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[Serusers] Stop Radius Accounting on Status 404.
by Mohamed Omar 02 Feb '05

02 Feb '05

How can make SER flags status 404 for accounting and send a stop accouting record to the radius server? Am currently doing accounting for calls going to PSTN network and it's working fine if the call is not completed and if I get an error like status 404 then no stop accounting record is sent to the radius server. I would like to force SER to send stop accouting flag to radius server. Can I do this on SER 0.8.14 if (!lookup("location")) { setflag(1); # doesn't seem to be needed, problably because sl_send_reply # acc_rad_request("stop"); sl_send_reply("404", "Not Found"); break; Thanks. Mohamed --------------------------------- Post your free ad now! Yahoo! Canada Personals

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[Serusers] Nathelper question
by btran 02 Feb '05

02 Feb '05

Hi, I am new to SER and I am looking for how to determine whether the caller and the callee are behind NAT. I found some example code from Zeus. But I do not understand how I can distinguished whether the caller is the caller and the callee is the callee. Here is the code: /**************************************************************************************************************************/ Jan, See below. Also, always CC the list so that other people can help you. Zeus > > > > Flag 4 is set when the caller is behind NAT. > > Flag 5 is set when the callee is behind NAT. > Would you be so kind and tell me how do you test messages to > set these > flags? /* Check for UAC behind NAT This is done by nat_uac_test("7") which checks for 1. RFC 1918 address in Contact - 01 2. different src_ip and Via header - 02 3. RFC1918 in Top Via - 04 */ /*********************************************************************************************************************1. Right here - it seems to me that if both the caller and the callee are behind NAT. It will be flag as (4) and do the force_rport.*********************************************************************************************************************/ if (nat_uac_test("3")) { /* 1 + 2 = 3 */ xlog("L_NOTICE", "%rm: Client is behind NAT\n"); setflag(4); /* Flag it's a nated R-URI */ if (method=="REGISTER" || !search("^Record-Route:")) { force_rport(); if (!method=="MESSAGE") { /* MESSAGE has no Contact header, see RFC 3428 */ fix_nated_contact(); }; }; } else { xlog("L_NOTICE", "%rm: Client is not behind a NAT\n"); };... Other checking .../********************************************************************************************************************2. Then down here - it seems to me that the caller and the callee will be mark as flag 5. Then save into the location. Later on, how can I tell ifthe packet comes to SER is a caller or the callee ????. I could not understand this.********************************************************************************************************************/ if (method=="REGISTER") { if (isflagset(4)) { setflag(5); /* Turn on NAT Ping and flag it as a NAT callee */ }; if (!save("location")) { xlog("L_ERR", "%rm: Failed\n"); sl_reply_error(); }; break; }; /********************************************************************************************************************/ I am trying to make SER works with two UAs behind a same NAT. Any help would be very appreciated. Regards,Trong

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[Serusers] Is this an acc bug ?
by mr. barker 02 Feb '05

02 Feb '05

Using 8.14 compiled from source with acc enabled I found what I think is either a bug or maybe I am missing something . I am have found that if a PSTN conection terminates the call first that the BYE's are not logged to MSQL db however if the SIP Phone terminates the call it is then logged. The ser debug comes up first with the a 481 response . Call Leg /Transaction Does Not Exist . which makes sense The SIP phone when hung up does send a BYE with the proper tag ID but this is not logged to the db ? Posted my ser.cfg in January 2005 as a mini howto .. Lol . just when you think you have it done hahahah . INVITE's are logged and so are ACK's to the db. Has anyone else had this problem ? I am using a Sipura 2000 if it makes a difference . and a Cisco For PSTN termination. Thanks Peter

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[Serusers] Trying to install ser-0.8.14 from source
by info＠beeplove.com 01 Feb '05

01 Feb '05

Hello List: I am trying to install ser-0.8.14 from source. I ran make and got the lines below: Makefile.rules:77: action.d: No such file or directory Makefile.rules:77: crc.d: No such file or directory Makefile.rules:77: data_lump.d: No such file or directory Makefile.rules:77: data_lump_rpl.d: No such file or directory Makefile.rules:77: dprint.d: No such file or directory Makefile.rules:77: dset.d: No such file or directory Makefile.rules:77: error.d: No such file or directory Makefile.rules:77: fifo_server.d: No such file or directory Makefile.rules:77: flags.d: No such file or directory Makefile.rules:77: forward.d: No such file or directory Any idea, how to fix it. Thanks, -- Mohammad Khan

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[Serusers] Hello from a brand new user !
by info＠beeplove.com 01 Feb '05

01 Feb '05

Hello List: I am going to install 'ser' on my FC3 box. Wish me good luck! And get ready to get more mails from me !! Thanks, -- Mohammad Khan

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RE: [Serusers] Trusted IP
by Ricardo Martinez 01 Feb '05

01 Feb '05

Hello. I have an "if" statement for a trusted IP, for which i don't ask for Proxy-Auth. This is what i have. if (method=="INVITE" && src_ip!=my.trusted.addr.ip) { if ( !radius_proxy_authorize("")) { proxy_challenge("", "1"); break; }; }; Hope this helps. Cheers. Ricardo.- > -----Mensaje original----- > De: Doug Eubanks [mailto:admin@dougware.net] > Enviado el: Martes, 01 de Febrero de 2005 15:31 > Para: serusers(a)lists.iptel.org > Asunto: [Serusers] Trusted IP > > > Does anyone have an example cfg with a call forwarded from a > trusted ip (ie, no registration is ever sent)? > > This is what I am trying, but with no luck... > > if (src_ip==DIALINGATEAY && method=="ACK" ) { > record_route(); > use_media_proxy(); > setflag(1); > log(1, "Prepaid Call ACK Detected\n"); > rewritehostport("TERMGATEWAY:5060"); > forward(uri:host, uri:port); > break; > }; > > Thanks, > Doug > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] Trusted IP
by Doug Eubanks 01 Feb '05

01 Feb '05

Does anyone have an example cfg with a call forwarded from a trusted ip (ie, no registration is ever sent)? This is what I am trying, but with no luck... if (src_ip==DIALINGATEAY && method=="ACK" ) { record_route(); use_media_proxy(); setflag(1); log(1, "Prepaid Call ACK Detected\n"); rewritehostport("TERMGATEWAY:5060"); forward(uri:host, uri:port); break; }; Thanks, Doug

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[Serusers] send_rtpp_command error in nat configuration
by Ashling O'Driscoll 01 Feb '05

01 Feb '05

Hello, This is partially linked to my post this morning regarding having no audio between clients on different nats. I have spent the whole day trying different configurations and at this stage I think the reason its not working is coming down to the rtp errors (ERROR: send_rtpp_command etc). My problem as such is exactly the same detailed in the below thread (see below) which I found in the archives. However whilst suggestions were made regarding this problems (check that the versions are up to date and that you have sufficient permissions to rtp.sock etc)...no actual definitive solution was given. I have tried all the recommendations and none work. My deadline is looming so if someone (anybody) who has experience this problem and solved it, or just someone who has ideas, could give me pointers, I would be VERY appreciative. Kindest Regards, Aisling. Hello, I am have having difficulty getting rtpproxy and ser to work together correctly. Each time I start ser (after first starting rtpproxy) I get the following errors: textops - initializing stateless - initializing Maxfwd module- initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 42080 0(0) INFO: udp_init: SO_RCVBUF is finally 231808 0(0) INFO: udp_init: SO_RCVBUF is initially 42080 0(0) INFO: udp_init: SO_RCVBUF is finally 231808 5(2195) 4(2194) 2(2192) 1(2191) ERROR: send_rtpp_command: can't read reply from a RTP proxy ERROR: send_rtpp_command: can't read reply from a RTP proxy 13(2203) 14(2204) 15(2205) 16(2206) 17(2207) ERROR: send_rtpp_command: can't read reply from a RTP proxy 3(2193) ERROR: send_rtpp_command: can't read reply from a RTP proxy 5(2195) 4(2194) ERROR: send_rtpp_command: can't connect to RTP proxy <rest of errors removed; more of the same> Looking through the list I thought it was an incorrect rtpproxy version and downloaded and installed the stable 0.8.14 ser version and latest rtpproxy from CVS. But I still see the same errors. I can see the rtpproxy process running and rtpproxy.sock is present. Currently I have ser 0.8.14 and rtpproxy 1.21 2004/08/13 running on freebsd 4.10. Aside from these errors the calls set up fine but with no audio. I have been searching through the list and google for a couple of days but still seem to be missing something. Any thoughts on what I should try next? Regards, Allan -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

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[Serusers] SER & Asterisk - RTP Proxy Problem
by Felipe Martins 01 Feb '05

01 Feb '05

Hi Everybody, I'm trying to make SER work with ASTERISK, ser as a proxy server and SIP terminator, and asterisk for other services like voicemail and International calls like an UAC (thing that I know SER is not able to do). I've configured SER with the following forward route, to forward all "1" beggining call phones to forward to my asterisk for international calls: if (uri=~"^sip:8[0-9]{7}.*") { forward( 10.0.18.3, 5061 ); break; } When I restart SER, it gives me this errors: Jul 22 12:49:54 sipproxy /usr/local/sbin/ser[479]: ERROR: send_rtpp_command: can't connect to RTP proxy Jul 22 12:49:54 sipproxy /usr/local/sbin/ser[479]: WARNING: rtpp_test: can't get version of the RTP proxy Jul 22 12:49:54 sipproxy /usr/local/sbin/ser[479]: WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily I've also tried with the following rewritehost but it gives me the same error: if (uri=~"^sip:8[0-9]{7}.*") { rewritehost("10.0.18.3, 5061"); break; } I know that i'm missing something with RTP that i'm still working on. I've tried to look for it at google, but find nothing interesting. Do anyone know what is going on ? My ser.cfg is attached. -- Felipe Martins Linux System Administrator Tep Solution Provider Mundivox Communications Rua Lauro Muller, 116/Sala 505 RJ - Brasil - 22290-906 Tel.: 55 21 3820-8839 Fax.: 55 21 3820-8844

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[Serusers] Ser & Asterisk - RTP Proxy problem
by Felipe Martins 01 Feb '05

01 Feb '05

Sorry for sending this mail again, we've had a e-mail server problem and i'm not sure if that arrived at list cause I have no log due to the problem. Hi Everybody, I'm trying to make SER work with ASTERISK, ser as a proxy server and SIP terminator, and asterisk for other services like voicemail and International calls like an UAC (thing that I know SER is not able to do). I've configured SER with the following forward route, to forward all "1" beggining call phones to forward to my asterisk for international calls: if (uri=~"^sip:8[0-9]{7}.*") { forward( 10.0.18.3, 5061 ); break; } When I restart SER, it gives me this errors: Jul 22 12:49:54 sipproxy /usr/local/sbin/ser[479]: ERROR: send_rtpp_command: can't connect to RTP proxy Jul 22 12:49:54 sipproxy /usr/local/sbin/ser[479]: WARNING: rtpp_test: can't get version of the RTP proxy Jul 22 12:49:54 sipproxy /usr/local/sbin/ser[479]: WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily I've also tried with the following rewritehost but it gives me the same error: if (uri=~"^sip:8[0-9]{7}.*") { rewritehost("10.0.18.3, 5061"); break; } I know that i'm missing something with RTP that i'm still working on. I've tried to look for it at google, but find nothing interesting. I've trie to download and install rtpproxy from http://www.portaone.com/~sobomax/rtpproxy.tar but inside the tar there is no configure, nothing to install it but main.c and myqueue.c code files ... Do anyone know what is going on, or how to compile that ? My ser.cfg is attached. I don't know what kind of error is that 'cause all the errors like this mentioned at google where just another questions about the same error ... --------------- 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 ------------------------ # Uncomment these lines to enter debugging mode debug=9 fork=yes log_stderror=no listen=192.168.4.10 port=5060 # hostname matching an alias will satisfy the condition "uri==myself". alias="mundivox.com" alias="sipserver.com" # sip_warning - Should replies include extensive warnings? By default # yes, it is good for trouble-shooting #sip_warnings=yes # server_signature - Should locally-generated messages include server's # signature? By default yes, it is good for trouble-shooting. server_signature=yes # reply_to_via - A hint reply modules whether they should send reply # to IP advertised in Via. Turned off by default, which means that # replies are sent to IP address from which requests came. # reply_to_via=no # mhomed -- enable calculation of outbound interface; useful on # multihomed servers. # mhomed=0 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) children=4 fifo_mode=0666 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database 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/domain.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" loadmodule "/usr/local/lib/ser/modules/enum.so" loadmodule "/usr/local/lib/ser/modules/nathelper.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" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- # 0 = disable # 1 = write-through # 2 = write-back # MySQL Based User Authentication modparam("auth_db", "db_url", "sql://ser:heslo@localhost/ser") # Checking for Username Column #modparam("auth_db", "user_column", "username") # Checking for Domain Column #modparam("auth_db", "domain_column", "domain") # Checking for Password Column modparam("auth_db", "password_column", "password") # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "desc_time_order", 1) # ----- Acc Params ------ modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) # ----- auth params ----- # Uncomment if you are using auth module # Permite tambem clear text passwords no mysql modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # # ------ 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 { # ---------------------------------------------------------------------------- # Sanity Checks -- messages with 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; }; # ------------------------------------------------------------ # OPTIONS Section # # This is used by sipsak to monitor the heath of our sip proxy #------------------------------------------------------------- # if (search("^From: sip:sipsak@") && # (method=="OPTIONS") && (!uri=~"sip:.*[@]+.*")) { # options_reply(); # break; # }; # ------------------------------------------------------------ # Registration Section # ------------------------------------------------------------ # if (method=="REGISTER") { # # if (!is_from_local()) { # # sl_send_reply("403", "Unknown Domain"); # break; # }; # # if (is_user_in("Request-URI", "disabled")) { # # sl_send_reply("403", "Your evaluation period has expired"); # break; # }; # # if (!www_authorize("", "subscriber")) { # # www_challeng("", "0"); # break; # }; # # if (!check_to()) { # # sl_send_reply("401", "Unauthorized"); # break; # }; # # if (!save("location")) { # # sl_reply_error(); # }; # # break; # # }; # ----------------------------------------------------------------- # Open Relay Section # ----------------------------------------------------------------- # if (method=="INVITE") { # # if (!(is_from_local() || is_uri_host_local())) { # sl_send_reply("403", "Please register to use our service"); # break; # }; # }; # ----------------------------------------------------------------- # Accounting Section # ----------------------------------------------------------------- # if (method=="INVITE" || method=="BYE") { # setflag(1); # }; # ----------------------------------------------------------------- # Record Route Section # # we record-route all messages -- to make sure that subsequent messages # will go through our proxy; that's particularly good if upstream and # donwstream entities use different transport protocol # ----------------------------------------------------------------- if (method=="INVITE") record_route(); # ----------------------------------------------------------------- # Loose Route Section # # Grant route routing if route headers present # ----------------------------------------------------------------- if (loose_route()) { route(2); break; }; # ----------------------------------------------------------------- # Alias Routing Section # ----------------------------------------------------------------- lookup("aliases"); if (!uri==myself) { route(2); 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; }; # Rota usada para guardar Logs no CDRTool - Billing # if (method=="REGISTER" || method=="INVITE" || method=="BYE" || method=="CANCEL") { # # Salvar mensagens no myslq para o CDRTool # exec_msg("/var/www/html/serweb/sertrace.py; exit 1"); # }; # 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) # uri==myself retorna true se o nome de dominio URI for igual ao nome do # host ao qual SER esta rodando. Para configurar quais domain names o ser # aceita, deve-se configurar os ALIASES do sistema para tais nomes. if (uri==myself) { ######### Repassando chamadas comecando por 1 para o Asterisk # if (uri =~ "sip:2[0-9]*@192.168.4.4"){ # log(1, "Forwarding to Asterisk\n"); # rewritehostport("192.168.4.10:5061"); # t_relay(); # break; # } # Repassando Chamadas Internacionais para Asterisk if (uri=~"^sip:1[0-9].*") { # forward( 192.168.4.10, 5061 ); rewritehostport("192.168.4.10:5061"); break; } if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("192.168.4.10", "subscriber")) { www_challenge("192.168.4.10", "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(); }; } ------------------------- ser.cfg EOF ------------------------------- On Mon, 31 Jan 2005 16:00:18 +0100 Alexander Hoffmann <alexander.hoffmann(a)netgenius.de> wrote: > Hello Felipe, > > I don't think that the error msg you saw is related to the forwarding block > you pasted in your mail. It just tells you that SER cannot connect to SRR. > Since I can NOT see the attached ser.cfg file, I cannot give you any more > details. > > -alex > > On Monday 31 January 2005 15:13, Felipe Martins wrote: > > Hi Everybody, > > > > I'm trying to make SER work with ASTERISK, ser as a proxy server and SIP > > terminator, and asterisk for other services like voicemail and > > International calls like an UAC (thing that I know SER is not able to do). > > I've configured SER with the following forward route, to forward all "1" > > beggining call phones to forward to my asterisk for international calls: > > > > if (uri=~"^sip:8[0-9]{7}.*") { > > forward( 10.0.18.3, 5061 ); > > break; > > } > > > > > > When I restart SER, it gives me this errors: > > > > Jul 22 12:49:54 sipproxy /usr/local/sbin/ser[479]: ERROR: > > send_rtpp_command: can't connect to RTP proxy Jul 22 12:49:54 sipproxy > > /usr/local/sbin/ser[479]: WARNING: rtpp_test: can't get version of the RTP > > proxy Jul 22 12:49:54 sipproxy /usr/local/sbin/ser[479]: WARNING: > > rtpp_test: support for RTP proxyhas been disabled temporarily > > > > > > I've also tried with the following rewritehost but it gives me the same > > error: > > > > if (uri=~"^sip:8[0-9]{7}.*") { > > rewritehost("10.0.18.3, 5061"); > > break; > > } > > > > > > I know that i'm missing something with RTP that i'm still working on. > > I've tried to look for it at google, but find nothing interesting. > > Do anyone know what is going on ? My ser.cfg is attached. > -- Felipe Martins Linux System Administrator Tep Solution Provider Mundivox Communications Rua Lauro Muller, 116/Sala 505 RJ - Brasil - 22290-906 Tel.: 55 21 3820-8839 Fax.: 55 21 3820-8844 -- Felipe Martins Linux System Administrator Tep Solution Provider Mundivox Communications Rua Lauro Muller, 116/Sala 505 RJ - Brasil - 22290-906 Tel.: 55 21 3820-8839 Fax.: 55 21 3820-8844

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[Serusers] mediaproxy/rtpproxy efficiency comparison
by Darek Baca 01 Feb '05

01 Feb '05

As you know there are (at least) two options to resolve the symmetric NAT traversal with SIP/RTP: 1. nathelper module and rtpproxy server 2. mediaproxy module and dispatcher/mediaproxy server However during our test we discovered that in terms of efficiency there is a big difference between these two approaches. Using mediaproxy and rtp generator tool (comes with mediaproxy server) on 1G RAM 3000 MHZ cpu machine, server was able to handle about 200 calls at the same time (rtpgenerator rate parameter set to 100). To be able to compare mediaproxy result to rtpproxy I made some modifications inside the rtpproxy server code and used the same rtpgenerator (there is different communication interface that I modified) and the result of the same test was quit different. Using the same environment I was able to generate 900 calls, I also made additional normal call (SER/mediaproxy/proxydispatcher/rtpproxy) thru this rtprpoxy to test quality of voice. So, my question is why there is not compatibility between these servers? There would be useful to implement these small changes into the rtp proxy code (I can send my version) and take all advantages of mediaproxy load balancing and rtpproxy efficiency. Darek

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[Serusers] Your mail send to biswas@stprkl.soft.net has been received.
by Mrinmoy Biswas 01 Feb '05

01 Feb '05

Your mail send to biswas(a)stprkl.soft.net has been received. Will reply as soon as I get back. Thank you. Regards, Mrinmoy Biswas

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Re: [Serusers] no audio if clients are behind different nat
by Ashling O'Driscoll 01 Feb '05

01 Feb '05

Thanks for the reply Philip. However, before I try STUN and totally abandon nathelper and rtpproxy,I just want to see if anyone know how to solve the below problem. I would suspect since the person who posted the original thread didnt reply after Klaus suggested that he include force_rtp_proxy that he managed to make it work. What I dont understand is where the force_rtp_proxy should be included since it was already in the original config file as shown below. Any pointers appreciated. (I also tried changing fix_nated_sdp ("1") to ("3") but that didnt work) Regards, Aisling. ---- Original Message ---- From: lintenhofer(a)aon.at To: ashling.odriscoll(a)cit.ie Subject: Re: [Serusers] no audio if clients are behind different nat Date: Mon, 31 Jan 2005 18:22:58 +0100 >Ashling O'Driscoll schrieb: > >>Hi, >> >>I have the exact same problem as described in the below email which >I >>found on the serusers mailing archives. That is clients behind the >>same nat can ring each other and transmit audio but if a client >tries >>to ring a client behind another nat no audio is transmitted. >>According to the below email this is because the sdp is not >rewritten >>correctly. The below solution mentions calling force_rtpproxy but >>that function is called in the original config (see below)...Is >there >>a specific plce that it should be invoked? >> >>Thank you, >>Aisling. >> >>You have to call force_rtpproxy -> this will rewrite the IP address >>and >>port with the one of the rtpproxy. >> >>klaus >> >>Sun Yen-Rong (Dan) wrote: >> >> >> >>>Hi Klaus, >>>Firstly, thanks for your help. I used ethereal to capture the SIP >>>messages, and found out that the SDP in the forwarded INIVITE and >>> >>> >>200 OK >> >> >>>message are not rewritten correctly. As a result, the client with a >>>public IP address is sending RTP packets to the private IP address >>> >>> >>of >> >> >>>the other client who is behind a NAT. One seruser suggested me to >>> >>> >>use >> >> >>>fix_nated_sdp("3") instead of fix_nated_sdp("1"), however, it >didn't >>>solve my problem. Do I also need to add a parameter to other >>> >>> >>nathelper >> >> >>>functions as well? >>> >>>Regards, >>>Dan. >>> >>> >>>-----Original Message----- >>>From: Klaus Darilion [mailto:klaus.mailinglists at pernau.at] >>>Sent: Thursday, May 06, 2004 9:45 PM >>>To: Sun Yen-Rong (Dan) >>>Cc: serusers at iptel.org >>>Subject: Re: [Serusers] no audio problem if clients behind NATs >>> >>>use a packet sniffer (ethereal or ngrep) and watch the SIP messages >>> >>> >>at >> >> >>>the SIP proxy. Take a look at the sdp in the forwarded INVITE and >>> >>> >>200 OK >> >> >>>message and verify that the IP address and port in the SDP are >>> >>> >>rewritten >> >> >>>correctly and points to the rtpproxy. >>> >>>regards, >>>klaus >>> >>>Sun Yen-Rong (Dan) wrote: >>> >>> >>> >>> >>>>Hi all, >>>> >>>>I am using ser 0.8.12 with nathelper and rtpproxy. When I tried to >>>> >>>> >>>make >>> >>> >>> >>>>a call between two clients which are behind the same NAT, >everything >>>>work fine. However, when I try to make a call between clients >which >>>> >>>> >>>are >>> >>> >>> >>>>behind different NATs, niether client can hear each other's audio. >>>> >>>> >>my >> >> >>>>configuration file is shown in the end of this message. Can >someone >>>> >>>> >>>help >>> >>> >>> >>>>me, please? >>>> >>>>Thanks, >>>> >>>>Dan. >>>> >>>># >>>># $Id: nathelper.cfg,v 1.1.2.1 2003/11/24 14:47:18 janakj Exp $ >>>># >>>># simple quick-start config script including nathelper support >>>> >>>># This default script includes nathelper support. To make it work >>>># you will also have to install Maxim's RTP proxy. The proxy is >>>> >>>> >>>enforced >>> >>> >>> >>>># if one of the parties is behind a NAT. >>>># >>>># If you have an endpoing in the public internet which is known to >>>># support symmetric RTP (Cisco PSTN gateway or voicemail, for >>>> >>>> >>>example), >>> >>> >>> >>>># then you don't have to force RTP proxy. If you don't want to >>>> >>>> >>enforce >> >> >>>># RTP proxy for some destinations than simply use t_relay() >instead >>>> >>>> >>of >> >> >>>># route(1) >>>># >>>># Sections marked with !! Nathelper contain modifications for >>>> >>>> >>>nathelper >>> >>> >>> >>>># >>>># NOTE !! This config is EXPERIMENTAL ! >>>># >>>># ----------- global configuration parameters >>>> >>>> >>------------------------ >> >> >>>>debug=3 # debug level (cmd line: -dddddddddd) >>>>fork=yes >>>>log_stderror=no # (cmd line: -E) >>>> >>>>check_via=no # (cmd. line: -v) >>>>dns=no # (cmd. line: -r) >>>>rev_dns=no # (cmd. line: -R) >>>>port=5060 >>>>children=4 >>>>fifo="/tmp/ser_fifo" >>>> >>>># ------------------ module loading >>>> >>>> >>---------------------------------- >> >> >>>>",# >>>># $Id: nathelper.cfg,v 1.2 2003/04/15 20:35:29 jiri Exp $ >>>># >>>># example script showing use of nathelper module >>>># (incomplete for sake of brevity) >>>># >>>> >>>># ----------- global configuration parameters >>>> >>>> >>------------------------ >> >> >>>># ------------------ module loading >>>> >>>> >>---------------------------------- >> >> >>>>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/textops.so" >>>> >>>># !! Nathelper >>>>loadmodule "/usr/local/lib/ser/modules/nathelper.so" >>>> >>>># ----------------- setting module-specific parameters >>>> >>>> >>--------------- >> >> >>>># -- usrloc params -- >>>> >>>>modparam("usrloc", "db_mode", 0) >>>> >>>># -- rr params -- >>>># add value to ;lr param to make some broken UAs happy >>>>modparam("rr", "enable_full_lr", 1) >>>> >>>># !! Nathelper >>>>modparam("registrar", "nat_flag", 6) >>>>modparam("nathelper", "natping_interval", 30) # Ping interval 30 s >>>>modparam("nathelper", "ping_nated_only", 1) # Ping only clients >>>> >>>> >>>behind >>> >>> >>> >>>>NAT >>>> >>>># ------------------------- request routing logic >>>> >>>> >>------------------- >> >> >>>># main routing logic >>>> >>>>route{ >>>> >>>> # initial sanity checks -- messages with >>>> # 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; >>>> }; >>>> >>>> # !! Nathelper >>>> # Special handling for NATed clients; first, NAT test is >>>> # executed: it looks for via!=received and RFC1918 addresses >>>> # in Contact (may fail if line-folding is used); also, >>>> # the received test should, if completed, should check all >>>> # vias for rpesence of received >>>> if (nat_uac_test("3")) { >>>> # Allow RR-ed requests, as these may indicate that >>>> # a NAT-enabled proxy takes care of it; unless it is >>>> # a REGISTER >>>> >>>> if (method == "REGISTER" || ! search("^Record-Route:")) >>>> >>>> >>>{ >>> >>> >>> >>>> log("LOG: Someone trying to register from private >>>> >>>> >>>IP, >>> >>> >>> >>>>rewriting\n"); >>>> >>>> # This will work only for user agents that support >>>> >>>> >>>symmetric >>> >>> >>> >>>> # communication. We tested quite many of them and >>>> >>>> >>>majority is >>> >>> >>> >>>> # smart enough to be symmetric. In some phones it >>>> >>>> >>>takes a >>> >>> >>> >>>>configuration >>>> # option. With Cisco 7960, it is called >>>> >>>> >>>NAT_Enable=Yes, with >>> >>> >>> >>>>kphone it is >>>> # called "symmetric media" and "symmetric >>>> >>>> >>>signalling". >>> >>> >>> >>>> fix_nated_contact(); # Rewrite contact with source >>>> >>>> >>>IP of >>> >>> >>> >>>>signalling >>>> if (method == "INVITE") { >>>> fix_nated_sdp("1"); # Add direction=active to >>>> >>>> >>>SDP >>> >>> >>> >>>> }; >>>> force_rport(); # Add rport parameter to topmost Via >>>> setflag(6); # Mark as NATed >>>> }; >>>> }; >>>> >>>> # 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 >>>> if (!method=="REGISTER") record_route(); >>>> >>>> # subsequent messages withing a dialog should take the >>>> # path determined by record-routing >>>> 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 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==myself) { >>>> >>>> if (method=="REGISTER") { >>>> >>>># Uncomment this if you want to use digest authentication >>>># if (!www_authorize("iptel.org", "subscriber")) { >>>># www_challenge("iptel.org", "0"); >>>># break; >>>># }; >>>> >>>> save("location"); >>>> break; >>>> }; >>>> >>>> lookup("aliases"); >>>> if (!uri==myself) { >>>> append_hf("P-hint: outbound alias\r\n"); >>>> route(1); >>>> break; >>>> }; >>>> >>>> # native SIP destinations are handled using our USRLOC >>>> >>>> >>>DB >>> >>> >>> >>>> if (!lookup("location")) { >>>> sl_send_reply("404", "Not Found"); >>>> break; >>>> }; >>>> }; >>>> append_hf("P-hint: usrloc applied\r\n"); >>>> route(1); >>>>} >>>> >>>>route[1] >>>>{ >>>> # !! Nathelper >>>> if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" >>>> >>>> >>>&& >>> >>> >>> >>>>!search("^Route:")){ >>>> sl_send_reply("479", "We don't forward to private IP >>>> >>>> >>>addresses"); >>> >>> >>> >>>> break; >>>> }; >>>> >>>> # if client or server know to be behind a NAT, enable relay >>>> if (isflagset(6)) { >>>> force_rtp_proxy(); >>>> }; >>>> >>>> # NAT processing of replies; apply to all transactions (for >>>> >>>> >>>example, >>> >>> >>> >>>> # re-INVITEs from public to private UA are hard to identify as >>>> # NATed at the moment of request processing); look at replies >>>> t_on_reply("1"); >>>> >>>> # send it out now; use stateful forwarding as it works reliably >>>> # even for UDP2TCP >>>> if (!t_relay()) { >>>> sl_reply_error(); >>>> }; >>>>} >>>> >>>># !! Nathelper >>>>onreply_route[1] { >>>> # NATed transaction ? >>>> if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { >>>> fix_nated_contact(); >>>> force_rtp_proxy(); >>>> # otherwise, is it a transaction behind a NAT and we did not >>>> # know at time of request processing ? (RFC1918 contacts) >>>> } else if (nat_uac_test("1")) { >>>> fix_nated_contact(); >>>> }; >>>>} >>>> >>>> >> >> >> >>-------------------Legal >Disclaimer--------------------------------------- >> >>The above electronic mail transmission is confidential and intended >only for the person to whom it is addressed. Its contents may be >protected by legal and/or professional privilege. Should it be >received by you in error please contact the sender at the above >quoted email address. Any unauthorised form of reproduction of this >message is strictly prohibited. The Institute does not guarantee the >security of any information electronically transmitted and is not >liable if the information contained in this communication is not a >proper and complete record of the message as transmitted by the >sender nor for any delay in its receipt. >> >>_______________________________________________ >>Serusers mailing list >>Serusers(a)iptel.org >>http://mail.iptel.org/mailman/listinfo/serusers >> >> >> >If it is not symmetric NAT, try to ask a STUN Server outside in the >internet. You just need a client which supports STUN (and it works)! >The >clients learn about the NAT (kind of NAT and outbound address) and >change their SDP therefore. A good software choice is XTen lite. A >STUN >Server is at sip.technikum-wien.at (Ports 3478, 3479) > >regards, >Philipp > > >-------------------Legal >Disclaimer--------------------------------------- > >The above electronic mail transmission is confidential and intended >only for the person to whom it is addressed. Its contents may be >protected by legal and/or professional privilege. Should it be >received by you in error please contact the sender at the above >quoted email address. Any unauthorised form of reproduction of this >message is strictly prohibited. The Institute does not guarantee the >security of any information electronically transmitted and is not >liable if the information contained in this communication is not a >proper and complete record of the message as transmitted by the >sender nor for any delay in its receipt. > -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

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[Serusers] NAT Keep alive message in SER.
by Ricardo Martinez 01 Feb '05

01 Feb '05

Hello list, I have a question regarding to the NAT keep alive message sent by a Lynksys PAP2 device. I configured the NAT keep Alive message as NOTIFY. When this messsage arrives to SER the next configuration line handle it. if (method=="NOTIFY") { break; }; i'm not sure if this behavior is ok. Does SER must answer to the NOTIFY (keep alive messages).?? if the answer is yes, which message is the answer to a NOTIFY? is better to do the NAT keep alive message with a REGISTER? I hope that someone could help me Ricardo Martinez.-

2 1

[Serusers] 500 Server Internal Error
by ron 01 Feb '05

01 Feb '05

Hi All, Can anyone help me on this. On my ngrep I saw this 212.218.189.14:5060 -> 212.218.189.15:5060 SIP2.0 500 Server Internal Error The source, 212.218.189.14, is the gateway, the destination, 212.218.189.15, is the SIP server. What does this mean? Does it men the erro is coming from the gateway and that there is no problem on the SIP server? I don't understand the 500 error. Thank you in advanced Regards, Ron -- -- iTouch Communications, Inc.

1 0

[Serusers] SER users disable account
by Walter Willis 01 Feb '05

01 Feb '05

somebody knows like desabilitar, the user without having to erase it of the data base? exists algun field in the table to subscriber that desabilite to the user of login or something asi?

1 0

[Serusers] presence in ser
by Meena 01 Feb '05

01 Feb '05

Hi, I am using Windows Messenger 5.1. Do I have to use mysql database for presence to work, or should the clients be able to see each other even without it? Right now, I can sign in, but cannot see my contacts even though they are online. Thanks, Meena.

3 3

[Serusers] Codec configuration
by Jain, Hitesh 01 Feb '05

01 Feb '05

Hi all, If I need to shut off the Media Attribute: rtpmap:3 GSM/8000 in SER, which file do I need to make changes in? Also does anyone have any idea what rtpmap:101 telephone-event/8000 is? Thanks, Hitesh.

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Re: AW: [Serusers] Propem with the SER Web interface
by Karel Kozlik 31 Jan '05

31 Jan '05

> Thank you first for your answer, but where can I find the error_reporting > variable? oh, sorry this variable is in php.ini > Second, in spite of changing the access permission of the ser_fifo for > writing and reading php scripts, php scripts could not open ser_fifo for > writing and thus got on the web browser (user_interface/my_account.php...) > the following error "sorry -- cannot open write fifo". > How can I solve this problem? > Best regards, This is a frequently asked question. The solution is in ser.cfg - variable mod_fifo - or something like this I don't know exact name now. Or search list archive, there are a lot of questions and answers to this. Karel > > -----Ursprüngliche Nachricht----- > Von: Karel Kozlik [mailto:kozlik@kufr.cz] > Gesendet: Mittwoch, 22. Dezember 2004 13:07 > An: Adel Al-Hezmi > Cc: serusers(a)lists.iptel.org > Betreff: Re: [Serusers] Propem with the SER Web interface > > Hello, > this is a problem of used library phplib (not errors, only notices). > Will be fixed in next version of serweb. For now you can disable this > messages by set: > > error_reporting = E_ALL & ~E_NOTICE > > Karel > > Adel Al-Hezmi wrote: > > >>Hello all, >> >> >> >>Last week I sent a question about my problem with ser_fifo and ser web >>interface, that the php script could not open ser_fifo for writing, in >>spite of changing access permission of ser_fifo allowing php scripts for >>read and write. I parsed the error_log file of http server and found >>that the oohforms.inc in phplib generates a lot of errors, such as >>undefined index or property. Here is a part of the error_log file: >> >>[ >> >>client 10.10.11.36] PHP Notice: Undefined offset: 9 in >>/var/www/html/ser/html/page.php on line 219, referer: >> > > http://10.10.11.36/ser/html/user_interface/index.php?phplib_Session=783a0f6d > 9ac75df3a91a5bc0b1b7c3b0 > >>[client 10.10.11.36] PHP Notice: Undefined offset: 10 in >>/var/www/html/ser/html/page.php on line 223, referer: >> > > http://10.10.11.36/ser/html/user_interface/index.php?phplib_Session=783a0f6d > 9ac75df3a91a5bc0b1b7c3b0 > >>[client 10.10.11.36] PHP Notice: Undefined variable: message in >>/var/www/html/ser/html/user_interface/my_account.php on line 356, >>referer: >> > > http://10.10.11.36/ser/html/user_interface/index.php?phplib_Session=783a0f6d > 9ac75df3a91a5bc0b1b7c3b0 > >>[client 10.10.11.36] PHP Notice: Undefined index: frozen in >>/var/www/html/ser/phplib/oohforms.inc on line 284, referer: >> > > http://10.10.11.36/ser/html/user_interface/index.php?phplib_Session=783a0f6d > 9ac75df3a91a5bc0b1b7c3b0 > >>[client 10.10.11.36] PHP Notice: Undefined index: frozen in >>/var/www/html/ser/phplib/oohforms.inc on line 284, referer: >> > > http://10.10.11.36/ser/html/user_interface/index.php?phplib_Session=783a0f6d > 9ac75df3a91a5bc0b1b7c3b0 > >> >> >>I hope some can help me J >> >> >> >>Best regards Adel Al-Hezmi >> >> >>------------------------------------------------------------------------ >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers > >

2 1

Re:Re: [Serusers] control SER with D.J. Bernstein's daemontools
by u2 31 Jan '05

31 Jan '05

Hi, thanks for your hint. Could you please let me know what you mean with SER multihomed will not work anymore without forking. Does multihomed mean one SER is listening to more than one IP addresses? This is not the case in my scenario. Does forking only influence the multihomed "mode" or does it also affect the SER performance or call processing behavior (e.g. parallel call attempts are being serialized without forking)?? Regards, Urs Uschmann You can use fork=no in ser.cfg and it will allow supervise to control it, the only caveat is that SER can no longer be multihomed when it doesn't fork. Also since SER only listens on the one interface you need to make sure it's listening on the correct interface by passing the -l flag in your run script in the /service directory. -Evan On Thu, 2005-01-27 at 18:49 +0100, Jan Janak wrote: > We have been using tool called monit: > > http://www.tildeslash.com/monit/ > > Jan. > > On 21-01 15:36, u2 wrote: > > <http://cr.yp.to/djb.html>Hello, > > > > did somebody success to control a SER proxy with D.J. Bernstein's > > daemontools? > > As soon as SER is run, it puts itself to the shell background. As far as > > I found out, this behavior prevents the daemontools from controlling SER. > > What did anybody already experiment with this toolkit? What do you use > > to control you SER processes (auto-respawn, logging, etc.)? > > > > It would be also great to use Bernstein's multilog to collect SER logs > > in a comfortable manner (with per instance log, log rotation, > > timestamps, etc). > > What do you use here? Any recommendation?

2 1

Re: [Serusers] no audio if clients are behind different nat
by Alexander Philipp Lintenhofer 31 Jan '05

31 Jan '05

Ashling O'Driscoll schrieb: > Hi, > > I have the exact same problem as described in the below email which I > found on the serusers mailing archives. That is clients behind the > same nat can ring each other and transmit audio but if a client tries > to ring a client behind another nat no audio is transmitted. > According to the below email this is because the sdp is not rewritten > correctly. The below solution mentions calling force_rtpproxy but > that function is called in the original config (see below)...Is there > a specific plce that it should be invoked? > > Thank you, > Aisling. > > You have to call force_rtpproxy -> this will rewrite the IP address > and port with the one of the rtpproxy. > > klaus > > Sun Yen-Rong (Dan) wrote: > > > >> Hi Klaus, >> Firstly, thanks for your help. I used ethereal to capture the SIP >> messages, and found out that the SDP in the forwarded INIVITE and >> > > 200 OK > > >> message are not rewritten correctly. As a result, the client with a >> public IP address is sending RTP packets to the private IP address >> > > of > > >> the other client who is behind a NAT. One seruser suggested me to >> > > use > > >> fix_nated_sdp("3") instead of fix_nated_sdp("1"), however, it didn't >> solve my problem. Do I also need to add a parameter to other >> > > nathelper > > >> functions as well? >> >> Regards, >> Dan. >> >> >> -----Original Message----- >> From: Klaus Darilion [mailto:klaus.mailinglists at pernau.at] Sent: >> Thursday, May 06, 2004 9:45 PM >> To: Sun Yen-Rong (Dan) >> Cc: serusers at iptel.org >> Subject: Re: [Serusers] no audio problem if clients behind NATs >> >> use a packet sniffer (ethereal or ngrep) and watch the SIP messages >> > > at > >> the SIP proxy. Take a look at the sdp in the forwarded INVITE and >> > > 200 OK > > >> message and verify that the IP address and port in the SDP are >> > > rewritten > > >> correctly and points to the rtpproxy. >> >> regards, >> klaus >> >> Sun Yen-Rong (Dan) wrote: >> >> >> >> >>> Hi all, >>> >>> I am using ser 0.8.12 with nathelper and rtpproxy. When I tried to >>> >> >> make >> >> >> >>> a call between two clients which are behind the same NAT, everything >>> work fine. However, when I try to make a call between clients which >>> >> >> are >> >> >> >>> behind different NATs, niether client can hear each other's audio. >>> >> > my > > >>> configuration file is shown in the end of this message. Can someone >>> >> >> help >> >> >> >>> me, please? >>> >>> Thanks, >>> >>> Dan. >>> >>> # >>> # $Id: nathelper.cfg,v 1.1.2.1 2003/11/24 14:47:18 janakj Exp $ >>> # >>> # simple quick-start config script including nathelper support >>> >>> # This default script includes nathelper support. To make it work >>> # you will also have to install Maxim's RTP proxy. The proxy is >>> >> >> enforced >> >> >> >>> # if one of the parties is behind a NAT. >>> # >>> # If you have an endpoing in the public internet which is known to >>> # support symmetric RTP (Cisco PSTN gateway or voicemail, for >>> >> >> example), >> >> >> >>> # then you don't have to force RTP proxy. If you don't want to >>> >> > enforce > > >>> # RTP proxy for some destinations than simply use t_relay() instead >>> >> > of > > >>> # route(1) >>> # >>> # Sections marked with !! Nathelper contain modifications for >>> >> >> nathelper >> >> >> >>> # >>> # NOTE !! This config is EXPERIMENTAL ! >>> # >>> # ----------- global configuration parameters >>> >> > ------------------------ > > >>> debug=3 # debug level (cmd line: -dddddddddd) >>> fork=yes >>> log_stderror=no # (cmd line: -E) >>> >>> check_via=no # (cmd. line: -v) >>> dns=no # (cmd. line: -r) >>> rev_dns=no # (cmd. line: -R) >>> port=5060 >>> children=4 >>> fifo="/tmp/ser_fifo" >>> >>> # ------------------ module loading >>> >> > ---------------------------------- > > >>> ",# >>> # $Id: nathelper.cfg,v 1.2 2003/04/15 20:35:29 jiri Exp $ >>> # >>> # example script showing use of nathelper module # (incomplete for >>> sake of brevity) >>> # >>> >>> # ----------- global configuration parameters >>> >> > ------------------------ > > >>> # ------------------ module loading >>> >> > ---------------------------------- > > >>> 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/textops.so" >>> >>> # !! Nathelper >>> loadmodule "/usr/local/lib/ser/modules/nathelper.so" >>> >>> # ----------------- setting module-specific parameters >>> >> > --------------- > > >>> # -- usrloc params -- >>> >>> modparam("usrloc", "db_mode", 0) >>> >>> # -- rr params -- >>> # add value to ;lr param to make some broken UAs happy >>> modparam("rr", "enable_full_lr", 1) >>> >>> # !! Nathelper >>> modparam("registrar", "nat_flag", 6) >>> modparam("nathelper", "natping_interval", 30) # Ping interval 30 s >>> modparam("nathelper", "ping_nated_only", 1) # Ping only clients >>> >> >> behind >> >> >> >>> NAT >>> >>> # ------------------------- request routing logic >>> >> > ------------------- > > >>> # main routing logic >>> >>> route{ >>> >>> # initial sanity checks -- messages with >>> # 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; >>> }; >>> >>> # !! Nathelper >>> # Special handling for NATed clients; first, NAT test is >>> # executed: it looks for via!=received and RFC1918 addresses >>> # in Contact (may fail if line-folding is used); also, >>> # the received test should, if completed, should check all >>> # vias for rpesence of received >>> if (nat_uac_test("3")) { >>> # Allow RR-ed requests, as these may indicate that >>> # a NAT-enabled proxy takes care of it; unless it is >>> # a REGISTER >>> >>> if (method == "REGISTER" || ! search("^Record-Route:")) >>> >> >> { >> >> >> >>> log("LOG: Someone trying to register from private >>> >> >> IP, >> >> >> >>> rewriting\n"); >>> >>> # This will work only for user agents that support >>> >> >> symmetric >> >> >> >>> # communication. We tested quite many of them and >>> >> >> majority is >> >> >> >>> # smart enough to be symmetric. In some phones it >>> >> >> takes a >> >> >> >>> configuration >>> # option. With Cisco 7960, it is called >>> >> >> NAT_Enable=Yes, with >> >> >> >>> kphone it is >>> # called "symmetric media" and "symmetric >>> >> >> signalling". >> >> >> >>> fix_nated_contact(); # Rewrite contact with source >>> >> >> IP of >> >> >> >>> signalling >>> if (method == "INVITE") { >>> fix_nated_sdp("1"); # Add direction=active to >>> >> >> SDP >> >> >> >>> }; >>> force_rport(); # Add rport parameter to topmost Via >>> setflag(6); # Mark as NATed >>> }; >>> }; >>> >>> # 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 >>> if (!method=="REGISTER") record_route(); >>> >>> # subsequent messages withing a dialog should take the >>> # path determined by record-routing >>> 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 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==myself) { >>> >>> if (method=="REGISTER") { >>> >>> # Uncomment this if you want to use digest authentication >>> # if (!www_authorize("iptel.org", "subscriber")) { >>> # www_challenge("iptel.org", "0"); >>> # break; >>> # }; >>> >>> save("location"); >>> break; >>> }; >>> >>> lookup("aliases"); >>> if (!uri==myself) { >>> append_hf("P-hint: outbound alias\r\n"); >>> route(1); >>> break; >>> }; >>> >>> # native SIP destinations are handled using our USRLOC >>> >> >> DB >> >> >> >>> if (!lookup("location")) { >>> sl_send_reply("404", "Not Found"); >>> break; >>> }; >>> }; >>> append_hf("P-hint: usrloc applied\r\n"); route(1); >>> } >>> >>> route[1] { >>> # !! Nathelper >>> if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" >>> >> >> && >> >> >> >>> !search("^Route:")){ >>> sl_send_reply("479", "We don't forward to private IP >>> >> >> addresses"); >> >> >> >>> break; >>> }; >>> >>> # if client or server know to be behind a NAT, enable relay >>> if (isflagset(6)) { >>> force_rtp_proxy(); >>> }; >>> >>> # NAT processing of replies; apply to all transactions (for >>> >> >> example, >> >> >> >>> # re-INVITEs from public to private UA are hard to identify as >>> # NATed at the moment of request processing); look at replies >>> t_on_reply("1"); >>> >>> # send it out now; use stateful forwarding as it works reliably >>> # even for UDP2TCP >>> if (!t_relay()) { >>> sl_reply_error(); >>> }; >>> } >>> >>> # !! Nathelper >>> onreply_route[1] { >>> # NATed transaction ? >>> if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { >>> fix_nated_contact(); >>> force_rtp_proxy(); >>> # otherwise, is it a transaction behind a NAT and we did not >>> # know at time of request processing ? (RFC1918 contacts) >>> } else if (nat_uac_test("1")) { >>> fix_nated_contact(); >>> }; >>> } >>> >> > > > > -------------------Legal > Disclaimer--------------------------------------- > > The above electronic mail transmission is confidential and intended > only for the person to whom it is addressed. Its contents may be > protected by legal and/or professional privilege. Should it be > received by you in error please contact the sender at the above quoted > email address. Any unauthorised form of reproduction of this message > is strictly prohibited. The Institute does not guarantee the security > of any information electronically transmitted and is not liable if the > information contained in this communication is not a proper and > complete record of the message as transmitted by the sender nor for > any delay in its receipt. > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > If it is not symmetric NAT, try to ask a STUN Server outside in the internet. You just need a client which supports STUN (and it works)! The clients learn about the NAT (kind of NAT and outbound address) and change their SDP therefore. A good software choice is XTen lite. A STUN Server is at sip.technikum-wien.at (Ports 3478, 3479) regards, Philipp

1 0

[Serusers] no audio if clients are behind different nat
by Ashling O'Driscoll 31 Jan '05

31 Jan '05

Hi, I have the exact same problem as described in the below email which I found on the serusers mailing archives. That is clients behind the same nat can ring each other and transmit audio but if a client tries to ring a client behind another nat no audio is transmitted. According to the below email this is because the sdp is not rewritten correctly. The below solution mentions calling force_rtpproxy but that function is called in the original config (see below)...Is there a specific plce that it should be invoked? Thank you, Aisling. You have to call force_rtpproxy -> this will rewrite the IP address and port with the one of the rtpproxy. klaus Sun Yen-Rong (Dan) wrote: > Hi Klaus, > Firstly, thanks for your help. I used ethereal to capture the SIP > messages, and found out that the SDP in the forwarded INIVITE and 200 OK > message are not rewritten correctly. As a result, the client with a > public IP address is sending RTP packets to the private IP address of > the other client who is behind a NAT. One seruser suggested me to use > fix_nated_sdp("3") instead of fix_nated_sdp("1"), however, it didn't > solve my problem. Do I also need to add a parameter to other nathelper > functions as well? > > Regards, > Dan. > > > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists at pernau.at] > Sent: Thursday, May 06, 2004 9:45 PM > To: Sun Yen-Rong (Dan) > Cc: serusers at iptel.org > Subject: Re: [Serusers] no audio problem if clients behind NATs > > use a packet sniffer (ethereal or ngrep) and watch the SIP messages at > the SIP proxy. Take a look at the sdp in the forwarded INVITE and 200 OK > > message and verify that the IP address and port in the SDP are rewritten > > correctly and points to the rtpproxy. > > regards, > klaus > > Sun Yen-Rong (Dan) wrote: > > >>Hi all, >> >>I am using ser 0.8.12 with nathelper and rtpproxy. When I tried to > > make > >>a call between two clients which are behind the same NAT, everything >>work fine. However, when I try to make a call between clients which > > are > >>behind different NATs, niether client can hear each other's audio. my >>configuration file is shown in the end of this message. Can someone > > help > >>me, please? >> >>Thanks, >> >>Dan. >> >># >># $Id: nathelper.cfg,v 1.1.2.1 2003/11/24 14:47:18 janakj Exp $ >># >># simple quick-start config script including nathelper support >> >># This default script includes nathelper support. To make it work >># you will also have to install Maxim's RTP proxy. The proxy is > > enforced > >># if one of the parties is behind a NAT. >># >># If you have an endpoing in the public internet which is known to >># support symmetric RTP (Cisco PSTN gateway or voicemail, for > > example), > >># then you don't have to force RTP proxy. If you don't want to enforce >># RTP proxy for some destinations than simply use t_relay() instead of >># route(1) >># >># Sections marked with !! Nathelper contain modifications for > > nathelper > >># >># NOTE !! This config is EXPERIMENTAL ! >># >># ----------- global configuration parameters ------------------------ >> >>debug=3 # debug level (cmd line: -dddddddddd) >>fork=yes >>log_stderror=no # (cmd line: -E) >> >>check_via=no # (cmd. line: -v) >>dns=no # (cmd. line: -r) >>rev_dns=no # (cmd. line: -R) >>port=5060 >>children=4 >>fifo="/tmp/ser_fifo" >> >># ------------------ module loading ---------------------------------- >>",# >># $Id: nathelper.cfg,v 1.2 2003/04/15 20:35:29 jiri Exp $ >># >># example script showing use of nathelper module >># (incomplete for sake of brevity) >># >> >># ----------- global configuration parameters ------------------------ >> >># ------------------ module loading ---------------------------------- >>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/textops.so" >> >># !! Nathelper >>loadmodule "/usr/local/lib/ser/modules/nathelper.so" >> >># ----------------- setting module-specific parameters --------------- >> >># -- usrloc params -- >> >>modparam("usrloc", "db_mode", 0) >> >># -- rr params -- >># add value to ;lr param to make some broken UAs happy >>modparam("rr", "enable_full_lr", 1) >> >># !! Nathelper >>modparam("registrar", "nat_flag", 6) >>modparam("nathelper", "natping_interval", 30) # Ping interval 30 s >>modparam("nathelper", "ping_nated_only", 1) # Ping only clients > > behind > >>NAT >> >># ------------------------- request routing logic ------------------- >> >># main routing logic >> >>route{ >> >> # initial sanity checks -- messages with >> # 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; >> }; >> >> # !! Nathelper >> # Special handling for NATed clients; first, NAT test is >> # executed: it looks for via!=received and RFC1918 addresses >> # in Contact (may fail if line-folding is used); also, >> # the received test should, if completed, should check all >> # vias for rpesence of received >> if (nat_uac_test("3")) { >> # Allow RR-ed requests, as these may indicate that >> # a NAT-enabled proxy takes care of it; unless it is >> # a REGISTER >> >> if (method == "REGISTER" || ! search("^Record-Route:")) > > { > >> log("LOG: Someone trying to register from private > > IP, > >>rewriting\n"); >> >> # This will work only for user agents that support > > symmetric > >> # communication. We tested quite many of them and > > majority is > >> # smart enough to be symmetric. In some phones it > > takes a > >>configuration >> # option. With Cisco 7960, it is called > > NAT_Enable=Yes, with > >>kphone it is >> # called "symmetric media" and "symmetric > > signalling". > >> fix_nated_contact(); # Rewrite contact with source > > IP of > >>signalling >> if (method == "INVITE") { >> fix_nated_sdp("1"); # Add direction=active to > > SDP > >> }; >> force_rport(); # Add rport parameter to topmost Via >> setflag(6); # Mark as NATed >> }; >> }; >> >> # 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 >> if (!method=="REGISTER") record_route(); >> >> # subsequent messages withing a dialog should take the >> # path determined by record-routing >> 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 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==myself) { >> >> if (method=="REGISTER") { >> >># Uncomment this if you want to use digest authentication >># if (!www_authorize("iptel.org", "subscriber")) { >># www_challenge("iptel.org", "0"); >># break; >># }; >> >> save("location"); >> break; >> }; >> >> lookup("aliases"); >> if (!uri==myself) { >> append_hf("P-hint: outbound alias\r\n"); >> route(1); >> break; >> }; >> >> # native SIP destinations are handled using our USRLOC > > DB > >> if (!lookup("location")) { >> sl_send_reply("404", "Not Found"); >> break; >> }; >> }; >> append_hf("P-hint: usrloc applied\r\n"); >> route(1); >>} >> >>route[1] >>{ >> # !! Nathelper >> if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" > > && > >>!search("^Route:")){ >> sl_send_reply("479", "We don't forward to private IP > > addresses"); > >> break; >> }; >> >> # if client or server know to be behind a NAT, enable relay >> if (isflagset(6)) { >> force_rtp_proxy(); >> }; >> >> # NAT processing of replies; apply to all transactions (for > > example, > >> # re-INVITEs from public to private UA are hard to identify as >> # NATed at the moment of request processing); look at replies >> t_on_reply("1"); >> >> # send it out now; use stateful forwarding as it works reliably >> # even for UDP2TCP >> if (!t_relay()) { >> sl_reply_error(); >> }; >>} >> >># !! Nathelper >>onreply_route[1] { >> # NATed transaction ? >> if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { >> fix_nated_contact(); >> force_rtp_proxy(); >> # otherwise, is it a transaction behind a NAT and we did not >> # know at time of request processing ? (RFC1918 contacts) >> } else if (nat_uac_test("1")) { >> fix_nated_contact(); >> }; >>} -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

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[Serusers] DID in France
by Graziano Poretti 31 Jan '05

31 Jan '05

hi to all i hope that this message is not OT here .. i'm looking for DID in France ... is there someone here that provides this service? (well if not it's great also do have suggestions about where to find it) thnx a lot for ur support Graziano

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[Serusers] Serweb does not auto increment numerical alias
by Holden Hao 31 Jan '05

31 Jan '05

I have installed ser_0.8.14_i386.deb on Debian Testing (Sarge). I have also installed from CVS serweb rel_0_8_12. I found solutions to some of the problems I encountered by searching the archives but I just could not find a solution for my constant numerical alias problem. Serweb is supposed to automatically increment numerical aliases for new users but the users that i tried adding, both have the same numerical aliases (first_alias_number). What coud be the problem? Thanks in advance. Holden Diagnostics: ---------------------------------------------------- test# serctl alias show 82000 200 OK <sip:johnny@test.net>;q=1.00;expires=567636634 <sip:kyne@test.net>;q=1.00;expires=567639620 ---------------------------------------------------- test# serctl alias show johnny 404 Username johnny in table aliases not found ----------------------------------------------------- Below are my configs: ser.cfg -------------------------------------------------- fifo="/tmp/ser_fifo" fifo_mode=438 loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) route{ 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; }; record_route(); if (loose_route()) { t_relay(); break; }; if (uri==myself) { lookup("aliases"); if (method=="REGISTER") { if (!www_authorize("test.net", "subscriber")) { www_challenge("test.net", "0"); break; }; save("location"); break; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; if (!t_relay()) { sl_reply_error(); }; } Portions of config.php ---------------------------------------------- $this->db_host="localhost"; //database host $this->db_name="ser"; //database name $this->db_user="ser"; //database conection user $this->db_pass="heslo"; //database conection password $this->root_path="/webser/html/"; $this->root_uri="http://".$_SERVER['SERVER_NAME']; $this->zonetab_file = "/usr/share/zoneinfo/zone.tab"; $this->mail_header_from="registrar(a)test.net"; $this->web_contact="sip:daemon@192.168.2.16"; $this->voice_silo_dir = '/var/spool/voicemail/'; $this->greetings_spool_dir = '/var/greetings/'; $this->fifo_server="/tmp/ser_fifo"; $this->prolog="/webser/html/prolog.html"; $this->separator="/webser/html/separator.html"; $this->epilog="/webser/html/epilog.html"; $this->title="iptel.org, the IP Telephony Site"; $this->realm=$this->domainname=$this->default_domain="test.net"; $this->first_alias_number=82000; $this->infomail = "info(a)test.net"; $this->regmail = "registrar(a)test.net";

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[Serusers] Serweb User tabs not showing up
by Mr B 31 Jan '05

31 Jan '05

I have installed the latest SerWeb from the cvs directory . Also am using 8.14 scr release . with 1.62 ser_mysql.sh to create the tables I am able to login in as admin and see the admin tabs but when I long in as a user I see no tabs at all. I am missing something in the serweb config's or . ? Thanks Peter

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