asterisk . ondo sip server is able to do far-end near-end nat I guess ser too. I do hope i will find some people who help me to configure that . Regards Harry Accédez au courrier électronique de La Poste : www.laposte.net ; 3615 LAPOSTENET (0,34/mn) ; tél : 08 92 68 13 50 (0,34/mn)

1 0

[Users] harry's project
by harry gaillac 25 Nov '05

25 Nov '05

Hello, here is an other diagram for people who don't yet understand what i expect to do. Look at sip_call_flow.png file i wish to substitute ondo sip server with ser and ondo pbx with asterisk . ondo sip server is able to do far-end near-end nat I guess ser too. I do hope i will find some people who help me to configure that . Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Users] harry's project
by harry gaillac 25 Nov '05

25 Nov '05

Hello, here is an other diagram for people who don't yet understand what i expect to do. Look at sip_call_flow.png file i wish to substitute ondo sip server with ser and ondo pbx with asterisk . ondo sip server is able to do far-end near-end nat I guess ser too. I do hope i will find some people who help me to configure that . Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Users] harry's project
by harry gaillac 25 Nov '05

25 Nov '05

Hello, here is an other diagram for people who don't yet understand what i expect to do. Look at sip_call_flow.png file i wish to substitute ondo sip server with ser and ondo pbx with asterisk . ondo sip server is able to do far-end near-end nat I guess ser too. I do hope i will find some people who help me to configure that . Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

Re: [Users] Dispatcher module - Does it actually work?
by Daniel-Constantin Mierla 24 Nov '05

24 Nov '05

On 11/24/05 19:50, Douglas Garstang wrote: > I am doing record routing as far as I know. I've pretty much still got a default openser.cfg, with a few modifications. The record route statement is still in there towards the top. And besides, it's OpenSER that's adding itself to the record-route header, so the route is always going to be the same anyway. I don't follow how that helps. > The Record-Route ensures that all messages within same dialog will follow same path. So the BYE will go same way (or opposite if callee sends it) as the first INVITE. > > I'm still new at this.... at ideas how I'd tell if it was the first invite? Check Cseq? > The first invite of a dialog has no To-tag. There is a function in uri module to check that. Cheers, Daniel > > Thanks. > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Sent: Thu 11/24/2005 10:32 AM > To: Douglas Garstang > Cc: users(a)openser.org > Subject: Re: [Users] Dispatcher module - Does it actually work? > > > > > On 11/24/05 02:16, Douglas Garstang wrote: > > Actually, you know, now that I think about it, what I think should happen is that OpenSER should _never_ switch to another proxy when the Dialog/Callid is the same. This just confuses everyone. > > > Actually you should use the dispatcher only for the first INVITE and > CANCEL. To ensure that all the other requests within the dialog follow > same path, use record-routing. > > Cheers, > Daniel > > > > > It doesn't matter if you use the dispatcher module or DNS SRV. Either one when configured for round robin has the potential to send a second INVITE with requested auth credentials (with the same call-id) to another proxy server. This is bad. This was happening with DNS SRV and it's why I started looking at the dispatcher module instead. > > > > There has to be a way to do this.... > > > > Doug > > > > -----Original Message----- > > From: Douglas Garstang > > Sent: Wed 11/23/2005 4:37 PM > > To: users(a)openser.org > > Cc: > > Subject: [Users] Dispatcher module - Does it actually work? > > > > > > > > After many many fruitless hours spent on trying to get OpenSER to work with DNS SRV records (I made a post to the group earlier today on this), I stumbled across the dispatcher module, and have been trying to get it to work instead. > > > > > > The module was designed to load balance, right? It doesn't seem to have been designed very well, unless I am missing something. > > > > > > If you set the algorithm with ds_select_dst() to 4, round robin, it doesn't work! Here's what happens. > > > > 1. UA sends INVITE to OpenSER > > 2. OpenSER forwards INVITE to the first proxy in the dispatcher list. > > 3. Proxy sends back a digest challenge for authorisation credentials to OpenSER. > > 4. OpenSER forwards the challenge back to the UA. > > 5. The UA sends the INVITE again to OpenSER, this time with the requested credentials. > > 6. OpenSER forwards the new INVITE to the _OTHER_ proxy in the dispatcher list. > > > > This is where things start to fall over. This proxy has not issued a challenge request yet so it _again_ sends a digest challenge back to the UA through OpenSER. The UA does not like to receive a second challenge to what it's already sent and gives up. > > > > > > If you set the algorithm to something else besides 4, say 0, to hash to the callid, then OpenSER tries to use an arbitrary proxy from the list. Given that it's hashed against the callid it doesn't switch proxies like it does above in mid call. However, if that proxy is down, it _NEVER_ tries another proxy, thus rending the dispatcher module as a load balancer completely useless. > > > > > > If this module was designed to be a load balancer, how have people gotten it to work? I find it incredibly hard to believe that the dispatcher module was not designed with the above scenarios in mind. I'm so frustrated because what I am trying to do is so simple (had issues with SRV behaving in a similar way). Send requests from OpenSER to another proxy/pbx in a redundant and load balanced fashion. If a system is down, simply go to the next! It's that simple! > > > > > > Btw, the 'proxy' that OpenSER is forwarding too is Asterisk (yeah ok so it isn't a proxy) and the UA's are Polycom 301/501/601 phones. > > > > > > Help very much appreciated! > > > > Thanks. > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > >

1 0

[Users] Dispatcher module algorithms
by Douglas Garstang 24 Nov '05

24 Nov '05

I'm trying to understand a little better the docs for the algorithms in the dispatcher module. What exactly does 'hash over call-id', or 'hash over from uri' mean? The emphasis here is on the use of the word 'hash'. For a given call-id or a given from uri, I think I understand that the selected proxy will always be the same. However, is there any way to determine how this will distribute call volume between proxies? Is there some magic formula that is used (ie the hash) to select the proxy? I'd like to know what it is so I could run some numbers through it and try and understand how it would break up call placement to proxies for certain input values. Doug.

2 1

RE: [Users] Dispatcher module - Does it actually work?
by Douglas Garstang 24 Nov '05

24 Nov '05

The request URI from the phones is always going to be the same. Doesn't that mean that OpenSER will always just use the same Asterisk box then? I'm trying to round-robin as evenly (as possible) and distribute call load amonst multiple Asterisk systems. -----Original Message----- From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] Sent: Thu 11/24/2005 10:25 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Dispatcher module - Does it actually work? On 11/24/05 01:37, Douglas Garstang wrote: > After many many fruitless hours spent on trying to get OpenSER to work with DNS SRV records (I made a post to the group earlier today on this), I stumbled across the dispatcher module, and have been trying to get it to work instead. > > The module was designed to load balance, right? It doesn't seem to have been designed very well, unless I am missing something. > If you read carefully the documentation of the module, you will see the scope and limitations of the module: http://openser.org/docs/modules/1.0.x/dispatcher.html If something is not according to the documentation, then it is a bug. Stateless means that the module does not keep any state, so it cannot select the same destination using algorithm 4 (round robin) after a authentication request, choose another algorithm (hashing r-uri, callid). Also, being stateless, it cannot try the next address if the selected destination fails. The module was not designed to be load balancer, but a request dispatcher. Load balancing is something more complex. In the last weeks I added destination failover support in dispatcher (serial forking within the destination set, with auto-detection of failed destinations). It will be on cvs very soon, I have to update the documentation and do some testing. Cheers, Daniel > > If you set the algorithm with ds_select_dst() to 4, round robin, it doesn't work! Here's what happens. > > 1. UA sends INVITE to OpenSER > 2. OpenSER forwards INVITE to the first proxy in the dispatcher list. > 3. Proxy sends back a digest challenge for authorisation credentials to OpenSER. > 4. OpenSER forwards the challenge back to the UA. > 5. The UA sends the INVITE again to OpenSER, this time with the requested credentials. > 6. OpenSER forwards the new INVITE to the _OTHER_ proxy in the dispatcher list. > > This is where things start to fall over. This proxy has not issued a challenge request yet so it _again_ sends a digest challenge back to the UA through OpenSER. The UA does not like to receive a second challenge to what it's already sent and gives up. > > If you set the algorithm to something else besides 4, say 0, to hash to the callid, then OpenSER tries to use an arbitrary proxy from the list. Given that it's hashed against the callid it doesn't switch proxies like it does above in mid call. However, if that proxy is down, it _NEVER_ tries another proxy, thus rending the dispatcher module as a load balancer completely useless. > > If this module was designed to be a load balancer, how have people gotten it to work? I find it incredibly hard to believe that the dispatcher module was not designed with the above scenarios in mind. I'm so frustrated because what I am trying to do is so simple (had issues with SRV behaving in a similar way). Send requests from OpenSER to another proxy/pbx in a redundant and load balanced fashion. If a system is down, simply go to the next! It's that simple! > > Btw, the 'proxy' that OpenSER is forwarding too is Asterisk (yeah ok so it isn't a proxy) and the UA's are Polycom 301/501/601 phones. > > Help very much appreciated! > > Thanks. > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

2 1

[Serusers] SER & SIPSAK: SER registration limitations
by Ramin Nikaeen 24 Nov '05

24 Nov '05

Friends, We are testing our SER set up using SIPSAK. Are there any limitations in terms of the number of registered subscribers that can be kept by SER in memory or in database? We tried with both db_mod 0 or 1. After some 40,000 registrations, SER starts Giving "No memory left" messages and sometimes terminates as well. And with db_mode=1, SER can not startup again, giving We checked the machine and out of 2GBytes of memory only about 10Mbytes Were used up. Any ideas? ramin ------------------------------------------------------------------------ ---------------- SER crash errors with db_mode= 0 or 1 ------------------------------------------------------------------------ ---------------- modparam("usrloc", "db_mode", 0) OR modparam("usrloc", "db_mode", 1) 12(24376) new_ucontact(): No memory left 12(24376) mem_insert_ucontact(): Can't create new contact 12(24376) insert_ucontact(): Error while inserting contact 12(24376) insert(): Error while inserting contact 12(24376) contacts(): Error while inserting record 12(24376) new_ucontact(): No memory left 12(24376) mem_insert_ucontact(): Can't create new contact 12(24376) insert_ucontact(): Error while inserting contact 12(24376) insert(): Error while inserting contact 12(24376) contacts(): Error while inserting record 19(24383) ERROR: tcpconn_new: mem. allocation failure 19(24383) ERROR: tcp_main_loop: tcpconn_new failed, closing socket 14(24378) Binding '2224449999335607','sip:2224449999335607@172.16.15.201:3006' has expired 14(24378) Binding '2224449999335608','sip:2224449999335608@172.16.15.201:3006' has expired 14(24378) Binding '2224449999335609','sip:2224449999335609@172.16.15.201:3006' has expired 14(24378) Binding '2224449999335610','sip:2224449999335610@172.16.15.201:3006' has expired 14(24378) Binding '2224449999127988','sip:2224449999127988@172.16.15.203:1815' has expired 12(24376) new_ucontact(): No memory left 12(24376) mem_insert_ucontact(): Can't create new contact 12(24376) insert_ucontact(): Error while inserting contact 12(24376) insert(): Error while inserting contact 12(24376) contacts(): Error while inserting record 19(24383) ERROR: tcpconn_new: mem. allocation failure 19(24383) ERROR: tcp_main_loop: tcpconn_new failed, closing socket 1(24365) 10(24374) INFO: signal 15 received 13(24377) INFO: signal 15 received 18(24382) 3(24367) INFO: signal 15 received INFO: signal 15 received 4(24368) 19(24383) INFO: signal 15 received 6(24370) INFO: signal 15 received ------------------------------------------------------------------------ ---------------- SER Startup error with db_mode=1 ------------------------------------------------------------------------ ---------------- And when I try with modparam("usrloc", "db_mode", 1) SER exits and to restart it, it gives me the following error and can't start: . [root@ser]/usr/local/ser/etc/ser/logs> 0(0) Maxfwd module- initializing textops - initializing callidselect - initializing callidselect: module initialized successfully! exec - initializing 0(0) AVPops - initializing 0(0) permissions - initializing 0(0) Default allow file (/usr/local/ser/etc/ser/permissions.allow) parsed 0(0) Default deny file (/usr/local/ser/etc/ser/permissions.deny) not found => empty rule set acc - initializing 0(0) info: mediaproxy: loaded SIP asymmetric clients file containing 2 entries. 0(0) info: mediaproxy: loaded RTP asymmetric clients file containing 2 entries. 0(0) convert_row(): No memory left 0(0) convert_rows(): Error while converting row #2549 0(0) convert_result(): Error while converting rows 0(0) store_result(): Error while converting result 0(0) preload_udomain(): Error while doing db_query 0(0) register_udomain(): Error while preloading domain 'location' 0(0) domain_fixup(): Error while registering domain 0(0) ERROR: fix_expr : fix_actions error ERROR: error -1 while trying to fix configuration [root@ser]/usr/local/ser/etc/ser/logs>service ser start Starting Sip Express Router (SER) server: serListening on udp: 127.0.0.1 [127.0.0.1]:5

3 2

[Serusers] NAT again
by bosinclar＠laposte.net 24 Nov '05

24 Nov '05

Hello, In fact I wish ser act like siproxd SER listen ppp0 and eth0 requests are sent to eth0 with private addresses so SER have to fix contact with ppp0 address . Regards Accédez au courrier électronique de La Poste : www.laposte.net ; 3615 LAPOSTENET (0,34/mn) ; tél : 08 92 68 13 50 (0,34/mn)

1 0

RE: [Users] Dispatcher module - Does it actually work?
by Douglas Garstang 24 Nov '05

24 Nov '05

Actually, you know, now that I think about it, what I think should happen is that OpenSER should _never_ switch to another proxy when the Dialog/Callid is the same. This just confuses everyone. It doesn't matter if you use the dispatcher module or DNS SRV. Either one when configured for round robin has the potential to send a second INVITE with requested auth credentials (with the same call-id) to another proxy server. This is bad. This was happening with DNS SRV and it's why I started looking at the dispatcher module instead. There has to be a way to do this.... Doug -----Original Message----- From: Douglas Garstang Sent: Wed 11/23/2005 4:37 PM To: users(a)openser.org Cc: Subject: [Users] Dispatcher module - Does it actually work? After many many fruitless hours spent on trying to get OpenSER to work with DNS SRV records (I made a post to the group earlier today on this), I stumbled across the dispatcher module, and have been trying to get it to work instead. The module was designed to load balance, right? It doesn't seem to have been designed very well, unless I am missing something. If you set the algorithm with ds_select_dst() to 4, round robin, it doesn't work! Here's what happens. 1. UA sends INVITE to OpenSER 2. OpenSER forwards INVITE to the first proxy in the dispatcher list. 3. Proxy sends back a digest challenge for authorisation credentials to OpenSER. 4. OpenSER forwards the challenge back to the UA. 5. The UA sends the INVITE again to OpenSER, this time with the requested credentials. 6. OpenSER forwards the new INVITE to the _OTHER_ proxy in the dispatcher list. This is where things start to fall over. This proxy has not issued a challenge request yet so it _again_ sends a digest challenge back to the UA through OpenSER. The UA does not like to receive a second challenge to what it's already sent and gives up. If you set the algorithm to something else besides 4, say 0, to hash to the callid, then OpenSER tries to use an arbitrary proxy from the list. Given that it's hashed against the callid it doesn't switch proxies like it does above in mid call. However, if that proxy is down, it _NEVER_ tries another proxy, thus rending the dispatcher module as a load balancer completely useless. If this module was designed to be a load balancer, how have people gotten it to work? I find it incredibly hard to believe that the dispatcher module was not designed with the above scenarios in mind. I'm so frustrated because what I am trying to do is so simple (had issues with SRV behaving in a similar way). Send requests from OpenSER to another proxy/pbx in a redundant and load balanced fashion. If a system is down, simply go to the next! It's that simple! Btw, the 'proxy' that OpenSER is forwarding too is Asterisk (yeah ok so it isn't a proxy) and the UA's are Polycom 301/501/601 phones. Help very much appreciated! Thanks.

2 1

[Users] Dispatcher module - Does it actually work?
by Douglas Garstang 24 Nov '05

24 Nov '05

After many many fruitless hours spent on trying to get OpenSER to work with DNS SRV records (I made a post to the group earlier today on this), I stumbled across the dispatcher module, and have been trying to get it to work instead. The module was designed to load balance, right? It doesn't seem to have been designed very well, unless I am missing something. If you set the algorithm with ds_select_dst() to 4, round robin, it doesn't work! Here's what happens. 1. UA sends INVITE to OpenSER 2. OpenSER forwards INVITE to the first proxy in the dispatcher list. 3. Proxy sends back a digest challenge for authorisation credentials to OpenSER. 4. OpenSER forwards the challenge back to the UA. 5. The UA sends the INVITE again to OpenSER, this time with the requested credentials. 6. OpenSER forwards the new INVITE to the _OTHER_ proxy in the dispatcher list. This is where things start to fall over. This proxy has not issued a challenge request yet so it _again_ sends a digest challenge back to the UA through OpenSER. The UA does not like to receive a second challenge to what it's already sent and gives up. If you set the algorithm to something else besides 4, say 0, to hash to the callid, then OpenSER tries to use an arbitrary proxy from the list. Given that it's hashed against the callid it doesn't switch proxies like it does above in mid call. However, if that proxy is down, it _NEVER_ tries another proxy, thus rending the dispatcher module as a load balancer completely useless. If this module was designed to be a load balancer, how have people gotten it to work? I find it incredibly hard to believe that the dispatcher module was not designed with the above scenarios in mind. I'm so frustrated because what I am trying to do is so simple (had issues with SRV behaving in a similar way). Send requests from OpenSER to another proxy/pbx in a redundant and load balanced fashion. If a system is down, simply go to the next! It's that simple! Btw, the 'proxy' that OpenSER is forwarding too is Asterisk (yeah ok so it isn't a proxy) and the UA's are Polycom 301/501/601 phones. Help very much appreciated! Thanks.

2 1

[Users] rewritehostport function!
by messa＠innsof.com 24 Nov '05

24 Nov '05

Hi all can i use the rewritehostport to make openser communicate with another server? Whether it's possible, how can i do it? Serge

2 1

[Serusers] NAT again
by harry gaillac 24 Nov '05

24 Nov '05

I need to add some functions in my ser.cfg for nat support. I use my open(ser) as outbound proxy with two interfaces ppp0 and eth0. sip agents send requests to eth0. I want my packets passing through the register and invite blocks by fixing contact with a public ip. All packets from private network must be sent to eth0. Any suggestions Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Users] NAT again
by harry gaillac 24 Nov '05

24 Nov '05

I need to add some functions in my ser.cfg for nat support. I use my open(ser) as outbound proxy with two interfaces ppp0 and eth0. sip agents send requests to eth0. I want my packets passing through the register and invite blocks by fixing contact with a public ip. All packets from private network must be sent to eth0. Any suggestions Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Users] Norway pub event
by Greger V. Teigre 24 Nov '05

24 Nov '05

Hi guys, Let's meet for another *SER (i.e. openser/ser) pub event on Thursday September 24 around 18. To avoid confusion, we stay with the venue Beer Palace at Aker Brygge. (Yes, Oslo, Norway) Suggestion for topic of the day ;-) : Ways of accessing header data and manipulate values from ser.cfg, including avps, avpops, textops... Do we need them all and isn't this really a big, complicated mess that is the result of ad-hoc needs for functionality? People with limited travel budgets may input their opinions on the list ;-) g-)

2 1

[Users] syntax for multihomed hosts
by harry gaillac 24 Nov '05

24 Nov '05

hello, what's the write syntax for mhomed=yes or mhomed=1 ? Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Users] SRV Lookups
by Douglas Garstang 24 Nov '05

24 Nov '05

All, I'm trying to get SRV lookups to work with OpenSER 1.0.0. I've exhausted every avenue of documentation including newsgroups, various forums etc. Using t_relay, OpenSER seems to be performing the correct DNS SRV queries and obtaining a list of hosts to try. Below are the relevant sections of my DNS zone file. We're talking UDP, and based on the priority and weights below, OpenSER should be basically doing a round-robin approach, and it seems to be. The problem occurs when one of the hosts is not contactable. OpenSER simply stops and does not try the next host in it's list. I would have thought that for OpenSER to fully support SRV lookups, it should try each until it is successful. After all, that's a large part of the whole point of SRV records. Does the t_relay() function have logic within itself to keep trying hosts until it gets a connection? What would be the definition a failure that would make it move onto the next host be anyway? Network failure certainly, but would a 'Not found' be considered a failure that would cause it to move on? Would anything other than an ACK be a failure? Is there some variable I can set or some function I can call that causes t_relay() to retry all hosts in it's SRV list? Without this feature, OpenSER's usefulness is very limited. Help very much appreciated. Doug bil-pdev-3 IN A 192.168.10.4 proxy1 IN CNAME bil-pdev-3 proxy2 IN CNAME bil-pdev-3 proxy.bil.voip.com. IN NAPTR 50 50 "s" "SIPS+D2T" "" _sips._tcp.proxy.bil.voip.com. proxy.bil.voip.com. IN NAPTR 60 60 "s" "SIP+D2U" "" _sip._udp.proxy.bil.voip.com. proxy.bil.voip.com. IN NAPTR 70 40 "s" "SIP+D2T" "" _sip._tcp.proxy.bil.voip.com. _sips._tcp.proxy.bil.voip.com. IN SRV 100 10 5061 proxy1.proxy.bil.voip.com. _sips._tcp.proxy.bil.voip.com. IN SRV 100 10 5061 proxy2.proxy.bil.voip.com. _sip._udp.proxy.bil.voip.com. IN SRV 100 10 5060 proxy1.proxy.bil.voip.com. ; This line and the next are SRV records being looked up _sip._udp.proxy.bil.voip.com. IN SRV 100 10 5060 proxy2.proxy.bil.voip.com. ; _sip._tcp.proxy.bil.voip.com. IN SRV 100 10 5060 proxy1.proxy.bil.voip.com. _sip._tcp.proxy.bil.voip.com. IN SRV 100 10 5060 proxy2.proxy.bil.voip.com. *._tcp IN SRV 0 0 0 . *._udp IN SRV 0 0 0 . ;

5 6

Re: [Users] textops modules
by harry gaillac 24 Nov '05

24 Nov '05

Remarque : message transféré en pièce jointe. ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Users] Please help...... how to configure outgoing calls to pstn gateway
by Lokesh Kumar 24 Nov '05

24 Nov '05

Hello Friends, I am having openser, some sip clients registered on it.how do I have to configure openser.cfg and/or pstn.cfg.Inorder to make call from sip clients to outside number. I can call from outside number to my sip clients. but other way around is not working. My mobile ---->-------->| PSTN GW |--------->------->|OPENSER|---------->--------->| SIP CLIENT| This is working fine My mobile <-------<-------| PSTN GW |<-------<-------|OPENSER|<--------<----------| SIP CLIENT| This case is not working I am not getting any log related to my outgoing call on openser , through ngrep. Please help, Regards Lokesh

2 4

[Serusers] Regading entry & exit function of stack & parser
by nitin ganjsinghani 24 Nov '05

24 Nov '05

i'm finding it very difficult to find the entry & exit functions of the parser & the stack moreover anyb can plz tell me the parser initialization function name as understanding the whole code is very tedious thanks nitin

2 1

Re: [Users] textops modules
by Bogdan-Andrei Iancu 24 Nov '05

24 Nov '05

I see....in this case you cannot use replace() as the public IP will all the time be different (the public IP of nat). Use the fix_nated_contact() function from nathelper module: if msg is detected from behind nat, do fix_nated_contact() before fwd to *. regards, bogdan harry gaillac wrote: >In fact I meant: > >all private ip of the network replace with a pubil ip. >what about substution ? > >if search("^Contact sip:user@private ip$)) { > replace_all ("private ip", "public ip"); >}; > >You understand my problem . > > asterisk > || > Internet=[Ser/nat]==private network > >When ser receive REGISTER or INVITE the contact in sip >hf would be fixed >so asterisk would not record sip agent with AOR like >sip:user@private ip. > > >Harry > >--- Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> a >écrit : > > > >>Hi Harry, >> >>yes you can do this, but not that the change will >>not be local visible >>(in further script processing). the change will >>actually apply only when >>the message is sent out on the network. >> >>regards, >>bogdan >> >>harry gaillac wrote: >> >> >> >>>Hello, >>> >>>How can I use textops module to rewrite all ip >>> >>> >>private >> >> >>>addresses in sip contact hf with a public ip >>> >>>I can't write the right regexp. >>> >>>if search("^Contact sip:user@private ip$)) { >>>replace_all ("private ip", "public ip"); >>>}; >>> >>>Harry >>> >>> >>> >>> >>> >>> >>> >>> >>___________________________________________________________________________ >> >> >> >>>Appel audio GRATUIT partout dans le monde avec le >>> >>> >>nouveau Yahoo! Messenger >> >> >>>Téléchargez cette version sur >>> >>> >>http://fr.messenger.yahoo.com >> >> >>>_______________________________________________ >>>Users mailing list >>>Users(a)openser.org >>>http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> >>> >> >> > > > > > > > >___________________________________________________________________________ >Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger >Téléchargez cette version sur http://fr.messenger.yahoo.com > >

1 0

Re: [Serusers] Problem with serweb...
by Karel Kozlik 24 Nov '05

24 Nov '05

Hello, sorry I don't know. I never meet similar problem. Use google or try ask on a PHP forum (e.g. http://news.php.net/) regards Karel 昂達 napsal(a): > Hello Karel > I have checked the apache log file in "/var/log/httpd/error_log" > the file show this: > -------------------------------------------------------------------------------------------------------------------------------------------------------------------- > PHP Warning: PHP Startup: Unable to load dynamic library > '/usr/lib/php/modules/classkit.so' - /usr/lib/php/modules/classkit.so: > cannot open shared object file: Permission denied in Unknown on line 0 > -------------------------------------------------------------------------------------------------------------------------------------------------------------------- > > Then i change classkit.so permission to "777" and change owner to "apache". > But the error also exist. > How to slove this problem ?? > > Thank you so much > > 昂達，hunter1104(a)tw-airnet.net > <mailto:hunter1104(a)tw-airnet.net> > 2005-11-24 > > ----- 收到以下來信內容 ----- > *寄件人:* Karel Kozlik <mailto:karel@iptel.org> > *收件人:* hunter1104 <mailto:hunter1104@tw-airnet.net> > *時間:* 2005-11-22, 18:21:28 > *主題:* Re: [Serusers] Problem with serweb... > > Hi, > make sure if you are changeing right php.ini file. May be there is more > php.ini files in your system in various locations or may be php is > searching for php.ini in diferent directory. > > Also check log files: > - syslog > - apache log > - php log > > for messages related to this problem. > > regards > Karel > > 昂達 napsal(a): > > Hello karel > > > > I have tried the method that you offered > > In my phpinfo() , i haven't seen this: > > ---------------------------------------- > > classkit support enabled > > version 0.4 > > ----------------------------------------- > > but i have really installed classkit! > > and put it in "/usr/lib/php/modules/" > > i also set : > > extension_dir = "/usr/lib/php/modules" > > extension=classkit.so > > > > why it can not be loaded to php?? > > > > regards > > Hunter

2 1

[Serusers] Tr: Re: [Users] textops modules
by harry gaillac 24 Nov '05

24 Nov '05

Remarque : message transféré en pièce jointe. ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Users] textops modules
by harry gaillac 24 Nov '05

24 Nov '05

Hello, How can I use textops module to rewrite all ip private addresses in sip contact hf with a public ip I can't write the right regexp. if search("^Contact sip:user@private ip$)) { replace_all ("private ip", "public ip"); }; Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

2 1

[Users] communication between openser and another server!
by messa＠innsof.com 24 Nov '05

24 Nov '05

Hi all! I want to know if it is possible to configure openser to communicate with another sip server? if it's possible , how can i do it? Best regards! Serge!

2 1

[Users] Call queue
by Andréa Giordanna 24 Nov '05

24 Nov '05

Hi all, Does openser offer any support for call queues? Thanks, Giordanna --------------------------------- Yahoo! Acesso Grátis: Internet rápida e grátis. Instale o discador agora!

2 1

[Serusers] 486 Busy & 2 UAs with identical usernames
by Ramin Nikaeen 24 Nov '05

24 Nov '05

Friends, I am using 2 SJPhone UAs (B, C) on two different networks, that register with my SER server using the same (username=XXXXXXX, password). >SJPhone - B (username:XXXXXXX) / INVITE / SJPhone - A (username: YYYYYYY) \ INVITE \ >SJPhone - C (username: XXXXXXX) I dial the phone number XXXXXXX from a third SJPhone (A). When only one of the destination SJPhones is registered with SER, let's say B, then B rings. Then while B is ringing, if "ignore" button is pushed, a 486 is generated by B and sent back to SER. Then SER receives the 486 in failure_route and processes it. But when both B and C are online, both B and C ring. when "ignore" button is pressed on B, 486 is generated by B and received by SER but SER does not enter the failure_route. When ignore is pressed on the first destination, I expect SER to send a CANCEL to the second destination and receive the 486 generated by the first destination in the failure_route so that I can forward the first destination to voicemail! Can anyone share any insight as to why this is happening? Thank you for your help in advance ramin

2 1

[Serusers] Regarding Isolation of sip stack from SER
by nitin ganjsinghani 24 Nov '05

24 Nov '05

HI all *I'm doing a project on comparison of SER an OSIP* *can anybody plz assist me in telling that * *is it possible to isolate SIP stack and install only that* *if yes then how* ** *thanks* ** *Nitin*

2 1

[Serusers] Which is Better!
by Goran Donev 24 Nov '05

24 Nov '05

Which Gateways is better. I want to know which gateways are better. I am comparing Audiocodes and Mediatrix. Anyone have any opinions on either one good or bad? Thanks

7 6

[Serusers] RTP Proxy
by Kapil Dhawan 24 Nov '05

24 Nov '05

Hi List I am using rtpproxy from long now without actually knowing its proper internals. As far as i know, it is used to modify the SDP coming from Natted clients. Am i correct? Second, can i use it to develop Prepaid Application? Best Regards _________________________________________________________________ Vroom, Vrooooom! Get ready for the real speed. Drive the machine of your choice http://www.sulekha.com/classifieds/cllist.aspx?catid=280&nma=IN&c=Autos&ref…

1 0

[Users] rewrite contact field
by harry gaillac 24 Nov '05

24 Nov '05

I try to rewrite contact field in sip hf because of fix_nat_contact() fail with this diagram -----ppp0[ ser/nathelper ] eth0----private network debian 3.1 when agents send INVITE or REGISTER to ser which listen on ppp0 the ip address in contact sip hf stay a private ip address, why ? I try siproxd which rewrite it . how can i build a ser.cfg in order to contact in sip hf will be rewritte Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

2 5

[Serusers] Quintum and radius_aith
by Ashutosh kumar 24 Nov '05

24 Nov '05

Hi all, I am using Quintum tenor platform (radius + pstn gw) to send my calls outwards to pstn. I am using radius authorization. The radius server is provided by Quintum itself as a packaged billing+gw solution. The radius auth packets seem to go from my ser and they even land on the radius server, but they are not visible in the "incoming requests" monitor of the radius server or its log, although they are visible in the Ethereal on the radius machine. What I doubt is whether the Quintum radius is supports digest requests or not, because my radius packets are working very fine with freeradius, although on a different machine. Correct me or ask for clarification when and where needed. Any help will be highly appreciated. PS: - As evident, I am not having any problem in my radiusclient.conf. - The radius server is running on standard ports. -There is no firewall policy on the Quintum radius server that blocks/denies/rejects any packets from ser. Regards, Ashutosh Kumar

2 1

[Serusers] Same NAT again
by deviator 24 Nov '05

24 Nov '05

Hello! Can someone explain me how i can correctly check if 2 UA's are behind same NAT ? for example: A and B UA's have private 192.168.1.1 and 192.168.1.2, SER have a public ip A calling B For A all is clear for me - SRCIP have an A external ip and i can use it But i dont see any B external ip, RURI have a private ip part (192.168.1.2) Pls show me the way!

3 3

[Serusers] Too Many Hops
by Matteo Piazza 24 Nov '05

24 Nov '05

I have Ser and asterisk on the same machine. When i try to call with a SIP phone registred on asterisk another sip phone also registred on asterisk through SER I receve this error message: Too many hops I not understand the reason. Someone have an idea? Thank's This is my ser.cfg # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) listen=192.168.9.97 #alias=192.168.9.97 #alias=panoramix.create-net.it #listen=127.0.0.1 #port=5060 check_via=no dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" #fifo_db_url="mysql://ser:heslo@localhost/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/src/ser-0.9.4/modules/mysql/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" #loadmodule "/usr/local/lib/ser/modules/textops.so" #loadmodule "/usr/local/lib/ser/modules/nathelper.so" # added #loadmodule "/usr/local/lib/ser/modules/domain.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" #loadmodule "/usr/local/lib/ser/modules/uri_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("auth_db|uri_db|usrloc","db_url","mysql://ser:heslo@localhost/ser") #modparam("auth_db","user_column","username") 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) #modparam("usrloc","time_interval",60) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", 1) # # 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{ # lookup("aliases"); # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { log("LOG: Too many hops\n"); sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # 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(); break; }; if (method == "INVITE") { if (uri =~"^sip:0[0-9]*@*"){ log(1, "Check 1 succed Forwarding to Asterisk\n"); rewritehostport("192.168.9.97:5061"); t_relay(); break; }; }; # allow NOTIFY messages only from the Asterisk server # if (method=="NOTIFY" & src_ip==192.168.9.97) { # if (!t_relay()) { # sl_reply_error(); # break; # }; # }; # 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=~"create-net.it") { if (method=="REGISTER") { log(1,"registrazione da utente\n"); # Uncomment this if you want to use digest authentication # if (!www_authorize("create-net.it", "subscriber")) { # www_challenge("create-net.it", "0"); # break; # }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } -- ======================================= Matteo Piazza, Junior Researcher CREATE-NET Via Solteri, 38 - 38100 Trento - Italy email: matteo.piazza(a)create-net.it Tel: +39-0461-408400ext:308 www.create-net.it =======================================

4 7

Re: [Serusers] INVITE Timeout
by Pol 23 Nov '05

23 Nov '05

Hi users, please cat your ser.cfg file and grep INVITE route and then >> to the list for helping me a moment ;) I wanna know how can be a not answered INVITE, being processed by some failure route or some route. ej: UA#1 -------------->INVITE--------->SER-------------------------->UA#2 | not answered | | timeout SER ---------------------------> VOICEMAIL How are failure_routes processed, when are they execute? any online documentation for they? Thx!

2 1

[Users] new textops functions
by Daniel-Constantin Mierla 23 Nov '05

23 Nov '05

Hello, there are a set of new functions added to textops module: - insert_hf(txt) - inserts the 'txt' as header before the first header of the SIP message - insert_hf(txt, hdr) - inserts the 'txt' as header before the first 'hdr' header of the SIP message - append_hf(txt, hdr) - adds the 'txt' as header after the first 'hdr' header of the SIP message The 'txt' parameter can contain pseudo-variables which will be replaced at run time. For the existing function 'append_hf(txt)' was added support for pseudo-variables in 'txt' parameter as well. This functions adds the 'txt' as header after the last header of the SIP message. Note that if some other functions inserts headers in different positions (e.g., record_route()), so the result will be according to the order of using such functions. To give an example, if you want to add a header composed from src-ip and src port just after From header: append_hf("From-addr: $si:$sp\r\n", "From"); Updated documentation for textops module: http://openser.org/docs/modules/1.1.x/textops.html The list with pseudo-variables: http://openser.org/docs/pseudo-variables-1.1.x.html Cheers, Daniel

1 0

[Users] [Asterisk-Dev] hello
by harry gaillac 23 Nov '05

23 Nov '05

hello ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com _______________________________________________ Asterisk-Dev mailing list Asterisk-Dev(a)lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

1 0

[Serusers] Problem with serweb...
by 昂達 23 Nov '05

23 Nov '05

Hello When i open serweb with IE , it always show this error message: Function aggregate_methods() doesn't exists. Try install Classkit extension. http://pecl.php.net/package/classkit -------------------------------------------------------------------------------------------------------------------------- But i have install classkit , and i have set extension=classkit.so in php.ini and the problem still exist. why?? my system is FC4 httpd-2.0.54-10.2 + php-5.0.4-10.5 [root@15 ~]# pear list Installed packages: =================== Package Version State Archive_Tar 1.1 stable Console_Getopt 1.2 stable DB 1.7.6 stable HTML_Template_IT 1.1 stable HTTP 1.3.5 stable Mail 1.1.4 stable Net_SMTP 1.2.6 stable Net_Socket 1.0.6 stable Net_UserAgent_Detect 2.0.1 stable PEAR 1.3.5 stable XML_Parser 1.2.6 stable XML_RPC 1.4.0 stable classkit 0.4 beta

3 5

[Users] hello
by harry gaillac 23 Nov '05

23 Nov '05

hello ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Users] [Fwd: Re: [Devel] Patch - LCR Strip Option]
by Bayan William Towfiq 23 Nov '05

23 Nov '05

We've just commited a patch to allow stripping of uri before applying a prefix on a gateway-specific basis. This allows interoperability with providers that require different formats on the RURI (E164, 011+CC, 00+CC, etc.) I hope this is useful to all of you. Bayan Towfiq

1 0

[Users] open letter (2)
by harry gaillac 23 Nov '05

23 Nov '05

Dear users, This letter is addressed to the most experienced users for the ser openser and asterisk projects. Advice me and I'll stop to mail my question. How a session between two user agents behind nat could stay in the path ? Harry Kinds Regards |register | | register | agent1 asterisk| |ser/nat box | | | 200 OK | | 200 OK | agent2 One box --------------------------- | ---------------- | | | asterisk pbx | | | ---------------- | | || | | ---------- ---------- | | SER |====|NAT box |==== private network | ---------- ---------- --------------------------- ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

11 26

[Serusers] New Presence Server in SER
by Jan Janak 23 Nov '05

23 Nov '05

Folks, We have been working on SIMPLE support in SER for a while and I am happy to announce that recent SER versions contain vastly improved support for SIP-based presence (presence server). Here is (incomplete) list of features: 1) Resource list support 2) XCAP 3) XCAP-based authorization 4) presence package 5) watcherinfo package 6) Database persistence (server can be restarted without loosing existing subscriptions) The new code was extensively tested with EyeBeam user agent. There is a preeliminary version of SER Presence Handbook available at: http://www.iptel.org/ser/presence The handbook describes various modules that, together, can be used to implement almost fully-featured presence server. You can also find installation instructions there. Note that this is work in progress and both the code and documentation get updated frequently. Please report bugs and feature requests in our bug tracking system at http://bugs.sip-router.org The new presence server contains contributions from Jamey Hicks, XTen, and members of Internet2 PIC working group. Integration and development of new code was done by Vaclav Kubart <vaclav.kubart(a)iptel.org>, who is the new maintainer of presence code in SER. Special thanks go to Candace Holman, Jamey Hicks, and Ben Teitelbaum for their support. Jan.

6 9

Re: [Users] table version error
by Bogdan-Andrei Iancu 23 Nov '05

23 Nov '05

Hi Raymond, this happens because of the newly added support for regular expression based groups. Just do set modparam("group","re_table","") do disable the support. regards, Raymond Chen wrote: > Dear all, > > > > Start the CVS HEAD in debug mode gave me this error, and idea? > > > > 0(0) table_version(): No row for table re_grp found > > 0(0) ERROR:group:mod_init: Invalid table version for re_grp (use > ser_mysql.sh reinstall) > > 0(0) init_mod(): Error while initializing module group > > ERROR: error while initializing modules > > 0(0) XLOG: destroy module ... > > 0(0) pool_remove: Removing connection from the pool > > 0(0) DEBUG: tm_shutdown : start > > 0(0) DEBUG: unlink_timer_lists : emptying DELETE list > > 0(0) DEBUG: tm_shutdown : emptying hash table > > 0(0) DEBUG: tm_shutdown : releasing timers > > 0(0) DEBUG: tm_shutdown : removing semaphores > > 0(0) DEBUG: tm_shutdown : destroying tmcb lists > > 0(0) DEBUG: tm_shutdown : done > > > > Ray > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > >

1 0

[Users] Multiple IP in one box
by Fernando Rodriguez 23 Nov '05

23 Nov '05

Im trying to use openser as a Proxy for different interfaces in one box I have tried almos averithing to make it work properly but for some reason I cant. Here is my setup and I hope someone can helpme or guide me in the right way. One Linux-OpenSer box with 4 netcards 1 Has a public ip 2 Has a 192.168.23.10 ip 3 Has a 10.254.10.240 ip 4 Is not being used I want to be able to make calls from the ATAs in all the different networks, but the fact is that as it is now I cant, I want to know if I have to set up multiple media proxies or rtpproxy if I have to use a proxy dispatcher or not. This is my configuration file # ----------- global configuration parameters ------------------------ #debug=9 # 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 */ 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" fifo_db_url="mysql://openser:openserrw@localhost/openser" alias="test.net" # ------------------ module loading ---------------------------------- #loadmodule "/usr/lib/openser/modules/acc.so" loadmodule "/usr/lib/openser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/auth.so" loadmodule "/usr/lib/openser/modules/auth_db.so" loadmodule "/usr/lib/openser/modules/uri.so" loadmodule "/usr/lib/openser/modules/uri_db.so" loadmodule "/usr/lib/openser/modules/domain.so" loadmodule "/usr/lib/openser/modules/mediaproxy.so" loadmodule "/usr/lib/openser/modules/nathelper.so" loadmodule "/usr/lib/openser/modules/textops.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 2) modparam("usrloc", "timer_interval", 10) modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") # -- auth params -- modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", yes) 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) # -- tm Params -- modparam("tm", "fr_inv_timer", 45 ) modparam("tm", "fr_timer", 12 ) modparam("registrar", "nat_flag", 6) modparam("nathelper", "rtpproxy_disable", 1) #modparam("nathelper", "natping_interval", 30) #modparam("nathelper", "sipping_from", "sip:ping@sipserver.net") #modparam("mediaproxy", "ping_nated_only", 0) # Ping only clients behind NAT modparam("mediaproxy", "natping_interval", 20) modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("registrar", "nat_flag", 6) # ------------------------- request routing logic ------------------- # main routing logic route { # ----------------------------------------------------------------- # Procesos de Limpieza # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Registro # ----------------------------------------------------------------- if (method!="REGISTER") { log(1, "Area de Registro\n"); record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { log(1, "Proceso de fin de llamada\n"); end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (uri==myself) { if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); # if(dst_ip==192.168.1.1) { # log("message received on loopback interface\n"); # setflag(7); # fix_nated_register(); # force_rport(); # }; # if(dst_ip==192.168.108.253) { # log("message received on loopback interface\n"); # setflag(6); # fix_nated_register(); # force_rport(); # }; if (!www_authorize("test.net","subscriber")) { www_challenge("test.net","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; fix_nated_register(); } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); # ------------------------------------------------------------------------ # Reglas de LLamada # ------------------------------------------------------------------------ if(uri=~"(^sip:1[0-9]*@.*)" && from_uri=~"(^sip:2[0-9]*@.*)") { log(1, "Llamada de extenciones 1xxx a 2xxx\n"); use_media_proxy(); }; if(uri=~"(^sip:1[0-9]*@.*)" && from_uri=~"(^sip:3[0-9]*@.*)") { log(1, "Llamada de extenciones 1xxx a 3xxx\n"); fix_contact(); use_media_proxy(); }; if(uri=~"(^sip:1[0-9]*@.*)" && from_uri=~"(^sip:4[0-9]*@.*)") { log(1, "Llamada de extenciones 1xxx a 4xxx\n"); fix_contact(); use_media_proxy(); }; if(uri=~"(^sip:2[0-9]*@.*)" && from_uri=~"(^sip:1[0-9]*@.*)") { log(1, "Llamada de extenciones 2xxx a 1xxx\n"); fix_contact(); use_media_proxy(); }; if(uri=~"(^sip:2[0-9]*@.*)" && from_uri=~"(^sip:3[0-9]*@.*)") { log(1, "Llamada de extenciones 2xxx a 3xxx\n"); fix_contact(); use_media_proxy(); }; if(uri=~"(^sip:2[0-9]*@.*)" && from_uri=~"(^sip:4[0-9]*@.*)") { log(1, "Llamada de extenciones 2xxx a 4xxx\n"); fix_contact(); use_media_proxy(); }; if(uri=~"(^sip:3[0-9]*@.*)" && from_uri=~"(^sip:1[0-9]*@.*)") { log(1, "Llamada de extenciones 3xxx a 1xxx\n"); fix_contact(); use_media_proxy(); }; if(uri=~"(^sip:3[0-9]*@.*)" && from_uri=~"(^sip:2[0-9]*@.*)") { log(1, "Llamada de extenciones 3xxx a 2xxx\n"); fix_contact(); use_media_proxy(); }; if(uri=~"(^sip:3[0-9]*@.*)" && from_uri=~"(^sip:4[0-9]*@.*)") { log(1, "Llamada de extenciones 3xxx a 4xxx\n"); fix_contact(); use_media_proxy(); }; if(uri=~"(^sip:4[0-9]*@.*)" && from_uri=~"(^sip:1[0-9]*@.*)") { log(1, "Llamada de extenciones 4xxx a 1xxx\n"); fix_contact(); use_media_proxy(); }; if(uri=~"(^sip:4[0-9]*@.*)" && from_uri=~"(^sip:2[0-9]*@.*)") { log(1, "Llamada de extenciones 4xxx a 2xxx\n"); fix_contact(); use_media_proxy(); }; if(uri=~"(^sip:4[0-9]*@.*)" && from_uri=~"(^sip:3[0-9]*@.*)") { log(1, "Llamada de extenciones 4xxx a 3xxx\n"); fix_contact(); use_media_proxy(); }; route(1); } onreply_route[1] { #if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { # if (!search("^Content-Length:\ +0")) { log(1, "Mensaje de prueba en reply route\n"); fix_contact(); use_media_proxy(); # }; # }; } As you can see I use different starting numbers for the ATA for each of the interfaces, I tested using only one mediaproxy listening to 127.0.0.1 but I have tried a lot of configurations, all the clients register perfectly but cant seem to call one another exept the ones that are on the same interface. Im using fedora core 3 with openser 0.95 mediaproxy 1.42 Hope any one can help me or point me the way to do this of if its not viable or cant be done. Thanks a lot for your kind help Yours truly Fernando Rodriguez V. frod(a)aitelecom.net AITelecom S.A. de C.V. http://www.aitelecom.net

2 2

[Serusers] mysql necessary to add users to ser?
by Corey 23 Nov '05

23 Nov '05

Hello! I'm reading through a couple ser tutorials, and they're both suggesting to install mysql and to use the mysql.so module. But I'm confused - is mysql necessary or not? The default ser.conf seems to suggest that it's an option; but the serctl usage text shows '*' next to each action that requires a mysql pwd... including adding/deleting users. How would I add/delete users if not using mysql? Thanks! Corey

7 6

[Users] help harry...and hopefully return us back to sanity
by Iqbal 23 Nov '05

23 Nov '05

Hi This harry thread , I now look forward to in the morning, wastes alot of space, but then so does spam. I have completely lost where I was on the thread about his problem, so harry I suggest you scrap what you have posted up until now, cause it makes no sense, also post only to serusers and openser, lets ignore asterisk for now. I am going to post my understanding of what you want, all those who have a clue please add, and lets see if we can solve Harry's problem, harry if we cannot, please accept that what you are doing is far too advanced for the mortal man/women (are there any women ?) on this list, and hence you must carve a path, where no one has gone before...either that or change what you are doing. So here goes..... 1 You have ser and asterisk running. Both have a public IP address, and asterisk runs on port 5050 and ser 5060 2. You want asterisk to handle INVITE and REGISTER via SER....my question is why do you want the UA to register in SER and asterisk 3. Or do you want SER to handle the REGISTER, INVITE and then when IM/presence is needed to use asterisk 4. Can you explain what you have till now, do you have UA registering with SER, and maiing calls to each other (without asterisk)..i.e normal calls IP <---> IP, with either or both of them behind a NAT...if not do this first, ignore the rest. 5. If 4 is correct, then have you got asterisk and ser connected together, i.e talking to each other Iqbal

3 4

[Users] Re: Cesc ask for expelling Harry
by harry gaillac 23 Nov '05

23 Nov '05

Without success How can I solve this ? Harry --- Yair Hakak <yhakak(a)gmail.com> a écrit : > did you try > fix_nated_contact() as well as > fix_nated_register()? > > -yair > > > On 11/23/05, harry gaillac <gaillacharry(a)yahoo.fr> > wrote: > > something like this > > > > sip:user@private ip > > > > --- Yair Hakak <yhakak(a)gmail.com> a écrit : > > > > > what does a registered agent look like on serctl > ul > > > show? > > > > > > -yair > > > > > > On 11/23/05, harry gaillac > <gaillacharry(a)yahoo.fr> > > > wrote: > > > > > > > > > > > > --- Yair Hakak <yhakak(a)gmail.com> a écrit : > > > > > > > > > if you are using natheper and mediaproxy the > > > agent > > > > > should be > > > > > registering with the public ip, not the > private > > > one. > > > > > do a serctl ul show and let's see what a > > > registered > > > > > agent looks like. > > > > > > > > Here is a part of my RREGISTER block. > > > > How do you fix public ip in contact sip HF ( > not > > > sdp) > > > > > > > > if (nat_uac_test("19")) { > > > > setflag(7); > > > > force_rport(); > > > > fix_nated_register(); > > > > }; > > > > > > > > Harry > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > ___________________________________________________________________________ > > > > Appel audio GRATUIT partout dans le monde avec > le > > > nouveau Yahoo! Messenger > > > > Téléchargez cette version sur > > > http://fr.messenger.yahoo.com > > > > > > > > > > > > > > > > > > > > > > > > ___________________________________________________________________________ > > Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > > Téléchargez cette version sur > http://fr.messenger.yahoo.com > > > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Users] RE : Harry's list???
by harry gaillac 23 Nov '05

23 Nov '05

Dear people, Don't make me morals please I am not nutcase Harry --- Frank Fischer <frank.fischer(a)digitalnomads.ch> a écrit : > > > -----Original Message----- > > From: serusers-bounces(a)iptel.org > > [mailto:serusers-bounces@iptel.org] On Behalf Of > harry gaillac > > Sent: Wednesday, November 23, 2005 1:29 PM > > To: Olivier Taylor > > Cc: serusers(a)iptel.org; users(a)openser.org > > Subject: [Serusers] RE: RE : Harry's list??? > > > > I don't swear my english is too poor . > > However some people are quite aggressive on these > > lists. > > Such as you are Harry. > Please ask your questions kind and polite. > And do not blame others if they don't answer your > questions. No one has to. > Just try to understand, that only the software is > opensource, not the > people in this list. If they are kind to help > others, and many here are, > then you should be thankful, because they spent a > part of THEIR time for > You at no costs. > Just let me say, that in some cases, you would be by > far better of to pay > for > Support (and i know what i'm talking about). > Then you have the right to demand for a solution in > a certain time. > Unless you don't pay for it, just try to be polite > and to accept if you get > no > answer at all. That's how opensource is working. If > you may not deal with > this, the result is, as you see now, that some > think, you should be better > kicked from this list. > Addtional, think about how much time of all members > of this list your > senseless posting wasted. I think you should learn > to treat others > with a little bit more respect. Btw, that's not only > a question of this > particular mailing list. > > - Frank > > > Harry > > --- Olivier Taylor <olivier.taylor(a)gmail.com> a > écrit > > : > > > > > And if they don't as quickly as you expect, you > > > swear :( > > > Not a good idea. > > > > > > Olivier > > > > > > -----Message d'origine----- > > > De : harry gaillac > [mailto:gaillacharry@yahoo.fr] > > > Envoyé : mercredi 23 novembre 2005 13:20 > > > À : Olivier Taylor > > > Cc : serusers(a)iptel.org; users(a)openser.org > > > Objet : RE: Harry's list??? > > > > > > > > > Olivier, > > > > > > I hate "Sarkozy mode" However i guess some > > > experienced > > > people on these lists can tell me yes it's > possible > > > or > > > not ! > > > > > > That's all I ask to the lists > > > > > > Regards Harry > > > > > > --- Olivier Taylor <olivier.taylor(a)gmail.com> a > > > écrit > > > : > > > > > > > There is something buggy in this list, where > is > > > the > > > > bug reporting tool? > > > > > > > > Maybe it's time to stop that flooding now. > > > > > > > > Please, Harry, post your problems one by one > and > > > > please be patient, you must > > > > know that the 'Sarkozy mode' is not really > > > > appreciated and will ot help. > > > > > > > > Kind regards, > > > > > > > > Olivier > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > ______________________________________________________________ > > _____________ > > > > > > Appel audio GRATUIT partout dans le monde avec > le > > > nouveau Yahoo! Messenger > > > Téléchargez cette version sur > > > http://fr.messenger.yahoo.com > > > > > > > > > > > > > > > > > > > > > > > ______________________________________________________________ > > _____________ > > Appel audio GRATUIT partout dans le monde avec le > nouveau > > Yahoo! Messenger > > Téléchargez cette version sur > http://fr.messenger.yahoo.com > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)iptel.org > > http://mail.iptel.org/mailman/listinfo/serusers > > > > > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

3 2

[Users] One way voice problem with SIP client + openser + pstn gw without NAT
by Lokesh Kumar 23 Nov '05

23 Nov '05

Hello Friends, I am having some strange problem of one way voice without any NAT.I am having pstn gw on public IP, openser on public IP, and a SIP client on public IP.I am able to dial from outside phone to my SIP client. When I try to talk , the voice passes only in one direction, i.e..from SIP client to my outside phone, and not towards other side. What is the reason of this problem. Please help me.openser is having all default configuration where anyone with any name and password can be allowed to dial. I will be very thankfull to you. Regards

3 3

[Serusers] Radius accounting
by Rosa De Santis 23 Nov '05

23 Nov '05

Hello all. Please, I am trying to make Radius authorization and accounting using SER 0.9.3 and radiusclient-ng. I already set all the configuration as the ser_radius HowTo. Now, I'm making Radius authorization successfully, but not accounting, I don't get anything going out through port 1813. The Radius Server is in another machine, i'm configuring only the radius client. Is there other things to do besides the ser_radius howto? Please help... Thanks a lot Rosa.

5 14

[Serusers] How to do a large number of relay attempts?
by Tim Klein 23 Nov '05

23 Nov '05

Imagine I have 50 salespeople, and I know the URIs of all of their phones. When an INVITE comes in to SER, I want to keep trying to forward it to the salespeople, one at a time (in series, not in parallel), forever or until one of them accepts the call. (I randomize the order.) QUESTION 1: At first, I tried rewriting the Request URI in a failure route each time t_relay() returned failure, and then calling t_relay() again. But t_relay() doesn't work that way -- when used in a failure route, it requires you to append a branch, and apparently ignores any changes in the Request URI after the failure happens. OK, so in my failure route, I'll append a branch instead of rewriting the Request URI. But the default value of MAX_BRANCHES in config.h is only 12. I think this means I can never try to relay to more than 13 salespeople -- the first t_relay() call, plus 12 t_relay() calls in failure routes. Is this correct? QUESTION 2: OK, so I'll make the value of MAX_BRANCHES greater than 12. But I see in modules/tm/tm.c that MAX_BRANCHES cannot be given a value greater than 31, since each branch occupies one bit in a 32-bit bitmap. So how can I do 50 stateful relays? Or, for that matter, an indefinite number? Thank you for any help! Tim

3 4

Re: [Users] open letter (2)
by Klaus Darilion 23 Nov '05

23 Nov '05

Hi Harry! As this emails are on-topic you should cc: to the list. harry gaillac wrote: > In fact the problem is in contact sip header field > (private ip) > agent send ReGISTER to SER (outbound proxy) which one > send REGISTER to ASTERISK . > Asterisk register agent with AOR sip:users@private ip > > When agent send INVITE to an other agent ASTERISK use > > AOR sip:user@private ip but the firewall don't allow > this > Asterisk SHOULD resend INVITE to SER. > > Does SER is able to rewrite contact field in SIP HF? Which IPaddress:port do you want to have in the REGISTER's Contact: header sent from ser to Asterisk? klaus > > Regards > Thanks for your advices > > Harry > > > --- Klaus Darilion <klaus.mailinglists(a)pernau.at> a > écrit : > > >>harry gaillac wrote: >> >>>>Have you ever used SIP clients with presence and >> >>IM? >> >>>>I suggest to setup >>>>ser (without Asterisk) just to test the IM >> >>features. >> >>>>SIP based >>>>IM/presence implementations are very poor yet. >>> >>> >>>I've done it >> >>And what were your experiences? Which clients do you >>use? >> > > > Polycom IP300 > > >>>>In your picture, the NAT router is on the same PC >> >>as >> >>>>ser and asterisk. >>>>Is this correct? >>> >>>this is correct >> >>It would be a good idea to split things. This is a >>rather complicated >>setup. >> >> >>>>what scenario do you have? Are all the users >> >>behding >> >>>>the same NAT (in >>>>the same subnet) and you provide VoIP within this >>>>network (e.g. an >>>>enterprise) or do you have external users (e.g. >> >>like >> >>>>iptel or >>>>freeworlddialup)? >>> >>>in fact both >>> >>> >>> asterisk+ser >>> private net=====nathelper ======nat===private net >> >>> nat box >>> || >>> internet====== >> >>I suggest: >> >>1. Asterisk, ser and the RTP proxy 8rtpproxy or >>mediaproxy) should >>listen only on the public interface (this really >>must be a routable >>public IP address, no private). > > > SER asterisk listen on public ip > > > >>2. Setup the firewall (e.g. iptables) correctly to >>allow traffic from/to >>ser, asterisk and the RTP proxy > > > Done > > >>3. setup ser according the "getting started" >>document on onsip.org. >>AFAIK this document contains hints how to route to a >>gateway. Reuse this >>part of the config to route certain calls to the >>asterisk box. > > > Done > >>4. Try to solve things step by step: >>- REGISTER should work fine from Internet and LAN >>- Calls from Internet clients to Internet clients >>- Calls from LAN clients to LAN clients >>- Calls from LAN clients to Internet clients (and >>vice versa) >>- now try to add asterisk, e.g. calling a certain >>number will be routed >>to asterisk and starts the echo application >> >>If all the above works (DO NOT start integrating the >>asterisk as long as >>basic SIP call do not work!!!!!), you can implement >>your setup. >> >>5. Do really read every word in the "getting >>started" document, if >>things are unclear read it again. >> >>6. Do not post "how to make this setup". Ask small >>questions addressing >>particular (small) problems. >> >>7. Post to the related list. >>- do not post to developer lists >>- if you use ser, post to ser's list >>- if you use openser, post to openser's list >>- if you have an asterisk problem, ask at the >>asterisk list (e.g. you >>want to solve NAT traversal and registration with >>ser. Thus, do not ask >>this kind of questions at the asterisk list). >> >>8. always remember that this support is voluntary >> >>9. If you don't find the proper english word, look >>into the dictionary >>instead of using another word which might also have >>other meanings. >> >>10. Go and buy an english SIP book. (this will you >>help to learn the >>english terms for all the SIP stuff) >> >>11. use ngrep to watch the SIP call flow >># ngrep -t -d any port 5060 >> >> >>regards >>klaus >> > > > > > > > > ___________________________________________________________________________ > Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger > Téléchargez cette version sur http://fr.messenger.yahoo.com > >

2 1

[Users] table version error
by Raymond Chen 23 Nov '05

23 Nov '05

Dear all, Start the CVS HEAD in debug mode gave me this error, and idea? 0(0) table_version(): No row for table re_grp found 0(0) ERROR:group:mod_init: Invalid table version for re_grp (use ser_mysql.sh reinstall) 0(0) init_mod(): Error while initializing module group ERROR: error while initializing modules 0(0) XLOG: destroy module ... 0(0) pool_remove: Removing connection from the pool 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done Ray

1 0

[Users] RE: RE : Harry's list???
by harry gaillac 23 Nov '05

23 Nov '05

I don't swear my english is too poor . However some people are quite aggressive on these lists. Harry --- Olivier Taylor <olivier.taylor(a)gmail.com> a écrit : > And if they don't as quickly as you expect, you > swear :( > Not a good idea. > > Olivier > > -----Message d'origine----- > De : harry gaillac [mailto:gaillacharry@yahoo.fr] > Envoyé : mercredi 23 novembre 2005 13:20 > À : Olivier Taylor > Cc : serusers(a)iptel.org; users(a)openser.org > Objet : RE: Harry's list??? > > > Olivier, > > I hate "Sarkozy mode" However i guess some > experienced > people on these lists can tell me yes it's possible > or > not ! > > That's all I ask to the lists > > Regards Harry > > --- Olivier Taylor <olivier.taylor(a)gmail.com> a > écrit > : > > > There is something buggy in this list, where is > the > > bug reporting tool? > > > > Maybe it's time to stop that flooding now. > > > > Please, Harry, post your problems one by one and > > please be patient, you must > > know that the 'Sarkozy mode' is not really > > appreciated and will ot help. > > > > Kind regards, > > > > Olivier > > > > > > > > > > > > ___________________________________________________________________________ > > Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > Téléchargez cette version sur > http://fr.messenger.yahoo.com > > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

Re: [Users] Re: Cesc ask for expelling Harry
by Yair Hakak 23 Nov '05

23 Nov '05

i quote from your email: Asterisk: registrar with public ip port=5050 open(ser): outbound proxy with public ip port=5060 LOOKS LIKE THE SAME DAMN PORT TO ME. i tried to help you, i offered to help, but evidently you're more interested in being an ass than getting an answer to your question. here's one more person who might have helped you who you've pissed off and now won't even bother. -yair p.s. dont bother responding, you're on my blacklist. On 11/23/05, harry gaillac <gaillacharry(a)yahoo.fr> wrote: > > > > what about this is not clear? > > > > NO ONE CAN HELP YOU > > > > maybe it's because your problem is esoteric > > maybe it's because you're not explaining it well > > maybe it's because you continue to attempt to use > > idioms in english > > when it's clear that you have no understanding of > > them > > maybe it's because you frequently come across as > > threatening (as in > > "when i get an answer i'll stop posting") > > maybe it's because you cross-post > > > > it's up to you to make clear what you are looking > > for, and then to ASK > > FOR (as opposed to demanding) help. > > You're kidding I spend time to explain my problem > > > i, for one, use SER and asterisk together, SER as > > the SIP proxy and > > Asterisk as the dialplan server and PSTN gateway. I > > get SER to do the > > NAT transversal using mediaproxy and nathelper, and > > use the dialplan > > and gateway features in asterisk because of > > increased call control, > > cleaner dialplan logic, call logging, all that good > > stuff. If i > > understand you correctly, you want to use asterisk > > as a proxy (why, i > > don't know, it's not a great SIP proxy). If i can > > help, i will, but i > > have to first understand what you want to do. > > > > Thanks but I've ever done > > > for example, you claim to be running both asterisk > > and SER on the same > > box on the same port. unless i'm missing something > > with the NAT/public > > IP setup, that seems like a not such-a-good-idea. > > Not the same port > > > and stop sending diagrams. they only make it worse. > > > and, here is the golden rule, and you yourself say > > this: if you send a > > question and dont get an answer, DO NOT SEND THE > > SAME QUESTION AGAIN, > > because, surprisingly enough, you won't get an > > answer and you will > > piss people off. > > Here is an other golden rule > when you can help do it > > Regards > Harry > > > > > > > > ___________________________________________________________________________ > Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger > Téléchargez cette version sur http://fr.messenger.yahoo.com >

2 1

Re: [Users] open letter (2)
by Klaus Darilion 23 Nov '05

23 Nov '05

harry gaillac wrote: >>Have you ever used SIP clients with presence and IM? >>I suggest to setup >>ser (without Asterisk) just to test the IM features. >>SIP based >>IM/presence implementations are very poor yet. > > I've done it And what were your experiences? Which clients do you use? >>In your picture, the NAT router is on the same PC as >>ser and asterisk. >>Is this correct? > > this is correct It would be a good idea to split things. This is a rather complicated setup. >>what scenario do you have? Are all the users behding >>the same NAT (in >>the same subnet) and you provide VoIP within this >>network (e.g. an >>enterprise) or do you have external users (e.g. like >>iptel or >>freeworlddialup)? > > in fact both > > > asterisk+ser > private net=====nathelper ======nat===private net > nat box > || > internet====== I suggest: 1. Asterisk, ser and the RTP proxy 8rtpproxy or mediaproxy) should listen only on the public interface (this really must be a routable public IP address, no private). 2. Setup the firewall (e.g. iptables) correctly to allow traffic from/to ser, asterisk and the RTP proxy 3. setup ser according the "getting started" document on onsip.org. AFAIK this document contains hints how to route to a gateway. Reuse this part of the config to route certain calls to the asterisk box. 4. Try to solve things step by step: - REGISTER should work fine from Internet and LAN - Calls from Internet clients to Internet clients - Calls from LAN clients to LAN clients - Calls from LAN clients to Internet clients (and vice versa) - now try to add asterisk, e.g. calling a certain number will be routed to asterisk and starts the echo application If all the above works (DO NOT start integrating the asterisk as long as basic SIP call do not work!!!!!), you can implement your setup. 5. Do really read every word in the "getting started" document, if things are unclear read it again. 6. Do not post "how to make this setup". Ask small questions addressing particular (small) problems. 7. Post to the related list. - do not post to developer lists - if you use ser, post to ser's list - if you use openser, post to openser's list - if you have an asterisk problem, ask at the asterisk list (e.g. you want to solve NAT traversal and registration with ser. Thus, do not ask this kind of questions at the asterisk list). 8. always remember that this support is voluntary 9. If you don't find the proper english word, look into the dictionary instead of using another word which might also have other meanings. 10. Go and buy an english SIP book. (this will you help to learn the english terms for all the SIP stuff) 11. use ngrep to watch the SIP call flow # ngrep -t -d any port 5060 regards klaus

1 0

[Serusers] Custom headers in SER
by Andrey Kouprianov 23 Nov '05

23 Nov '05

Hello: I am using SER 0.9.3. Whenever I attach a custom header to REGISTER request, the response would come back without it. It looks like SER strips all unknown headers from the request, before generating response. Is there any way to keep custom headers in my responses? Andrey.

1 0

[Serusers] Network topology
by Kyriakos Mavromichalis 23 Nov '05

23 Nov '05

Hi, Im trying to figure out what the network topology should be for my SIP network. What I want to do is basically to divide my users into 3 subnets and assign a SER proxy with a public ip for each subnet. The domain should be the same for all users ie sip,otenet-telecom.com and every time a user would send a register message the dns resolution would point him to the proxy that I chose to serve the users of the subnet the user to be registered belongs to. Can this be done or will I need to define separate domains ie sip1.otenet-telecom.com for the first subnet sip2.otenet-telecom.com for the second subnet etc... -- Kyriakos Mavromichalis Otenet Telecom

1 0

[Serusers] Removing SUBSCRIBE info from SER
by Jimmy Montano 23 Nov '05

23 Nov '05

Greetings I've been trying to remove SUBSCRIBE info from SIP to stop softphone users to see one another using peer-to-peer presence Anyone knows how can this be accomplished? Thanks in Advance

2 1

[Users] redirect - timer C
by Andréa Giordanna 22 Nov '05

22 Nov '05

Hi all, How can I configure openser to send a 302 message only when timer C fires? Is it possible? Thanks, Giordanna --------------------------------- Yahoo! Acesso Grátis: Internet rápida e grátis. Instale o discador agora!

2 3

Re: [Users] open letter
by harry gaillac 22 Nov '05

22 Nov '05

In fact ser should keep nat opened of ua behind nat. Ser just need to keep location for im an presence Asterisk forward requests according to contact field to ser. --- Iqbal <iqbal(a)gigo.co.uk> a écrit : > Okay, so get ser to fix the NAT part before sending > to asterisk. Any is > ser just proxying all register commands, why not > register in ser, than > asterisk, I know you are doing IM in asterisk, and I > havent done that, Asterisk do not support IM/presence. > but I am using asterisk for features like call > pickup and transfer, they > might be different in operation but I think its best > to find out howto > let ser do all the hardwork and let asterisk only > work when it needs to. They can work together ! thanks for help harry > harry gaillac wrote: > > >not exactly ! > > > >something like this : > > > > asterisk > > | > > ser > > ua1| | ua2 > > > > > >ua1 and ua2 send registration to asterisk via ser . > > > >when ua1 invite ua2 sip INVITE is sent to ser > which > >one forward it to asterisk. > >asterisk lookup in its AORs so it bridge the call > and > >send INVITE to ua2 via ser. > > > >Harry > >--- Iqbal <iqbal(a)gigo.co.uk> a écrit : > > > > > > > >>Okay almost there :-) > >> > >>So UA ---> asterisk ---> SER ---> UA > >> > >>is that it > >> > >>harry gaillac wrote: > >> > >> > >> > >>> > >>> > >>> > >>> > >>>>okay, so ALL your users are registering to > >>>>asterisk...is that correct. > >>>> > >>>> > >>>> > >>>> > >>>Correct via ser as outbound sip proxy > >>> > >>> > >>> > >>> > >>>>If so the problem is howto accept users from > >>>> > >>>> > >>behind > >> > >> > >>>>a NAT into asterisk, > >>>>or am I confusing things further. > >>>> > >>>> > >>>> > >>>> > >>>the problem is in contact field. > >>>when user agents send register we have in sip hf > >>>Contact sip:user@privateip > >>>So asterisk store this AOR and try to contact > agent > >>>via nat box instead of SER > >>> > >>> > >>> > >>> > >>> > >>>>If the above are true, where is SER in this, or > >>>> > >>>> > >>are > >> > >> > >>>>users hitting SER > >>>>and you are sending the REGISTER from ser into > >>>>asterisk. > >>>> > >>>> > >>>> > >>>> > >>>SER is an outbound sip proxy which handle IM > >>> > >>> > >>presence > >> > >> > >>>nat > >>> > >>>Harry > >>> > >>> > >>> > >>> > >>> > >>>>>>> One box > >>>>>>> --------------------------- > >>>>>>> | ---------------- | > >>>>>>> | | asterisk pbx | | > >>>>>>> | ---------------- | > >>>>>>> | || | > >>>>>>> | ---------- ---------- > >>>>>>> | | SER |====|NAT box |==== private > >>>>>>> | ---------- ---------- > >>>>>>> |-------------------------- > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>___________________________________________________________________________ > >> > >> > >> > >>>Appel audio GRATUIT partout dans le monde avec le > >>> > >>> > >>nouveau Yahoo! Messenger > >> > >> > >>>Téléchargez cette version sur > >>> > >>> > >>http://fr.messenger.yahoo.com > >> > >> > >>>. > >>> > >>> > >>> > >>> > >>> > > > > > > > > > > > > > > > >___________________________________________________________________________ > > >Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > >Téléchargez cette version sur > http://fr.messenger.yahoo.com > > > > > >. > > > > > > > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Users] Inbound failover from third-party PSTN
by Daryl Sanders 22 Nov '05

22 Nov '05

Greeting Everyone, I hit a bit of a road block in my design. I'm using SRV for proxy failover on the customer side, but I'm wondering if third-party PSTN providers will support SRV when delivering calls to my SERs. I want to make sure if the PSTN provider cannot reach one SER they try another. I've only talked to a couple providers so far, and it seems they want to just send traffic to an IP address. Is this the norm? Is HA with heartbeat and a single IP the only solution here? Thanks! - Daryl

2 1

[Serusers] RFC3327
by harry gaillac 22 Nov '05

22 Nov '05

Hello, Does SEr support RFC3327 ? Regards harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

4 3

[Serusers] Values from a regular expression used in t_relay_to_tcp()
by Jamie Prady 22 Nov '05

22 Nov '05

Hi All, I am currently trying to set up SER to act as a proxy from a device which only sends UDP SIP to a host which only accepts TCP SIP. I have sort of got this working with the following config: t_relay_to_tcp("10.0.1.108", "5060"); The problem is, the host which only accepts TCP issues a redirect as part of the SIP handshaking, so I need to be able to dynamically select the port to relay to. I currently have the following set up for a few ports which the host seems to use, but the list of ports it could use is rather large, and I'd like to get it working properly with dynamic values e.g. the following is how I'm doing it at the minute: if (uri=~"^sip:(.+@)?(10\.0\.1\.108):1480(.*)$") { log (1, "port is 1480\n"); t_relay_to_tcp("10.0.1.108", "1480"); } else { log (1, "port is 5060\n"); t_relay_to_tcp("10.0.1.108", "5060"); } Whereas I'd like something like (with Perl style regexs): if (uri=~"^sip:(.+@)?10\.0\.1\.108:([0-9]+).*$") { log (1, "port is $2\n"); t_relay_to_tcp("10.0.1.108", "$2"); } else { log (1, "port is 5060\n"); t_relay_to_tcp("10.0.1.108", "5060"); } So I need to get the value as part of the regular expression in the URI pattern matching, then use this as a value in the t_relay_to_tcp(). I've searched the documentation but I can't seem to find an obvious way to do this. I've tried using the t_relay() method which doesn't seem to work for me. I need to use the t_relay_to_tcp() method. Thanks for your help. Jamie.

1 0

[Users] Configurable TOS
by Bogdan-Andrei Iancu 22 Nov '05

22 Nov '05

Hi, thanks to the patch from Andreas Granig, OpenSER has now support for configurable TOS (Type Of Service) via configuration script. I did some small enhancements like the possibility to use for definition both numbers and names for TOS (names are easier to remember). The name are exactly as the defines in libc. For how to use it, please refer to: http://www.openser.org/dokuwiki/doku.php?id=openser_core_cookbook#tos regards, bogdan

2 2

[Users] open letter
by harry gaillac 22 Nov '05

22 Nov '05

Hello open(ser) asterisk users Here is what i expect to do : Asterisk: registrar with public ip port=5050 open(ser): outbound proxy with public ip port=5060 Asterisk don't support IM and presence so i want to use SER because of it's a good proxy: I want user agents behind nat send registration to asterisk because of it's an ipbx :-) Look at this diagram when user agent behind nat send REGISTER to ser the contact field in sip header has a private address which one is forward to asterisk for registration. When user agent are registered in asterisk AOR is sip:user@private ip so asterisk query "sip:user@private" behind nat (not possible). How a session between two user agents behind nat could keep in the path |register | | register | agent1 asterisk| |ser/nat box | | | 200 OK | | 200 OK | agent2 One box --------------------------- | ---------------- | | | asterisk pbx | | | ---------------- | | || | | ---------- ---------- | | SER |====|NAT box |==== private network | ---------- ---------- --------------------------- Send me your questions if you don't understand what i expect to do . Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

6 13

[Users] TLS requirements and some brainstorming (long email)
by Klaus Darilion 22 Nov '05

22 Nov '05

Hi all! There are several scenarios where TLS will be used to interconnect SIP proxies. (open)ser's TLS implementation should be generic enough to handle all the useful scenarios. Thus, to better understand the requirements, first I present some examples where (open)ser+TLS will be useful. (I do not propose which of the following interconnect models are good or bad. However, openser should be capable to handle all of them, best in a mixed mode). Enterprise scenario: A company uses TLS to interconnect their SIP proxies via public Internet. The proxies import the companies selfsigned CA-cert as trusted CAs. The proxies trust other proxies as soon as their cert is validated using the root CA. This is already possible using openser 1.0.0 (= or ser+experimental TLS) Federation scenario: Some ITSPs form a federation. The federation-CA signs the certs of the ITSPs. Here, the validation is like in the enterprise scenario. (open)ser validates against the federations CA-cert. This works with openser 1.0.0 as long as the ITSP is only in one federation, or uses different egress/ingress points for each federation. If the ITSP is member of two federations and uses one egress/ingress proxy, it has to decide which certificate it should present to the peer. The originating proxy could choose the proper client certificate for example by using a table like (or having the certificate as blob directly in the DB): dst_domain certificate sip.atlanta.com /etc/openser/federationAcert.pem sip.biloxy.com /etc/openser/federationBcert.pem sip.chicago.com /etc/openser/federationAcert.pem Presenting the proper server certificate, is more difficult. The server does not know if the incoming TLS request belongs to a member of fedA, fedB or someone else. Thus, presenting the wrong certificate will lead to the clients rejecting the certificate due to failed validation. One solution would be sending the "trusted_ca_keys" (TLS extension) in Client Hello. Unfortunatelly this is not supported in openssl (and gnutls). Any workaround for this? Anyway, in this scenario it is important to have the certificate parameters (Subject, Issuer) available in the routing logic to make routing decisions based on the TLS authenticaten and adding them to the CDRs (e.g. via AVPs and extra accounting) Bilateral scenario: An ITSP has bilateral trust relationships. Each ITSP has its own CA which signs the certs of this ITSP. If another ITSP wants to trust this ISTP it only has to import the others CA-cert. This works already with openser 1.0.0, but exporting the cert parameters for extra accounting will be useful. Hosted SIP scenario: An ITSP hosts multiple SIP domains for its customers. If the server has to offer a certificate which includes the proper SIP domain, the server_name extension is needed to indicate the requested domain in the client_hello request. Then the server will present the proper certificate and domain validation (Subject domain == SIP domain) in the client will succeed. This will work fine with initial (out-of-dialog) requests as they usually will include the SIP domain in the request URI. There will be problems for responses and in-dialog requests as usually the Record-Route and Via headers only includes IP addresses. Thus, the SIP proxy either has to insert the SIP domain into Via and Record-Route, or the domain validation should only be done for in-dialog requests. This leads to the problem of domain validation. The TLS connection will be set up after all the routing logic, somewhere inside t_relay. Thus, if we want domain validation, it will be inside t_relay. Maybe we can use a certain flag to indicate if domain-validation should be done (on a per-transaction basis). This might cause problems if there is already a TLS connection to the requested destination, but without domain validation or validation against a different domain (virtual domain hosting). How to solve this? I can't propose a solution to all scenarios. But I think I showed that the certificate selection and validation should be very flexible, e.g. by choosing the proper client certificate for each transaction and different routing in the server depending on the presented client certificate and the cerfiticate signer (e.g. based on a whitelist). Further we have to take care to add certifcates and CA-certs during runtime, e.g. using a FIFO command "tls_reload". This should also drop all existing TLS connections. Having a maximum connection time after which we force re-validation will also be useful. Also (open)ser should allow to import CRL (certificate revocation lists) (shouldn't be a problem with openssl) or usage of OCSP (Online Certificate Status Protocol). Now I'm ready for some discussions :-) regards klaus

4 8

[Users] STUN still mandatory ?
by Christophe Irles 22 Nov '05

22 Nov '05

Hi, Is the use of a STUN server or any other solution of this type (another daemon) still mandatory to solve NAT problem since I have set up correctly a media proxy and the netHelper module in my configuration file ? If it's not the case which type of configuration is out of the scope if my server is in this configuration ? For example for this network configuration, is a STUN server with UA compatible mandatory ? UA1 --- [openSER+MediaProxy+FW wiht NAT] --- [ Internet ] --- [ FW/NAT ] --- UA2 If it the case, which module(s) must be used with openSER and which STUN server (open source/free) can be used ? I already tested that this network configuration is working without any STUN server UA1 --- [openSER+MediaProxy+FW wiht NAT] --- [Internet] --- [FW] --- UA2(softphone on PC with direct access on internet) Thanks, Christophe

3 2

[Serusers] loadbalancer for SER
by Ramin Nikaeen 22 Nov '05

22 Nov '05

Friends, I want use multiple instances of SER with a load balancer to ensure scalability and redundancy. Does any load balancer exist that has been proven to work well with SER? Thanks ramin

3 2

Re: [Users] open letter
by harry gaillac 22 Nov '05

22 Nov '05

> And in your special case it's really hard to > actually understand what your > problem is and what you want. This is partially due > to your english skills > (please do not take this as an insult, I'm not a > native english speaker > myself, but it's just a fact: I can't understand > most of your sentences). The > other problem (at least with the last help request > I've read from you) is > that there wasn't enough information in it to > actually understand what your > problem was and what you want. Ok I post I new mail I try my best in english Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] Conference call
by Matthieu anonymous 22 Nov '05

22 Nov '05

Hi, I just have a simple question: can i set up several-users conference calls with SER? And if yes, which softphone can I use to do that? Thank you for your help. Matthieu _________________________________________________________________ Trouvez vos fichiers en un clin dil : Windows Desktop Search http://desktop.msn.fr/

2 1

Re: [Users] open letter
by harry gaillac 22 Nov '05

22 Nov '05

--- Marc Haisenko <marc(a)darkdust.net> a écrit : > On Tuesday 22 November 2005 12:47, harry gaillac > wrote: > > Dear open(ser) users, > > > > I posted many mails in the past in order to find > some > > help to solve a problem. > > Some people can't answer however I don't think ALL > > people on these lists can not give me a way . > > > > I do hope some people will answer unless those > people > > has blacklisted me ! > > > > Regards > > Harry > > You get what you pay for. Since you paid nothing, > everything you get is a > present. So if someone answers you, be thankful. But > noone is required to > answer your support requests ! What do you mean "you paid nothing" ? I've never though people have to answer but when you can help why not doing it ! > What I've seen on the list is that most of your > support requests where > answered. So I find it quite rude and impolite of > you to complain if you > don't get an immediate answer or one that solves all > your problems ! An immediate answer ! What's a none "immediate answer" the next year may be ! I do hope you're still kidding I posted many mails without replies. I'm tired of the kind of people like you who waste its time to advice . You're not the moderator so if you can help don't pollute the list Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

4 4

[Users] dbtext maintenance
by Florian Overkamp 22 Nov '05

22 Nov '05

Hi guys, Okay, so with Daniels help we have a running config using dbtext. Are there any good cookbooks on how to administer a box using this type of setup ? Using openserctl keeps complaining about missing mysql and I apparently don't know how to add a ha1 hashed password to the database, it won't really do much with the cleartext one I have set. Best regards, Florian

2 1

[Users] open letter
by harry gaillac 22 Nov '05

22 Nov '05

Dear open(ser) users, I posted many mails in the past in order to find some help to solve a problem. Some people can't answer however I don't think ALL people on these lists can not give me a way . I do hope some people will answer unless those people has blacklisted me ! Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

4 5

[Users] how does openser work!
by messa＠innsof.com 22 Nov '05

22 Nov '05

Hi all I want to know how openser work. When i execute /usr/local/sbin/openser, how does this file use the openser.cfg file to make openser work with nathelper? I want to know how is a call process in openser from the client A to client B using nathelper/rtpproxy module. Regards! Serge

2 1

[Users] Encryption on one port (TLS) but not on another in same time ?
by Christophe Irles 22 Nov '05

22 Nov '05

Hi, According the documentation about TLS (http://openser.org/docs/tls.html) and the "listen" interface description, I understand that it would be possible to have openSER listening on a specific port for encryption and not for others. But I think it's was not possible as soon as we use "disable_tls = 0" So Is it possible (and is it working) to have in the same conf file something like this: disable_tls=0 listen=tls:1.2.3.4:5061 listen=udp:1.2.3.4:5060 Thanks, Christophe

3 2

[Users] Please help problem with AUTH with radius....using openser-1.0.9
by Lokesh Kumar 22 Nov '05

22 Nov '05

Hello Friends, I had installed openser-1.0.9, and enabled radius accounting in my openser.cfg file. But I am getting errors mentioned below. and my openser crashed without running. I am unable to find out whats the reason. No posts available relating to this matter also. Please help me......... 0(6348) shm_mem_init: success 0(0) core dump limits set to 4294967295 0(0) DEBUG: init_mod: sl_module stateless - initializing 0(0) DEBUG: register_fifo_cmd: new command (sl_stats) registered 0(0) unixsock_register_cmd: New command (sl_stats) registered 0(0) DEBUG: MD5 calculated: 329cfeaa6ded039da25ff8cbb8668bd2 0(0) DEBUG: init_mod: tm 0(0) TM - (size of cell=2832, sip_msg=768) initializing... 0(0) Call-ID initialization: '3f69f754' 0(0) DEBUG: register_fifo_cmd: new command (t_uac_dlg) registered 0(0) DEBUG: register_fifo_cmd: new command (t_uac_cancel) registered 0(0) DEBUG: register_fifo_cmd: new command (t_hash) registered 0(0) DEBUG: register_fifo_cmd: new command (t_reply) registered 0(0) unixsock_register_cmd: New command (t_uac_dlg) registered 0(0) unixsock_register_cmd: New command (t_uac_cancel) registered 0(0) unixsock_register_cmd: New command (t_hash) registered 0(0) unixsock_register_cmd: New command (t_reply) registered 0(0) DEBUG: lock_initialize: lock initialization started 0(0) DEBUG: register_fifo_cmd: new command (t_stats) registered 0(0) unixsock_register_cmd: New command (t_stats) registered 0(0) DEBUG: MD5 calculated: 533cb9e91f4b999cf76861cbb9ed54ed 0(0) DEBUG: MD5 calculated: fa997f81440371de71ab448ebdb9af56 0(0) DEBUG: init_mod: rr 0(0) rr - initializing 0(0) DEBUG: init_mod: maxfwd 0(0) Maxfwd module- initializing 0(0) DEBUG: init_mod: usrloc 0(0) usrloc - initializing 0(0) DEBUG: register_fifo_cmd: new command (ul_stats) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_rm) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_rm_contact) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_dump) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_flush) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_add) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_show_contact) registered 0(0) unixsock_register_cmd: New command (ul_stats) registered 0(0) unixsock_register_cmd: New command (ul_rm) registered 0(0) unixsock_register_cmd: New command (ul_rm_contact) registered 0(0) unixsock_register_cmd: New command (ul_dump) registered 0(0) unixsock_register_cmd: New command (ul_flush) registered 0(0) unixsock_register_cmd: New command (ul_add) registered 0(0) unixsock_register_cmd: New command (ul_show_contact) registered 0(0) DEBUG: init_mod: registrar 0(0) registrar - initializing 0(0) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/openser/modules/sl.so] 0(0) find_export: found <ul_bind_usrloc> in module usrloc [/usr/local/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_register_udomain> in module usrloc [/usr/local/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_get_all_ucontacts> in module usrloc [/usr/local/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_insert_urecord> in module usrloc [/usr/local/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_delete_urecord> in module usrloc [/usr/local/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_get_urecord> in module usrloc [/usr/local/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_lock_udomain> in module usrloc [/usr/local/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_unlock_udomain> in module usrloc [/usr/local/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_release_urecord> in module usrloc [/usr/local/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_insert_ucontact> in module usrloc [/usr/local/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_delete_ucontact> in module usrloc [/usr/local/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_get_ucontact> in module usrloc [/usr/local/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_update_ucontact> in module usrloc [/usr/local/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_register_watcher> in module usrloc [/usr/local/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_unregister_watcher> in module usrloc [/usr/local/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_register_ulcb> in module usrloc [/usr/local/lib/openser/modules/usrloc.so] 0(0) DEBUG: init_mod: textops textops - initializing 0(0) DEBUG: init_mod: auth 0(0) AUTH module - initializing 0(0) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/openser/modules/sl.so] 0(0) DEBUG: init_mod: auth_radius 0(0) auth_radius - Initializing 0(0) auth_radius: No `Cisco' vendor in Radius dictionary 0(0) find_export: found <bind_auth> in module auth [/usr/local/lib/openser/modules/auth.so] 0(0) ERROR: auth_radius: can't get code for the SIP-AVP attribute 0(0) init_mod(): Error while initializing module auth_radius ERROR: error while initializing modules 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done 0(6348) shm_mem_destroy 0(6348) destroying the shared memory lock [root@ser init.d]# Thanks you very much Lokesh

3 4

[Users] To use nathelper/rtpproxy in VOCAL system!
by messa＠innsof.com 22 Nov '05

22 Nov '05

Hi all I want to add nathelper/rtpproxy module in the VOCAL system of VOVIDA. I think openser core call the nathelper module by the openser.cfg config file. I want to use the exported nathelper's functions. What do you think about it? Regards!

1 0

[Users] outbound sip proxy
by harry gaillac 22 Nov '05

22 Nov '05

Hello, Here is my config : Asterisk as registrar server :public ip:5050 Ser as outbound proxy server :public ip 5060 I wish ser to handle the packets between Nat box (netfilter) and Asterisk However contact field in the sip HF don't change from nat box to asterisk which don't allow to keep the sessions via SER . Ser receive packets with private ip in contact field which one is forward to asterisk . How ser can handle the contact field to establish sip sessions between sip agents and asterisk ? I've been trying mangle and textops modules but i really need to be adviced. One box --------------------------- | ---------------- | | | asterisk pbx | | | ---------------- | | || | | ---------- ---------- | | SER |====|NAT box |==== private network | ---------- ---------- --------------------------- Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users