sr-users February 2005

sr-users@lists.kamailio.org

215 participants
467 discussions

[Serusers] behind NAT, endpoint to ednpoint rtp flow POSSIBLE???

by Syed Roquibul Ahsan

Dear all, I have a simple query. I want to configure SER in such a way that, when endpoints are behind different NAT or same NAT, RTP flow will be from endpoint to endpoint. Is there any way to configure SER in such a way? Now I am using RTP PROXY for rtp flow. But I need endpoint to endpoint at least NAT DEVICE to NATDEVICE rtp flow. Thank you in advance. Rokib

19 years, 9 months

[Serusers] ser 0.8.14-4 and sems

by ron

Hi List, I'm tryin to configure ser with sems, on a RedHat version 9. I'm receiving this errors: Feb 9 01:36:00 sip ser[4781]: ERROR: fifo_server: command DB is not available Feb 9 01:36:00 sip ser[4781]: ERROR: fifo_server:n command must begin with :: email_address Feb 9 01:36:00 sip ser[4781]: ERROR: fifo_server: command must begin with :: subscriber Feb 9 01:36:00 sip ser[4781]: ERROR: fifo_server: command must have at least 3 chars Feb 9 01:36:00 sip Sems[4816]: Error: (AmSession.cpp)(startSession)(468): 404 voicemail: no email address for user <1002> Feb 9 01:36:00 sip ser[4781]: ERROR: fifo_server: command t_reply is not available Feb 9 01:36:00 sip ser[4781]: ERROR: fifo_server: command must begin with :: voicemail: no email address for user <1002> Feb 9 01:36:00 sip ser[4781]: ERROR: fifo_server: command must begin with :: 000012D07BAA2BC1 Feb 9 01:36:00 sip ser[4781]: ERROR: fifo_server: command must have at least 3 chars Feb 9 01:36:00 sip Sems[4816]: Error: (AmRequest.cpp)(reply)(335): AmRequestUAS::reply: 500 command 't_reply' not available Can anyone help me with this, thank you in advanced. I've attached my ser.cfg and sems.conf below. ----- start of 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) /* Uncomment these lines to enter debugging mode debug=7 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 listen=192.168.0.2 alias="sip.nhadzter.com" alias="nhadzter.com" alias="192.168.0.2" 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/vm.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 -- #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) # -- 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) # ----------------- voicemail parameters --------------- #modparam("voicemail", "db_url","/home/rco/cvs/ser/sip_router/modules/vm/db") modparam("voicemail", "db_url", "mysql://ser:heslo@localhost/ser") modparam("voicemail", "email_column", "email_address") modparam("voicemail", "subscriber_table", "subscriber") modparam("voicemail", "user_column", "username") modparam("voicemail", "domain_column", "domain") # ------------------------- 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; }; # 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 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("nhadzter.com", "subscriber")) { www_challenge("nhadzter.com", "0"); break; }; save("location"); break; }; # Voicemail specific configuration - begin if(method=="ACK" || method=="INVITE" || method=="BYE" || method=="REFER"){ if(t_newtran()){ t_reply("100","Trying -- just wait a minute !"); if(method=="INVITE" || method=="REFER"){ log("**************** vm start - begin ******************\n"); if( uri =~ "conference" ){ if(!vm("/tmp/am_fifo","conference")){ log("could not contact conference server\n"); t_reply("500","could not contact conference server"); }; } else if( uri =~ "echo" ){ if(!vm("/tmp/am_fifo","echo")){ log("could not contact echo\n"); t_reply("500","could not contact echo"); }; } else { if(!vm("/tmp/am_fifo","voicemail")){ log("could not contact voicemail\n"); t_reply("500","could not contact voicemail"); }; }; log("**************** vm start - end ******************\n"); break; }; if(method=="BYE"){ log("**************** vm end/refer - begin ******************\n"); if(!vm("/tmp/am_fifo","bye")){ log("could not contact the media server\n"); t_reply("500","could not contact the media server"); }; log("**************** vm end/refer - end ********************\n"); break; }; } else { log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); }; }; # Voicemail specific configuration - end # 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(); }; } ------------- end of ser.cfg ------------------ ---------start of sems.conf ----------------------- # $Id: sems.conf.sample,v 1.11.2.1 2004/01/27 15:08:28 rco Exp $ # # sems.conf.sample # # Sip Express Media Server (sems) # # sample configuration file # # # whitespaces (spaces and tabs) are ignored # comments start with a "#" and may be used inline # # example: option=value1, value2 # i like this option # ################################## # global parameters # ################################## # optional parameter: fork={yes|no} # # - specifies if sems should run in daemon mode (background) # (fork=no is the same as -E) fork=yes # optional parameter: stderr={yes|no} # # - debug mode: do not fork and log to stderr # (stderr=yes is the same as -E) stderr=no # optional parameter: loglevel={0|1|2|3} # # - sets log level (error=0, warning=1, info=2, debug=3) # (same as -D) loglevel=1 # optional parameter: fifo_name=<filename> # # - path and file name of our fifo file (same as -i) fifo_name=/tmp/am_fifo # optional parameter: ser_fifo_name=<filename> # # - path and file name of Ser's fifo file (same as -o) ser_fifo_name=/tmp/ser_fifo # optional parameter: plugin_path=<path> # # - sets the path to the plug-ins # - may be absolute or relative to CWD plugin_path=/usr/local/lib/sems/plug-in/ # optional parameter: smtp_server=<hostname> # # - sets address of smtp server smtp_server=localhost # optional parameter: smtp_port=<port> # # - sets port of smtp server smtp_port=25 ################################## # module specific parameters # ################################## # sample voicemail configuration (inline) config.voicemail=inline # optional parameter: announce_path=<path> # # - sets the path where announce files are searched for # - the file to be played is determined the following way: # <announce_path>/<domainname>/<username>.wav # if this file is not available <announce_path>/<default_anounce> is used announce_path=/usr/local/lib/sems/audio/ # parameter: default_announce=<filename> # # - sets the name of the default announce WAV file default_announce=default_en.wav # parameter: max_record_time=<seconds> # # - maximum record time max_record_time=30 # parameter: accept_delay=<x> # # - delays accepting of the call for x seconds # - default value is 0 accept_delay=20 # end of configuration section # for voicemail module config.voicemail=end # sample announcement configuration (inline) config.announcement=inline # optional parameter: announce_path=<path> # # - sets the path where announce files are searched for announce_path=/usr/local/lib/sems/audio/ # parameter: default_announce=<filename> # # - sets the name of the default announce WAV file default_announce=default_en.wav # end of configuration section for announcement module config.announcement=end # sample isdngw module configuration (external file) # config.isdngw=/etc/isdngw.conf # sample ivr module configuration (inline) config.ivr=inline #parameter: python_script_path=<full path> python_script_path=/etc/ivr #parameter: python_script_file=<filename> python_script_file=example.py # end of configuration section for ivr module config.ivr=end # sample conference configuration (inline) config.conference=inline # parameter: default_announce=<filename> # # - sets the full pathed name of the default announce WAV file. # Will be played to lonely users. default_announce=/usr/local/lib/sems/audio/first_participant.wav # end of configuration section for conference module config.conference=end # example configuration for number reader config.number_reader=inline number_path=/usr/local/lib/sems/audio/ prolog_file=welcome_to_number_reader.wav epilog_file=thanks_calling_number_reader.wav # end of number_reader configuration config.number_reader=end # add more module configurations here (inline or external): # # config.mymodule=<filename> # or # config.mymodule=inline # ... # config.mymodule=end ------------- end of sems.conf ------------------- -- -- iTouch Communications, Inc.

19 years, 9 months

RE: [Serusers] Question on How Start MEDIAPROXY.

by Ricardo Martinez

Hello. For those who have the same problem i have found a "work-around" solution. In the pyhton file, for example mediaproxy.py, there is a #! at the beggining of the file with the route to the bin. You have to change this for : [prompt %]which pyhton2.3 /usr/local/bin/python2.3 this path is replaced in the mediaproxy.py file : #!/usr/local/bin/python2.3 '''Provides a RTP Proxy server to be used in conjunction with SER's mediaproxy module. MediaProxy stands in between the 2 SIP clients participating in a VoIP conversation and forwards the RTP and RTCP streams between them, allowing the SIP clients to work behind NAT without special settings on their behalf. MediaProxy must have a public IP. ''' .... at least that works for me ! Goos Luck. Ricardo Javier Martinez Ogalde Ingeniero de Desarrollo VOISS NET S.A. Cisco Certified Network Associate * : (56 2) 240 81 96 * : (56 2) 245 74 95 * : rmartinez(a)redvoiss.net > -----Mensaje original----- > De: Ricardo Martinez [mailto:rmartinez@redvoiss.net] > Enviado el: Martes, 08 de Febrero de 2005 13:03 > Para: 'serusers(a)lists.iptel.org' > Asunto: [Serusers] Question on How Start MEDIAPROXY. > > > Hello list. > I have a question on How to Start Mediaproxy server. > When i start > mediaproxy from the console i obtain the following error: > > [root@sipproxy mediaproxy]# ./mediaproxy.py > File "./mediaproxy.py", line 84 > print >>sys.stderr, "Mediaproxy error: %s. Exiting." % why > ^ > SyntaxError: invalid syntax > > however i have installed Python2.3 > [root@sipproxy mediaproxy]# python > python python1.5 python2 python2.2 python2.3 > > When i start mediaproxy with the next lines it works ok : > > [root@sipproxy mediaproxy]# python2.3 mediaproxy.py > > So i made a test and i started mediaproxy with the original python. > > [root@sipproxy mediaproxy]# python mediaproxy.py > File "mediaproxy.py", line 84 > print >>sys.stderr, "Mediaproxy error: %s. Exiting." % why > ^ > SyntaxError: invalid syntax > > And i obtained the same initial error. > My question is : Is there a way to force mediaproxy to start with the > pyhton2.3? and therefore make mediaproxy start with > %./mediaproxy.py and > pyhton2.3 > > Hope that someone can help me here. > > Thanks in advance, > > Ricardo Martinez.- > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

19 years, 9 months

[Serusers] t_relay results in 483 error

by pat newham

Hello, I am new to this list and SEr and am trying a very simple scenario. I would like to establish a call using SER via two grandstream BT100's(UA1 and UA2). The two agent and SER are all on public addresses. The two agents register with SER ok. However when I try to ring UA2 from UA1, I get a 483 error. I know that this mean there are too many hops. My config is practically the same as the basic (i.e. standard)configuration file (except for authorization and I have added a line including an alias equal to the public address of ser....the "if (uri==myself)" statement is correct I think...I checked). I checked the loopback address as suggested in the archives and can see that the invite is repeatedly sent to ser itself and 100 trying messages are sent to ser itself, therefore the max forwards value value must be reached and this is why the error occurs. I note that when I comment out all the t_relay()'s in ser.cfg, the error doesnt occur however the invite from UA1 reached ser and thats it.(i.e. its not forwarded)...Any help would be greatly appreciated(I run the most stable version of ser released i.e 0.8.14) Regards, Pat. ___________________________________________________________ ALL-NEW Yahoo! Messenger - all new features - even more fun! http://uk.messenger.yahoo.com

19 years, 9 months

[Serusers] Group module

by Alex

Hi all. If there any way to check group,username,password together. I am using radius with mysql. is_user_in("From", "somegroup") is checking if username is have some group privileges, but it's not checking the password. Another question if any one had deal with cisco 7960 phones, and if there any problem with registrations requests, for some reason i changed the sip proxy to mine sip server and i don't see any registration requests. Thanks a lot. --------------------------------- Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term'

19 years, 9 months

[Serusers] STUN Server??? Has anyone got a working one??

by Eric Haskins

Hey All, I am in need of a working STUN Server all the GPL'd versions are dead links?? Eric

19 years, 9 months

[Serusers] rtpproxy & PSTN config

by support

Hi, I am running ser0.8.14 and rtpproxy. Could anyone help me how to configure ser.cfg so that I can forward sip call to pstn gateway once I receive a prefix 9? I know that I need to add something like "t_relay_to_udp" and "rewritehost" commands, however, I do not know how to add these commands in my script. Thank you so much if you could give some advice to me. Thomas ---------------------- Here is my config. ---------------------- # # $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=7 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (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/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/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/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/ser/modules/auth.so" #loadmodule "/usr/lib/ser/modules/auth_db.so" # !! Nathelper loadmodule "/usr/lib/ser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- 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) # -- 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) # !! 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 { fix_nated_contact(); }; }

19 years, 9 months

[Serusers] rtpproxy & PSTN config

by support

19 years, 9 months

[Serusers] RIPProxy vs. MediaProxy

by info＠beeplove.com

Hello List: I am going to install SIP Proxy but not sure which one should I pick, RIPProxy or MediaProxy. Would you please post pros and cons of both. Please don't take this mail as I am going to start a debate between the users of RIP and Media. I am just trying to find which one will suit me best. Thanks, MOhammad sip:beeplove@iptel.org

19 years, 9 months

[Serusers] using variables, modifying "FROM" field

by Piet Van den Abeele

Hi, In our telephony network, telephones have a private telephone number and a public telephone number. I want the CLIP to be the private telephone number when calls are made to other company phones, and CLIP to be the public telephone number when calling the outside world through the gateway. eg: private number: 911/XXXXX <-> 02/52XXXXX public number 921/XXXX <-> 03/204XXXX and so on. I know I could hardcode this in SER using something similar to subst("/^From:."(.*)sip:([^@]*@[a-zA-Z0-9.]+(.*))$/From: "+\1sip:+\2/"); (got that from a previous mail :)), but my numbering plan is too complicated, and changes too often to implement it that way. I'd prefer it to be in some database table which I could access through some exec-statement. For Uri translation this would be easy with the exec_dst statement, but I can't see any way to put the outcome of an exec statement in some sort of variable, which I would then use in the subst() statement. Or am I overlooking something??? Thanks, Piet

19 years, 9 months

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sr-users February 2005