sr-users May 2004

sr-users@lists.kamailio.org

154 participants
333 discussions

Re: [Serusers] Registered or not
by Kapil Dhawan 18 May '04

18 May '04

what is your default domain SIP_DOMAIN >From: Klaus Darilion <klaus.mailinglists(a)pernau.at> >To: Luis Vazquez <luis(a)teledata.com.uy> >CC: serusers(a)lists.iptel.org >Subject: Re: [Serusers] Registered or not >Date: Mon, 17 May 2004 21:35:35 +0200 > >pepe(a)ipcontact.com.uy and >pepe(a)ser.ipcontact.com.uy and >pepe(a)192.168.2.176 > >are all different users. Probably the error occurs as pepe is registered as >pepe(a)192.168.2.176, but you are tring to call another pepe. > >A solution would be to disable domain support in ser, but ASFAIK this does >not work in current CVS. Thus, use only one domain with your SIP clients. > >regards, >klaus > >Luis Vazquez wrote: >>Hello all >> >>What could be the problem a SIP user agent: >>1) registers correctly with ser, >>2) the register is stored correctly in the MySQL Db (I see it my mysql!) >>3) "serctl ul show" shows the user logged-in (among others), >>but If I try to call this user I get 480 Temporarily Unavailable back from >>ser in the block: >> >> alias="ser.ipcontact.com.uy" >> alias="ipcontact.com.uy" >> ........ >> if (!lookup("location")) { >> if (does_uri_exist()) { >> log(2,"User not logged\n"); >> sl_send_reply("480", "Temporarily Unavailable"); <=== Here >> break; >> } else { >> log(2,"User not found\n"); >> sl_send_reply("404", "Not Found"); >> break; >> }; >> break; >> } >> >>Even stranger: >># serctl ul show pepe //(supose the username is pepe) fails with: >>404 Username pepe(a)ipcontact.com.uy in table location not found >> >>but if I run >> >># serctl ul show >>pepe is shown ... >>...Record(0x402f7d20)... >>domain: 'location' >>aor : 'pepe(a)192.168.2.176' (192.168.2.176 is the IP for the sip server) >> >>This only happens with the CVS version, but I need to use this version. >> >>Any help please??! >> >>Luis >> >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >> >> > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers _________________________________________________________________ Easiest Money Transfer to India. Send Money To 6000 Indian Towns. http://go.msnserver.com/IN/48198.asp Easiest Way To Send Money Home!

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[Serusers] RE: [Sems] sems voicemail error
by wiggler 18 May '04

18 May '04

Hi everyone, Sorry for my second post, i just cc'd the serusers list so ser users can help me with this. I want to have a working voicemail so i run 2 ser instance + sems but i got an error on the logs. Here's my ser.cfg, voicemail.cfg and sems.cfg. I hope anyone can help me, I still got an error on my logs and debugging sems : Running sems -D 3 -E: (27578) DEBUG: write_to_fifo (AmRequest.cpp:743): Write to fifo: completed (27578) DEBUG: getReturnCode (AmRequest.cpp:429): response from Ser: 481 fifo_t_reply: no such transaction (27578) DEBUG: sendToFIFO (AmRequest.cpp:229): sendToFifo returns ret=getReturnCode=0 (27578) ERROR: reply (AmRequest.cpp:325): AmRequestUAS::reply: 481 fifo_t_reply: no such transaction (27578) DEBUG: setLocalPort (AmRtpStream.cpp:133): setting random local port (27578) DEBUG: setLocalPort (AmRtpStream.cpp:163): local rtp port set to 52852 (27578) DEBUG: sendToFIFO (AmRequest.cpp:207): msg=<:vm_reply:00006BBA4D18C226 200 OK 38130:1148081401 00006BBA4C7CEF3F Contact: <sip:1234@212.18.36.252:5090> Content-Type: application/sdp . v=0 o=username 0 0 IN IP4 212.18.36.252 s=session c=IN IP4 212.18.36.252 t=0 0 m=audio 52852 RTP/AVP 3 a=rtpmap:3 // . > (27578) DEBUG: write_to_fifo (AmRequest.cpp:743): Write to fifo: completed (27578) DEBUG: getReturnCode (AmRequest.cpp:429): response from Ser: 481 fifo_t_reply: no such transaction (27578) DEBUG: sendToFIFO (AmRequest.cpp:229): sendToFifo returns ret=getReturnCode=0 (27578) ERROR: reply (AmRequest.cpp:325): AmRequestUAS::reply: 481 fifo_t_reply: no such transaction (27578) ERROR: run (AmSession.cpp:183): 500 could not send response. (27578) DEBUG: sendToFIFO (AmRequest.cpp:207): msg=<:vm_reply:00006BBA45A0D6CB 500 could not send response. 38130:1148081401 00006BBA4C7CEF3F Contact: <sip:1234@212.18.36.252:5090> . . > (27578) DEBUG: write_to_fifo (AmRequest.cpp:743): Write to fifo: completed (27578) DEBUG: getReturnCode (AmRequest.cpp:429): response from Ser: 481 fifo_t_reply: no such transaction (27578) DEBUG: sendToFIFO (AmRequest.cpp:229): sendToFifo returns ret=getReturnCode=0 (27578) ERROR: reply (AmRequest.cpp:325): AmRequestUAS::reply: 481 fifo_t_reply: no such transaction In my messages log: May 17 11:37:35 voip /usr/local/sbin/ser[27478]: ERROR: fifo_t_reply: lookup failed May 17 11:37:35 voip /usr/local/sbin/ser[27478]: ERROR: fifo_server: command (vm_reply) processing failed May 17 11:37:36 voip /usr/local/sbin/ser[27478]: ERROR: fifo_t_reply: lookup failed May 17 11:37:36 voip /usr/local/sbin/ser[27478]: ERROR: fifo_server: command (vm_reply) processing failed May 17 11:37:37 voip /usr/local/sbin/ser[27478]: ERROR: fifo_t_reply: lookup failed May 17 11:37:37 voip /usr/local/sbin/ser[27478]: ERROR: fifo_server: command (vm_reply) processing failed Heres my ser.cfg instance: # # $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=no #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 fifo="/tmp/ser_fifo" fifo_mode=438 alias=siptel.telecomverge.com # ------------------ 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/textops.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/vm.so" loadmodule "/usr/local/lib/ser/modules/enum.so" loadmodule "/usr/local/lib/ser/modules/group.so" # ----------------- setting module-specific parameters --------------- # -- registrar parameter # special NAT flag indicates that a registered client is behind NAT #modparam("registrar", "nat_flag", 6) # -- acc params -- modparam("acc", "db_url", "sql://ser:heslo@localhost/ser") modparam("acc", "log_level", 1) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 1) modparam("acc", "log_flag", 1) # -- 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) modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") # -- 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") modparam("auth_db", "db_url", "mysql://ser:heslo@localhost/ser") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- voicemail params -- modparam("voicemail", "db_url","mysql://ser:heslo@localhost/ser") #modparam("voicemail", "db_url","/etc/ivr/db") # -- voicemail params -- modparam("group", "db_url","mysql://ser:heslo@localhost/ser") # -- nathelper params -- #modparam("nathelper", "natping_interval", 60) #modparam("nathelper", "ping_nated_only", 1) modparam("tm", "fr_inv_timer", 15 ) # ------------------------- request routing logic ------------------- # main routing logic route{ #setflag(1); log(1, "-------------------------------------------\n"); log(1, "entering main loop\n"); if (method=="INVITE"){ record_route(); }; if (method=="INVITE" || method=="BYE" || method=="CANCEL"){ record_route(); setflag(1); }; /* if (method=="REGISTER") log(1, "REGISTER message received\n"); if (method=="INVITE") log(1, "INVITE message received\n"); if (method=="ACK") log(1, "ACK message received\n"); if (method=="BYE") log(1, "BYE message received\n"); if (method=="CANCEL") log(1, "CANCEL message received\n"); if (method=="SUBSCRIBE") log(1, "SUBSCRIBE message received\n"); if (method=="NOTIFY") log(1, "NOTIFY message received\n"); if (method=="OPTIONS") log(1, "OPTIONS message received\n"); if (method=="INFO") log(1, "INFO message received\n"); if (method=="MESSAGE") log(1, "MESSAGE message received\n"); if (method=="REFER") log(1, "REFER message received\n"); */ # 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) { #if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; # loose-route processing if (loose_route()) { log(1, "loose_route processing\n"); 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=~"voip.domain.com") { if (method=="REGISTER") { log(1, "analyzing REGISTER request\n"); record_route(); # Uncomment this if you want to use digest authentication if (!www_authorize("voip.domain.com", "subscriber")) { www_challenge("voip.domain.com", "0"); break; }; if (!save("location")) { log(1, "save location error\n"); sl_reply_error(); }; break; }; lookup("aliases"); #mark transaction for voicemail if (is_user_in("Request-URI", "voicemail\n")) { log(1, "requested user is in voicemail group"); setflag(4); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # handle user which was not found log(1, "requested user not found\n"); route(4); break; }; }; #add failure route which should be performed if response code >=300 if (method=="INVITE" && isflagset(4)) { log(1, "invite for voicemail user->initiate failureroute[1]\n"); t_on_failure("1"); }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { log(1, "t_relay error occured\n"); sl_reply_error(); }; } # all incoming replies for t_onrepli-ed transactions enter here route[4]{ log(1, "-------------------------------------------\n"); log(1, "entering route[4] = requested user not online\n"); # non-Voip -- just send "off-line" if (!(method == "INVITE" || method == "ACK" || method == "CANCEL" || method == "REFER" || method == "BYE")) { log(1, "no invite,ack,cancel,refer->return 404\n"); sl_send_reply("404", "Not Found"); break; }; # not voicemail subscriber and no echo/conference call if ( isflagset(4)) { log(1, "flag(4) active\n"); }; if (uri =~ "conference") { log(1, "conference call\n"); }; if (uri =~ "echo") { log(1, "echo call\n"); }; if ( !( isflagset(4) || (uri =~ "conference") || (uri =~ "echo") ) ) { log(1, "no voicemail subscriber->return 404"); sl_send_reply("404", "Not Found and no voicemail turned on"); break; }; # forward to voicemail now #rewritehostport("voip.domain.com:5090"); log(1, "forward to voicemail\n"); t_relay_to_udp("voip.domain.com", "5090"); } failure_route[1] { /* XX: note: unsafe if preloaded routes without username used */ log(1, "-------------------------------------------\n"); log(1, "failureroute[1] entered\"); revert_uri(); rewritehostport("voip.domain.com:5090"); append_branch(); t_relay_to_udp("voip.domain.com", "5090"); } Heres my voicemail.cfg instance: # # $Id: voicemail.cfg,v 1.2 2003/10/10 06:27:18 jiri Exp $ # # this script is configured for use as voicemail UAS; it processes # INVITEs and BYEs and asks SEMS to record media via "vm"; in this # script, all record-routing and other constructs known from proxy # scripts are not present -- it is a simple UAS # # ----------- global configuration parameters ------------------------ #debug= # debug level (cmd line: -dddddddddd) #fork=no #log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5090 children=4 fifo="/tmp/vm_ser_fifo" fifo_mode=438 # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/vm.so" # ----------------- setting module-specific parameters --------------- modparam("voicemail", "db_url","sql://ser:heslo@localhost/ser") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwars==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; #if (!uri==myself) { # sl_send_reply("404", "not reponsible for host in r-uri"); # break; #}; # Voicemail specific configuration - begin if(method=="ACK" || method=="INVITE" || method=="BYE"){ if (!t_newtran()) { log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); break; }; t_reply("100","Trying -- just wait a minute !"); if(method=="INVITE"){ log("**************** vm start - begin ******************\n"); if (uri=~"sip:as_welcome@.*" || uri=~"sip:as_nomoney@.*") { if (!vm("/tmp/am_fifo", "announcement")) { log("couldn't contact announcement server\n"); t_reply("500", "couldn not contact announcement server"); }; } else { if(!vm("/tmp/am_fifo","voicemail")){ log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; }; log("**************** vm start - end ******************\n"); } else if(method=="BYE"){ log("**************** vm end - begin ******************\n"); if(!vm("/tmp/am_fifo","bye")){ log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; log("**************** vm end - end ******************\n"); }; break; }; if (method=="CANCEL") { sl_send_reply("200", "cancels are junked here"); break; }; sl_send_reply("501", "method not understood here"); } Heres my sems.cfg: # $Id: sems.conf.sample,v 1.13 2004/05/08 17:06:40 ullstar 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=3 # 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=0 # 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=ivr.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 Thanks a lot. wiggler ----- Original Message ----- From: "Raphael Coeffic" <rco(a)iptel.org> To: <sysad(a)bembang.com>, <sems(a)lists.iptel.org> Sent: Wednesday, May 12, 2004 10:40 PM Subject: [Sems] sems voicemail error Hello, could you send us your ser.cfg & sems.conf? it looks like you forgot something... -Raphael. ----- Original Message ----- From: "wiggler" <sysad(a)bembang.com> To: <sems(a)lists.iptel.org> Sent: Wednesday, May 12, 2004 12:33 PM Subject: [Sems] sems voicemail error > Hi everyone, > > I am trying to configure sems for voicemail with no luck. i got an error on my log when i try to call an offline user. > > ---snip------ > May 12 18:30:01 siptel Sems[30819]: Error: AmRequestUAS::reply: 481 fifo_t_reply: no such transaction > ------o---- > > anyone can help me how to fix this problem. > > please advice me on how to setup ser and sems+voicemail properly. what should be the configuration to make it work. > > thanks a lot. > > wiggler > > -- > www.bembang.com ---------------------------------------------------------------------------- ---- > _______________________________________________ > Sems mailing list > sems(a)lists.iptel.org > http://lists.iptel.org/cgi-bin/mailman/listinfo/sems > -- www.bembang.com -- www.bembang.com

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RE: [Serusers] Registerting with new user name
by Karun Chemudugunta 17 May '04

17 May '04

Hi Andy, Thanks for quick responses. Let me expain my problem here. 1) We are trying to use SER to route calls to our speech servers. Each speech server may have multiple user agents. Typically, each server has 50 user agents and each speech channel registers with ser with different AOR and different contact info. Speech can capable of take every call with valid invite messages. 2) I would like to route every incoming from other side( PSTN -- > GW --> SER) any one of registered speech channel. Once speech channels receives incoming call, it deregisters with SER. Our speech channels are going invoke different applications based on incoming URI. How can I configure SER with these requirements ? Thanks, Karun -----Original Message----- From: Andy Blen [mailto:andy.blen@iptel.org] Sent: Monday, May 17, 2004 3:11 PM To: Karun Chemudugunta; serusers(a)lists.iptel.org Subject: RE: [Serusers] Registerting with new user name Karun, unfortunately your logic cannot work at all. First of all, it is not clear if changing user identity for user location database makes sense at all. This does not sound like a reasonably secure idea. Secondly, rewriteuser is used for something else: for rewriting request-URIs and not AOR as stored in To header field. Rewriting To header field is not advisable -- it is used in RFC2543 and RFC3261 for many other purposes. If you are interested like you say in forwarding user A to user B then provison aliases or user location entries and do not try to change the registration process. Andy At 11:47 PM 5/17/2004, Karun Chemudugunta wrote: >-------------------------------------------------------------------------- >REGISTER sip:172.16.12.231 SIP/2.0 >From: sip:olduser@bevocal.com >To: sip:olduser@bevocal.com >Contact: ><sip:olduser@encamed006:5060>;transport=udp;q=1;action=proxy;expires=3600 >Call-ID: 12729303125234000(a)encamed006.bevocal.com >CSeq: 12317 REGISTER >Content-Length: 0 >-------------------------------------------------------------------------- > >Hi List, > > I was trying to replace user name "olduser" using >rewriteuser("newuser") before saving in user db with save("location") with >"newuser", but it it ignoring my rewriteuser("newuser") function in ser.cfg. > >I am trying to forward newuser invite to olduser user agent. > >Please find attached ser.cfg. > >Plz help me to find how to configure ser.cfg to achieve this. > >Thanks in advance. >Karun > > > > > > > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Andy Blen iptel.org Services

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RE: [Serusers] Registerting with new user name
by Andy Blen 17 May '04

17 May '04

Karun, unfortunately your logic cannot work at all. First of all, it is not clear if changing user identity for user location database makes sense at all. This does not sound like a reasonably secure idea. Secondly, rewriteuser is used for something else: for rewriting request-URIs and not AOR as stored in To header field. Rewriting To header field is not advisable -- it is used in RFC2543 and RFC3261 for many other purposes. If you are interested like you say in forwarding user A to user B then provison aliases or user location entries and do not try to change the registration process. Andy At 11:47 PM 5/17/2004, Karun Chemudugunta wrote: >-------------------------------------------------------------------------- >REGISTER sip:172.16.12.231 SIP/2.0 >From: sip:olduser@bevocal.com >To: sip:olduser@bevocal.com >Contact: ><sip:olduser@encamed006:5060>;transport=udp;q=1;action=proxy;expires=3600 >Call-ID: 12729303125234000(a)encamed006.bevocal.com >CSeq: 12317 REGISTER >Content-Length: 0 >-------------------------------------------------------------------------- > >Hi List, > > I was trying to replace user name "olduser" using >rewriteuser("newuser") before saving in user db with save("location") with >"newuser", but it it ignoring my rewriteuser("newuser") function in ser.cfg. > >I am trying to forward newuser invite to olduser user agent. > >Please find attached ser.cfg. > >Plz help me to find how to configure ser.cfg to achieve this. > >Thanks in advance. >Karun > > > > > > > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Andy Blen iptel.org Services

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RE: [Serusers] Registerting with new user name
by Karun Chemudugunta 17 May '04

17 May '04

-------------------------------------------------------------------------- REGISTER sip:172.16.12.231 SIP/2.0 From: sip:olduser@bevocal.com To: sip:olduser@bevocal.com Contact: <sip:olduser@encamed006:5060>;transport=udp;q=1;action=proxy;expires=3600 Call-ID: 12729303125234000(a)encamed006.bevocal.com CSeq: 12317 REGISTER Content-Length: 0 -------------------------------------------------------------------------- Hi List, I was trying to replace user name "olduser" using rewriteuser("newuser") before saving in user db with save("location") with "newuser", but it it ignoring my rewriteuser("newuser") function in ser.cfg. I am trying to forward newuser invite to olduser user agent. Please find attached ser.cfg. Plz help me to find how to configure ser.cfg to achieve this. Thanks in advance. Karun

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[Serusers] Controlling time-outs for inbound vs outbound calling
by Jac Barben 17 May '04

17 May '04

My goal is to control timing for an inbound vs. outbound calls. An inbound call would be from the PSTN to a gateway to SER to a UA registered to SER. An outbound call would be from a UA registered to SER to SER to another UA registered to SER or to a gateway. My experience with the fr_timer and fr_inv_timer is that they seem to be global. Meaning if I set the timer(s) to 30 seconds both my inbound and outbound calls behave the same way. My reason for wanting to differentiate control between incoming and outgoing is that I'd like to cause incoming calls to go to voicemail if they are not answered within 30 seconds. I do not want to restrict my outbound calls to such a short limit, especially if they are bound for the PSTN. Any ideas? Any documentation that I missed? Thanks for all your SUPER support. SER ROCKS! Jac

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[Serusers] cpled authentication problem
by Richard 17 May '04

17 May '04

Hi, I tried to use SIP register method to upload cpl script. However I got a result "bad request". The algorithm used in cpled is set as null. I thought normally that it should be md5. The ser's debug message says "Credentials received are not filled properly". Can someone help me please? Thanks, Richard P.S. Has anyone used cpl extensively? Is it reasonably stable? __________________________________ Do you Yahoo!? SBC Yahoo! - Internet access at a great low price. http://promo.yahoo.com/sbc/ REGISTER sip:example.org SIP/2.0. Via: SIP/2.0/UDP 41.222.132.128:8700. From: sip:888@example.org. To: sip:888@example.org. Call-ID: 14tg4341-1084525599016-295028-32961147(a)41.222.132.128. CSeq: 1 REGISTER. Contact: sip:41.222.132.128:8700. Content-Type: application/cpl+xml. Content-Disposition: script;action=store. Content-Length: 1044. . <?xml version="1.0" encoding="UTF-8"?> <!DOCTYPE cpl PUBLIC '-//IETF//DTD RFCxxxx CPL 1.0//EN' 'cpl.dtd'> <cpl> <subaction id="trash" /> <subaction id="voicemail"> <location url="sip:55592@example.org" priority="0.0" clear="yes"> <redirect /> </location> </subaction> <incoming> <address-switch field="destination" subfield="user"> <address is="user@host"> <location url="sip:55592@example.org"> <proxy> <busy> <time-switch tzid="HST -10"> <time dtstart="20040510T151502" duration="P28DT00H00M00S" freq="weekly" interval="001" byday="MO,WE,FR" bymonthday="1-5,8-10" byyearday="10-20" bymonth="1,4,7,10" /> </time-switch> </busy> <noanswer> <sub ref="voicemail" /> </noanswer> <failure> <lookup source="gost.com" /> </failure> </proxy> </location> </address> <otherwise /> </address-switch> </incoming> </cpl> # U +0.003535 41.222.132.128:5060 -> 41.222.132.128:8700 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 41.222.132.128:8700. From: sip:888@example.org. To: sip:888@example.org;tag=b756ea191e1a1705f44b2100b0983a34.0f1f. Call-ID: 14tg4341-1084525599016-295028-32961147(a)41.222.132.128. CSeq: 1 REGISTER. WWW-Authenticate: Digest realm="example.org", nonce="40a48d4ba435a6dff240e5ceabc9fc5296089728", qop="auth". Server: Sip EXpress router (0.8.13-dev-28 (i386/linux)). Content-Length: 0. Warning: 392 41.222.132.128:5060 "Noisy feedback tells: pid=26138 req_src_ip=41.222.132.128 req_src_port=33739 in_uri=sip:example.org out_uri=sip:example.org via_cnt==1". . # U +0.002629 41.222.132.128:33739 -> 41.222.132.128:5060 REGISTER sip:example.org SIP/2.0. Via: SIP/2.0/UDP 41.222.132.128:8700. From: sip:888@example.org. To: sip:888@example.org. Call-ID: 14tg4341-1084525599016-295028-32961147(a)41.222.132.128. CSeq: 2 REGISTER. Authorization: Digest username="888", realm="example.org", uri="sip:example.org", algorithm=null, nonce="40a48d4ba435a6dff240e5ceabc9fc5296089728", qop="auth", cnonce="fc82c359317018651a99347", response="f0ad5bb30d3d1f11fc31bb69bd54bbab". Contact: sip:41.222.132.128:8700. Content-Type: application/cpl+xml. Content-Disposition: script;action=store. Content-Length: 1044. . <?xml version="1.0" encoding="UTF-8"?> <!DOCTYPE cpl PUBLIC '-//IETF//DTD RFCxxxx CPL 1.0//EN' 'cpl.dtd'> <cpl> <subaction id="trash" /> <subaction id="voicemail"> <location url="sip:55592@example.org" priority="0.0" clear="yes"> <redirect /> </location> </subaction> <incoming> <address-switch field="destination" subfield="user"> <address is="user@host"> <location url="sip:55592@example.org"> <proxy> <busy> <time-switch tzid="HST -10"> <time dtstart="20040510T151502" duration="P28DT00H00M00S" freq="weekly" interval="001" byday="MO,WE,FR" bymonthday="1-5,8-10" byyearday="10-20" bymonth="1,4,7,10" /> </time-switch> </busy> <noanswer> <sub ref="voicemail" /> </noanswer> <failure> <lookup source="gost.com" /> </failure> </proxy> </location> </address> <otherwise /> </address-switch> </incoming> </cpl> # U +0.002369 41.222.132.128:5060 -> 41.222.132.128:8700 SIP/2.0 400 Bad Request. Via: SIP/2.0/UDP 41.222.132.128:8700. From: sip:888@example.org. To: sip:888@example.org;tag=b756ea191e1a1705f44b2100b0983a34.0f1f. Call-ID: 14tg4341-1084525599016-295028-32961147(a)41.222.132.128. CSeq: 2 REGISTER. Server: Sip EXpress router (0.8.13-dev-28 (i386/linux)). Content-Length: 0. Warning: 392 41.222.132.128:5060 "Noisy feedback tells: pid=26138 req_src_ip=41.222.132.128 req_src_port=33739 in_uri=sip:example.org out_uri=sip:example.org via_cnt==1". . # U +0.001366 41.222.132.128:5060 -> 41.222.132.128:8700 SIP/2.0 400 Bad Request. Via: SIP/2.0/UDP 41.222.132.128:8700. From: sip:888@example.org. To: sip:888@example.org;tag=b756ea191e1a1705f44b2100b0983a34.0f1f. Call-ID: 14tg4341-1084525599016-295028-32961147(a)41.222.132.128. CSeq: 2 REGISTER. Server: Sip EXpress router (0.8.13-dev-28 (i386/linux)). Content-Length: 0. Warning: 392 41.222.132.128:5060 "Noisy feedback tells: pid=26138 req_src_ip=41.222.132.128 req_src_port=33739 in_uri=sip:example.org out_uri=sip:example.org via_cnt==1". . # I +0.000035 41.222.132.128 -> 41.222.132.128 3:3 ....E.....@.@...B...B.....!...#.SIP/2.0 400 Bad Request. Via: SIP/2.0/UDP 41.222.132.128:8700. From: sip:888@example.org. To: sip:888@example.org;tag=b756ea191e1a1705f44b2100b0983a34.0f1f. Call-ID: 14tg4341-1084525599016-295028-32961147(a)41.222.132.128. CSeq: 2 REGISTER. Server: Sip EXpress router (0.8.13-dev-28 (i386/linux)). Content-Length: 0. Warning: 392 41.222.132.128:5060 "Noisy feedback tells: pid=26138 req_src_ip=41.222.132.128 req_src_port=33739 in_uri=sip:example.org out_uri=sip:example.org via_cnt==1". . ## U +0.002265 41.222.132.128:5060 -> 41.222.132.128:8700 SIP/2.0 400 Bad Request. Via: SIP/2.0/UDP 41.222.132.128:8700. From: sip:888@example.org. To: sip:888@example.org;tag=b756ea191e1a1705f44b2100b0983a34.0f1f. Call-ID: 14tg4341-1084525599016-295028-32961147(a)41.222.132.128. CSeq: 2 REGISTER. Server: Sip EXpress router (0.8.13-dev-28 (i386/linux)). Content-Length: 0. Warning: 392 41.222.132.128:5060 "Noisy feedback tells: pid=26138 req_src_ip=41.222.132.128 req_src_port=33739 in_uri=sip:example.org out_uri=sip:example.org via_cnt==1". . # I +0.000024 41.222.132.128 -> 41.222.132.128 3:3 ....E.....@.@...B...B.....!...#.SIP/2.0 400 Bad Request. Via: SIP/2.0/UDP 41.222.132.128:8700. From: sip:888@example.org. To: sip:888@example.org;tag=b756ea191e1a1705f44b2100b0983a34.0f1f. Call-ID: 14tg4341-1084525599016-295028-32961147(a)41.222.132.128. CSeq: 2 REGISTER. Server: Sip EXpress router (0.8.13-dev-28 (i386/linux)). Content-Length: 0. Warning: 392 41.222.132.128:5060 "Noisy feedback tells: pid=26138 req_src_ip=41.222.132.128 req_src_port=33739 in_uri=sip:example.org out_uri=sip:example.org via_cnt==1". . ######################################################

2 3

[Serusers] UDP Ports used by RTPPROXY
by Jm LACOMBE 17 May '04

17 May '04

Hi ALL, which are UDP Ports used by RTPPROXY? what's the default value ? how to change the value? thank JML

2 1

[Serusers] correct download and makefile of rtpproxy
by Franz Edler 17 May '04

17 May '04

Hello, I have just downloaded rtpproxy from cvs using the commands - CVSROOT=:pserver:anonymous@cvs.berlios.de:/cvsroot/ser - export CVSROOT - cvs login - cvs co rtpproxy Now I have the sources but obviously no valid Makefile for my SuSE 9.0 distribution. There are now two Makefiles (Makefile.am and Makefile.in) in rtpproxy directory which do not work (--> Nothing to be done). Therefore I have two questions: - Did I download from the right repository ? (I am interested only in stable versions) - How to proceed with makefiles? Any help appreciated. Franz

2 2

[Serusers] SER - TOO MANY HOPS
by onyeka d ezenwoye 17 May '04

17 May '04

Hi, i keep getting this "too many hops" error when i try to call a user registered on my ser server (tiger.mydomain.com). The users is registered as user1(a)tiger.mydomain.com . I have the default ser config file but if i try to call user1(a)mydomain.com (without the machine name "tiger" as part of the address) the server returns "too many hops". However i can make the call if i specify address like user1(a)tiger.mydomain.com. why is this? the domain is the same. what do i do now? Thanks. regards, Onyeka

3 2

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