sr-users August 2004

sr-users@lists.kamailio.org

181 participants
424 discussions

[Serusers] SERWEB + PostgreSQL : Login trouble

by SB

Hello, I'm using SERWEB but I have some trouble to login when the login/password is correct. If the password is wrong, I have an error message telling me "wrong password". But if the login/password is Ok, then the page start a long processing cycle without anything in return (after 5 minutes, blank page or server unreachable). In my configuration, I have: clear_text_pw=1 My default domain match with the admin user's domain i'm using to login. The database I'm using is PostgreSQL (all serweb sql scripts have been ported to PGSQL, I will provide all the sql files as soon as everything is running smoothly). Any clue ? Thanks SB

20 years

[Serusers] VoiceMail Help

by Veselin Iordanov

ok i found example configuration fails for making ser+sems+voice mail somewhare in the mailing list but have one strange problem.. all of the servers are on one PC This is a configuration files 1. conf file for main ser server # # $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=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) listen=80.72.85.178 port=5060 children=4 fifo="/tmp/ser_fifo" alias=testinfo.intersoftpro.com 127.0.0.1 80.72.85.178 # ------------------ 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" # 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" # load the voicemail module #loadmodule "/usr/local/lib/ser/modules/vm.so" # load the enum module loadmodule "/usr/local/lib/ser/modules/enum.so" # load the group module, to verify if a user forwards to voicemail loadmodule "/usr/local/lib/ser/modules/group.so" # load the nathelper module loadmodule "/usr/local/lib/ser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- # -- registrar parameter # special NAT flag indicates that a registered client is behind NAT modparam("registrar", "nat_flag", 6) # -- 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") # -- voicemail params -- modparam("group", "db_url","mysql://serro:heslo@localhost/ser") # -- nathelper params -- modparam("nathelper", "natping_interval", 60) modparam("nathelper", "ping_nated_only", 1) modparam("tm", "fr_inv_timer", 30 ) #modparam("tm", "fr_inv_timer", 8 ) # ------------------------- request routing logic ------------------- # main routing logic route{ log(1, "-------------------------------------------\n"); log(1, "entering main loop\n"); if (nat_uac_test("2")) { log(1, "src address different than via header->NAT detected\n"); log(1, "force_rport and fix_nated_contact and setflag(5)\n"); #try NAT traversal, works only if the client is symmetrical force_rport(); fix_nated_contact(); append_hf("P-hint: fixed NAT contact for request\r\n"); # flag 5 indicates that incoming request is from NATed client setflag(5); }; 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; }; # create transaction state; abort if error occured # if ( !t_newtran()) { # sl_reply_error(); # break; # }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { log(1, "analyzing REGISTER request\n"); # Uncomment this if you want to use digest authentication if (!www_authorize("80.72.85.178", "subscriber")) { www_challenge("80.72.85.178", "0"); break; }; if (isflagset(5)) { #register from nated client, save nat_flag=6 #in location table setflag(6); }; if (!save("location")) { log(1, "save location error\n"); sl_reply_error(); }; break; }; lookup("aliases"); # check if number beginns with 00, then convert it into a + if (uri=~"^sip:359[0-9]*@") { log(1, "00 International number detected\n"); # strip booth leading "0" strip(2); prefix("+"); }; # check if request uri begins with an internation phone number, if yes, try enum to resolve if (uri=~"^sip:\+[0-9]*@") { log(1, "+ International number detected\n"); if (!enum_query("voice")) { log(1, "ENUM query failed, try enum without voice prefix\n"); if (enum_query("")) { log(1, "ENUM without voice prefix also failed\n"); }; }; if (uri != myself) { log(1, "ENUM lookup revealed an external address, relaying..."); route(1); break; }; }; #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 route(1); } route[1]{ log(1, "-------------------------------------------\n"); log(1, "entering route[1] - relaying SIP message\n"); if ((isflagset(5)) || (isflagset(6))) { log(1, "at least one of the participants is NATed->record_route\n"); record_route(); log(1, " -->setting up reply processing ->onreply_route[1]"); t_on_reply("1"); if (method=="INVITE") { log(1, " INVITE request-->force_rtp_proxy, set NATED-INVITE flag(7)"); force_rtp_proxy(); append_hf("P-hint: request forced to rtp proxy\r\n"); setflag(7); }; }; log(1, "relaying message ...\n"); if (!t_relay()) { log(1, "t_relay error occured\n"); sl_reply_error(); }; } # all incoming replies for t_onrepli-ed transactions enter here onreply_route[1] { log(1, "-------------------------------------------\n"); log(1, "onreply_route[1] entered\n"); if (isflagset(6)) { log(1, "transaction was sent to a NATED client -> fix nated contact\n"); fix_nated_contact(); append_hf("P-hint: fixed NAT contact for response\r\n"); } if ( (status=~"100") ) { log(1, "status 100 received\n"); }; if ( (status=~"180") ) { log(1, "status 180 received\n"); }; if ( (status=~"202") ) { log(1, "status 202 received\n"); }; if ( (status=~"200" || status=~"183") ) { log(1, "status 2xx or 183"); if ( isflagset(7) ) { log(1, "marked(7) as NATED-INVITE -> force_rtp_proxy \n"); force_rtp_proxy(); append_hf("P-hint: response forced to rtp proxy\r\n"); }; }; } 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; }; if ( isflagset(5) ) { log(1, "caller is NATed->record_route\n"); record_route(); log(1, " -->setting up reply processing ->onreply_route[1]"); t_on_reply("1"); if (method=="INVITE") { log(1, " INVITE request-->force_rtp_proxy"); force_rtp_proxy(); }; }; # forward to voicemail now #rewritehostport("machtnix.ict.tuwien.ac.at:5060"); log(1, "forward to voicemail\n"); t_relay_to_udp("80.72.85.178", "5090"); #klaus #forward(machtnix.ict.tuwien.ac.at, 5060) } failure_route[1] { /* XX: note: unsafe if preloaded routes without username used */ log(1, "-------------------------------------------\n"); log(1, "failureroute[1] entered\"); revert_uri(); #rewritehostport("machtnix.ict.tuwien.ac.at:5060"); # append_branch(); t_relay_to_udp("machtnix.ict.tuwien.ac.at", "5060"); } 2. Conf File for second instance of the ser server (voice mail ) # # $Id: voicemail.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=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) listen=80.72.85.178 port=5090 children=4 fifo="/tmp/ser_fifo" alias=testinfo.intersoftpro.com 127.0.0.1 80.72.85.178 # ------------------ 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" # load the voicemail module loadmodule "/usr/local/lib/ser/modules/vm.so" # ----------------- setting module-specific parameters --------------- modparam("voicemail", "db_url","mysql://ser:heslo@localhost/ser") # ------------------------- request routing logic ------------------- # main routing logic route{ log(1, "entering main loop\n"); 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"); # lookup("aliases"); # 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; }; # check changed from 0.8.11 to 0.8.12 # if (len_gt( max_len )) { 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 request-uri"); # break; # }; # Voicemail specific configuration - begin if(method=="ACK" || method=="INVITE" || method=="BYE" || method=="REFER" || method=="CANCEL"){ log(1, "Entering voicemail processing\n"); log(1, "check if new transaction ... \n"); if(t_newtran()){ log(1, "... yes, new transaction\n"); t_reply("100","voicemail ser - Trying -- just wait a minute !"); t_reply("100","voicemail ser - Trying 2 -- just wait a minute !"); if(method=="INVITE" || method=="REFER"){ log("**************** vm start - begin ******************\n"); if( uri =~ "conference" ){ log(1, "incoming conference call\n"); 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" ){ log(1, "incoming echo call\n"); if(!vm("/tmp/am_fifo","echo")){ log("could not contact echo\n"); t_reply("500","could not contact echo"); }; } else { log(1, "incoming voicemail call\n"); 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" || method=="CANCEL"){ 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 sl_send_reply("501", "method not understood here"); log(1, "method not understood here\n"); } 3 Sems configuration fail # $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 and i got this error into second instance of the ser server 0(19586) **************** vm start - end ****************** 0(19586) DEBUG:destroy_avp_list: destroing list (nil) 0(19586) receive_msg: cleaning up 2(19588) ERROR: fifo_server: command t_reply is not available 2(19588) ERROR: fifo_server: command must begin with :: ringing 2(19588) ERROR: fifo_server: command must begin with :: 00004C89728793F0 2(19588) ERROR: fifo_server: command must have at least 3 chars 2(19588) INFO: fifo_server: command empty 2(19588) ERROR: fifo_server: command t_reply is not available 2(19588) ERROR: fifo_server: command must have at least 3 chars 2(19588) ERROR: fifo_server: command must begin with :: 00004C89728793F0 2(19588) ERROR: fifo_server: command must begin with :: Content-Type: application/sdp 2(19588) ERROR: fifo_server: command must begin with :: v=0 2(19588) ERROR: fifo_server: command must begin with :: s=session 2(19588) ERROR: fifo_server: command must begin with :: t=0 0 2(19588) ERROR: fifo_server: command must begin with :: a=rtpmap:0 / 2(19588) INFO: fifo_server: command empty 2(19588) ERROR: fifo_server: command t_reply is not available 2(19588) ERROR: fifo_server: command must begin with :: could not send response. 2(19588) ERROR: fifo_server: command must begin with :: 00004C89728793F0 2(19588) ERROR: fifo_server: command must have at least 3 chars 2(19588) INFO: fifo_server: command empty Tnx in advance _________________________________________________________________ Help STOP SPAM with the new MSN 8 and get 2 months FREE* http://join.msn.com/?page=features/junkmail

20 years

Re: [Serusers] PSTN Routing

by Daniel-Constantin Mierla

Is not recommended to change To header for interoperability issues. Record-route has only meaning for request routing. Daniel On 8/25/2004 11:31 AM, Shirley Toh wrote: >Hi, > >Thank you for your reply. > >I am to change my Request URI but the To header and record_route remains.Is >there any way to overcome, I am trying on exec_dset, can this works? > >Regards, >Shirley > > > >>-----Original Message----- >>From: Daniel-Constantin Mierla [SMTP:daniel@iptel.org] >>Sent: Wednesday, August 25, 2004 4:55 PM >>To: Shirley Toh >>Cc: Serusers (E-mail) >>Subject: Re: [Serusers] PSTN Routing >> >>480 is Temporary unavailable, so there might be no error but the remote >>party unavailable. Check the logs from PSTN gateway, maybe you will find >>something useful to troubleshoot, if you are sure that the remote party >>is available. >> >>Daniel >> >>On 8/25/2004 7:33 AM, Shirley Toh wrote: >> >> >> >>>Hi all, >>> >>>Please help, I am having problem connecting to PSTN after I added a >>> >>> >>Prefix. >> >> >>>I am having the 480 error. Please advice is there any wrong with my >>>configuration file. Thanks in advance!!! >>> >>>Regards, >>>shirley >>> >>><<ser.cfg>> >>> >>> >>>------------------------------------------------------------------------ >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >>> >>> > > >

20 years

[Serusers] Problems Windows Messenger 5.0.0482 and SER

by Wilhelm Farrugia

Hello, Me and other user are both on-line as seen from the output of the serctl showdb command: [root@messenger src]# serctl showdb +----------+-----------------------+---------------------------------------- -------------+ | username | email_address | callid | +----------+-----------------------+---------------------------------------- -------------+ | gpd | gpd(a)maltanet.net | d165527c67a94651b99abc963c7778f2(a)194.158.38.175 | | wilhelm | wilhelm(a)maltanet.net | ad0daa7e6e14410eb54485249c17c262(a)194.158.38.149 | +----------+-----------------------+---------------------------------------- -------------+ Note: Due to usage of cache, server's list may differ from DB list. However when I try to send a message I get: The following message could not be delivered to all recipients The following is the output log using ngrep. U 194.158.38.149:3458 -> 194.158.35.3:5060 INVITE sip:gpd@maltanet.net SIP/2.0..Via: SIP/2.0/UDP 194.158.38.149:10450..Max-Forwards: 70..From: "wilhelm(a)maltanet.net" <sip :wilhelm@maltanet.net>;tag=57fc9d1e5a314be5a0f6d306f7acdc4a;epid=7a0b198ecd. .To: <sip:gpd@maltanet.net>..Call-ID: 86b56c443a1e4 decb9523580851bda50@194.158.38.149..CSeq: 1 INVITE..Contact: <sip:194.158.38.149:10450>..User-Agent: RTC/1.2..Content-Type: app lication/sdp..Content-Length: 109....v=0..o=- 0 0 IN IP4 194.158.38.149..s=session..c=IN IP4 194.158.38.149..t=0 0..m=x-ms-mess age 5060 sip null.. # U 194.158.35.3:5060 -> 194.158.38.149:10450 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 194.158.38.149:10450..From: "wilhelm(a)maltanet.net" <sip:wi lhelm(a)maltanet.net>;tag=57fc9d1e5a314be5a0f6d306f7acdc4a;epid=7a0b198ecd..To : <sip:gpd@maltanet.net>..Call-ID: 86b56c443a1e4dec b9523580851bda50@194.158.38.149..CSeq: 1 INVITE..Server: Sip EXpress router (0.8.12 (i386/linux))..Content-Length: 0..Warning: 392 194.158.35.3:5060 "Noisy feedback tells: pid=15398 req_src_ip=194.158.38.149 req_src_port=3458 in_uri=sip:gpd@maltanet.net out_uri=sip:194.158.38.175:14487 via_cnt==1".... # U 194.158.35.3:5060 -> 194.158.38.175:14487 INVITE sip:194.158.38.175:14487 SIP/2.0..Via: SIP/2.0/UDP 194.158.35.3;branch=z9hG4bKcbaa.ead71c41.0..Via: SIP/2.0/UDP 194.158. 38.149:10450..Max-Forwards: 69..From: "wilhelm(a)maltanet.net" <sip:wilhelm@maltanet.net>;tag=57fc9d1e5a314be5a0f6d306f7acdc4a;ep id=7a0b198ecd..To: <sip:gpd@maltanet.net>..Call-ID: 86b56c443a1e4decb9523580851bda50@194.158.38.149..CSeq: 1 INVITE..Contact: < sip:194.158.38.149:10450>..User-Agent: RTC/1.2..Content-Type: application/sdp..Content-Length: 109....v=0..o=- 0 0 IN IP4 194.1 58.38.149..s=session..c=IN IP4 194.158.38.149..t=0 0..m=x-ms-message 5060 sip null.. # U 194.158.38.175:1703 -> 194.158.35.3:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 194.158.35.3;branch=z9hG4bKcbaa.ead71c41.0..Via: SIP/2.0/UDP 194.158.38.149:10450..From: " wilhelm(a)maltanet.net" <sip:wilhelm@maltanet.net>;tag=57fc9d1e5a314be5a0f6d306f7acdc4a;epid=7a0b198 ecd..To: <sip:gpd@maltanet.ne t>;tag=fb29a56813ed466eb8a67d41f16a7deb..Call-ID: 86b56c443a1e4decb9523580851bda50@194.158.38.149..CSeq: 1 INVITE..User-Agent: RTC/1.2..Content-Length: 0.... # U 194.158.38.175:1703 -> 194.158.35.3:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 194.158.35.3;branch=z9hG4bKcbaa.ead71c41.0..Via: SIP/2.0/UDP 194.158.38.149:10450..From: "wilh elm(a)maltanet.net" <sip:wilhelm@maltanet.net>;tag=57fc9d1e5a314be5a0f6d306f7acdc4a;epid=7a0b198 ecd..To: <sip:gpd@maltanet.net>;t ag=fb29a56813ed466eb8a67d41f16a7deb..Call-ID: 86b56c443a1e4decb9523580851bda50@194.158.38.149..CSeq: 1 INVITE..Contact: <sip:19 4.158.38.175:14487>..User-Agent: RTC/1.2..Content-Type: application/sdp..Content-Length: 109....v=0..o=- 0 0 IN IP4 194.158.38. 175..s=session..c=IN IP4 194.158.38.175..t=0 0..m=x-ms-message 5060 sip null.. # U 194.158.35.3:5060 -> 194.158.38.149:10450 SIP/2.0 200 OK..Via: SIP/2.0/UDP 194.158.38.149:10450..From: "wilhelm(a)maltanet.net" <sip:wilhelm@maltanet.net>;tag=57fc9d1e5a31 4be5a0f6d306f7acdc4a;epid=7a0b198ecd..To: <sip:gpd@maltanet.net>;tag=fb29a56813ed466eb8a67d41f16a7deb..Call-ID: 86b56c443a1e4de cb9523580851bda50@194.158.38.149..CSeq: 1 INVITE..Contact: <sip:194.158.38.175:14487>..User-Agent: RTC/1.2..Content-Type: appli cation/sdp..Content-Length: 109....v=0..o=- 0 0 IN IP4 194.158.38.175..s=session..c=IN IP4 194.158.38.175..t=0 0..m=x-ms-messag e 5060 sip null.. # When using Windows Messenger 4.7.2009 I was working fine, however now that I installed SP2 you cannot work using 4.7.2009, you must install messenger 5. >From a google search it seems that there are a number of users who experience the same problem, one particular links states difference between messenger 4.7 and 5. http://www.mail-archive.com/sip-implementors@cs.columbia.edu/msg05848.html I have noticed that the Windows Messenger 5.0 client expects to receive an INVITE message before it will accept a MESSAGE message from another client. This is a bit problematic, as it is not how many other clients behave (even Windows Messenger 4.7 does not do this). However, in lieu of just complaining about it, I'm trying to get around this behaviour. By the way the strange thing is that sometimes I manage to send messages. Also it seems that I experience presence problems as well. Contacts who are registered to the server are not seen on-line. Thank you, Wil

20 years

[Serusers] sip_router/macosx

by Adrian Georgescu

I can confirm SER 8.14 compiles flawlessly and is functional on OSX 10.3 including Radius accounting and MySQL support. Adrian On Jul 27, 2004, at 10:37 AM, Andrei Pelinescu-Onciul wrote: > On Jul 27, 2004 at 10:19, Adrian Georgescu <ag(a)ag-projects.com> wrote: >> agprojects 2004/07/27 10:19:57 CEST >> >> SER CVS Repository >> >> Added files: (Branch: rel_0_8_14) >> macosx README >> macosx/SER SER StartupParameters.plist >> Log: >> Added startup script for MacOSX >> > > You seem to be really serious about the macosx port :-) > > So if it runs ok for you for a while, please send a message to the > list, > so we could advertise it as stable (and not experimental). > > I'm not completely sure of the locking code, which is a ppc assembly > piece of code contributed a long time ago. The question is if anyone > has tried to run it on a dual ppc and under some stress conditions. > > The locking code is probably ok, is just that I don't know power pc > assembly and I never tested it (as opposed to all the other archs). > > > Andrei

20 years

[Serusers] PSTN Routing

by Shirley Toh

Hi all, Please help, I am having problem connecting to PSTN after I added a Prefix. I am having the 480 error. Please advice is there any wrong with my configuration file. Thanks in advance!!! Regards, shirley <<ser.cfg>>

20 years

[Serusers] Two networks and rtpproxy

by Henry Jensen

I have two networks A and B which are connected through two machines. The clients from network A can't connect to the clients from network B and vice versa. Only the two machines can connect to each other. I have installed SER on both machines. If SER A receives a Request for Client B, it forwards this request to SER B. If SER B recieves a request for Client A, it forwards to SER A. Note that Client B can't connect to SER A directly, and Client A can't connect to SER B. 192.168.0.0/24 10.0.0.0/24 SER A------------------SER B | | | | | | Client A Client B (Grandstream (kphone) BT 100) I use the config from http://www.voip-info.org/wiki-SER+example+NAThelper (see below). As for the SIP part, all seems well. The Clients connect, but I hear nothing, so I think, that rtpproxy is not funtional. There are two things that worry me: 1. In the log from SER , there is an entry that says: "a=nortpproxy:yes" What does this mean? 2. Some errors in the log: ERROR: extract_body: message body has lenght zero ERROR: force_rtp_proxy2: can't extract body from the message ERROR: on_reply processing failed I'm not sure, if I understand the whole thing correctly. So, I have some questions: a) Must rtpproxy run on both machines (SER A and SER B) - they do now - or is one enough? I read, that rtpproxy supports "bridge mode", but it isn't documented anywhere. How can I activate the "bridge mode" ? b) Must rtpproxy run before I start ser or after (if it matters at all) ? c) can rtpproxy forward to another rtpproxy at all? If so, how? d) Is there an alternative solution for the scenario described above? Here's the important part of my config: --- cut --- ... if (method=="INVITE") { record_route(); force_rtp_proxy(); /* set up reply processing */ t_on_reply("1"); }; if (!t_relay()) { sl_reply_error(); }; } onreply_route[1] { if (status=~"[12][0-9][0-9]") force_rtp_proxy(); } Regards, Henry

20 years

[Serusers] Still trying to find out where the AA Application Framework can be downloaded.

by John Draper

Hi again, When visiting the IPTel site, they made references to this page: http://www.iptel.org/aa/ "The rapid SIP application development framework". They talk about it, but don't mention if it's a commercial product, or available for download under the open source license. But if you go to the URL you see above, you will not see any links allowing us to download it. Is this just currently under development? If so, by who? This would even be better if that were true. I'm thinking of inspiring a group to develop a Python module to interface to "ser" like some of the other modules, but lack the skills to do so, but I've worked extensively with mod_python under Apache, and I'm convinced this shouldn't be that much different. We also can use some TwistedPython SIP Servers, and I wouldn't be surprised something like this already exists. Anyway, any info on where I can find this AA project, I would really like to know. John

20 years

RE: [Serusers] Voicemail wierdness

by Dave Bath

Update.... I figured out why it was screwing up with the offline users... that append_branch(); statement was screwing it up (sensible when u think about it really). Take it out all is fine. However, still struggling with correct technique for using mediaproxy with vm. It seems to be mainly be a problem with failure routes..... If anyone has any immediate suggestions, let me know.... Otherwise I will have a more detailed look tomorrow and post a more detailed description. On a further note, I have an interesting conundrum... hopefully someone has found a way round this: All users have an alphanumeric address + a 4 digit numerical alias. I am trying to use asterisk for vm primarily because it allows users to dial in and retrieve their mail. To that end, I decided to store their VM boxes as the numerical alias. Herein lies the problem... If I call, for example, 1000@xxxxx then vm works correctly - the 1000 is passed to asterisk which does the lookup. If I call admin@xxxxxxx then obviously it doesn't work.... The vm box admin does not exist in asterisk. If I miss off the revert_uri(); statement, things are slightly better, but the wrong way around... if I call 1000, the alias lookup takes place and admin@xxxxx is passed to asterisk. However, this does not help me... as although always passing asterisk the same username, if a user dials in they cant enter "admin" as the mailbox from a normal phone. What I really need is the opposite of revert_uri in this case... something which will always use the alias...I could use some external script to do a dblookup and then modify the uri but that seems a bit messy, im sure there must be a way in ser to do it... any ideas are very welcome. Many tnanks again, Dave ________________________________ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Dave Bath Sent: 24 August 2004 17:02 To: serusers(a)lists.iptel.org Subject: [Serusers] Voicemail wierdness Hey all, Another wierdness for a sunny tuesday afternoon.... Have been using ser and * quite successfully so far, and am not into the stage of testing voicemail. Currently I am using * with mysql vm storage. I have a failure route which affter the fr_timer expires, does: failure_route[4] { # If we've got here, it means that a call timer has expired, and the user is in the # voicemail group. Therefore we're going to try and hand off the call to the Asterisk # voicemail system. revert_uri(); rewritehostport("asterisk.dev.inmarsat.com:5060"); log (1, "VOICEMAIL: VM-Divert --> Handing off to Asterisk\n"); append_branch(); append_urihf("CC-Diversion: ", "\r\n"); t_relay(); break; } This works great for normal users who are registered but not answering their phone (or indeed are busy it seems - i havent' worked out how to configure this behaviour -- any suggestions?). However, i have two problems: (1) If the call is from Nonnat UA --> SER --> Mediaproxy --> Nat UA | \/ Asterisk VM then the Nonnat UA doesn't get audio from asterisk vm. I think this has somethign to do with mediaproxy trying to proxy the vm request when it shouldnt and not knwoing where to send it - again, any suggestions? (2) The most annoying issue at the moment is for offline users. It seems logical that if you have an offline user (one that has been found in the subscriber db but just isnt' registered at the moment) you might want their address to divert to vm. However, no amount of fiddling has enabled be to use a failure route for this, and i'm not sure why. In desperation i just created a new route block to handle it - the same as above, but called route[4] instead of failure_route. This at least forwards the request correctly. However, all i get is "The p...." and then the media stream cuts out, and * console shows the channel has been hung up. Below is the ngrep. For some reason two INVITE reuqests are sent from ser(.136) to *(.137) and i'm not sure why. Notice also that a CANCEL is generated, which corresponds to the media stream dying almost as soon as it starts, but again i'm not sure why.... If anyone has anyideas... if you need to look at the ser.cfg then let me know. I'm sure there must be a mor elegant way of handling vm but im not sure what it is at the moment! Dave U 161.30.94.151:2378 -> 161.30.94.136:5060 INVITE sip:1000@sip.dev.inmarsat.com SIP/2.0..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..T o: sip:1000@sip.dev.inmarsat.com..Call-Id: call-1093359137-28@161.30.94.151..Cseq: 1 INVITE..Cont act: <sip:test3@161.30.94.151>..Content-Type: application/sdp..Content-Length: 308..Accept-Langua ge: en..Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE..Supported: sip-cc, sip-cc-01, timer, replaces..User-Agent: Pingtel/2.1.11 (VxWorks)..Date: Tue, 24 Aug 2004 14:52:19 GMT..Via: SIP/2.0/UDP 161.30.94.151....v=0..o=Pingtel 5 5 IN IP4 161.30.94.151..s=phone- call..c=IN IP4 161.30.94.151..t=0 0..m=audio 8766 RTP/AVP 96 97 0 8 18 98..a=rtpmap:96 eg711u/800 0/1..a=rtpmap:97 eg711a/8000/1..a=rtpmap:0 pcmu/8000/1..a=rtpmap:8 pcma/8000/1..a=rtpmap:18 g729/ 8000/1..a=fmtp:18 annexb=no..a=rtpmap:98 telephone-event/8000/1.. # U 161.30.94.136:5060 -> 161.30.94.151:5060 SIP/2.0 407 Proxy Authentication Required..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..To: sip:1000@sip.dev.inmarsat.com;tag=9d060f1cca80def62c99c3e616c718b0.21f2. .Call-Id: call-1093359137 -28@161.30.94.151..Cseq: 1 INVITE..Via: SIP/2.0/UDP 161.30.94.151..Proxy-Authenticate: Digest rea lm="sip.dev.inmarsat.com", nonce="412b574f5e5a864751e3e96fb32f3f7a4929a17a"..Server: Sip EXpress router (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392 161.30.94.136:5060 "Noisy feedback tells: pid=25819 req_src_ip=161.30.94.151 req_src_port=2378 in_uri=sip:1000@sip.dev.inmarsat.com out_uri=sip:admin@sip.dev.inmarsat.com via_cnt==1".... # U 161.30.94.151:2378 -> 161.30.94.136:5060 ACK sip:1000@sip.dev.inmarsat.com SIP/2.0..Contact: sip:test3@161.30.94.151..From: sip:test3@sip. dev.inmarsat.com;tag=2c24948..To: sip:1000@sip.dev.inmarsat.com;tag=9d060f1cca80def62c99c3e616c71 8b0.21f2..Call-Id: call-1093359137-28@161.30.94.151..Cseq: 1 ACK..Accept-Language: en..User-Agent : Pingtel/2.1.11 (VxWorks)..Date: Tue, 24 Aug 2004 14:52:19 GMT..Via: SIP/2.0/UDP 161.30.94.151.. Content-Length: 0.... # U 161.30.94.151:2378 -> 161.30.94.136:5060 INVITE sip:1000@sip.dev.inmarsat.com SIP/2.0..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..T o: sip:1000@sip.dev.inmarsat.com..Call-Id: call-1093359137-28@161.30.94.151..Cseq: 2 INVITE..Cont act: <sip:test3@161.30.94.151>..Content-Type: application/sdp..Content-Length: 308..Accept-Langua ge: en..Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE..Supported: sip-cc, sip-cc-01, timer, replaces..User-Agent: Pingtel/2.1.11 (VxWorks)..Date: Tue, 24 Aug 2004 14:52:19 GMT..Proxy-Authorization: DIGEST USERNAME="test3", REALM="sip.dev.inmarsat.com", NONCE=" 412b574f5e5a864751e3e96fb32f3f7a4929a17a", URI="sip:1000@sip.dev.inmarsat.com", RESPONSE="520ee04 3cedf1c3d2ccbc0fa05101d2c"..Via: SIP/2.0/UDP 161.30.94.151....v=0..o=Pingtel 5 5 IN IP4 161.30.94 .151..s=phone-call..c=IN IP4 161.30.94.151..t=0 0..m=audio 8766 RTP/AVP 96 97 0 8 18 98..a=rtpmap :96 eg711u/8000/1..a=rtpmap:97 eg711a/8000/1..a=rtpmap:0 pcmu/8000/1..a=rtpmap:8 pcma/8000/1..a=r tpmap:18 g729/8000/1..a=fmtp:18 annexb=no..a=rtpmap:98 telephone-event/8000/1.. # U 161.30.94.136:5060 -> 161.30.94.151:5060 SIP/2.0 100 trying -- your call is important to us..From: sip:test3@sip.dev.inmarsat.com;tag=2c24 948..To: sip:1000@sip.dev.inmarsat.com..Call-Id: call-1093359137-28@161.30.94.151..Cseq: 2 INVITE ..Via: SIP/2.0/UDP 161.30.94.151..Server: Sip EXpress router (0.8.14 (i386/linux))..Content-Lengt h: 0..Warning: 392 161.30.94.136:5060 "Noisy feedback tells: pid=25827 req_src_ip=161.30.94.151 req_src_port=2378 in_uri=sip:1000@sip.dev.inmarsat.com out_uri=sip:1000@asterisk.dev.inmarsat.com :5060 via_cnt==1".... # U 161.30.94.136:5060 -> 161.30.94.137:5060 INVITE sip:1000@asterisk.dev.inmarsat.com:5060 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:1000 @161.30.94.136;ftag=2c24948;lr=on>..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..To: sip:100 0@sip.dev.inmarsat.com..Call-Id: call-1093359137-28@161.30.94.151..Cseq: 2 INVITE..Contact: <sip: test3(a)161.30.94.151>..Content-Type <mailto:test3@161.30.94.151%3e..Content-Type> : application/sdp..Content-Length: 308..Accept-Language: en..Al low: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE..Supported: sip-cc, si p-cc-01, timer, replaces..User-Agent: Pingtel/2.1.11 (VxWorks)..Date: Tue, 24 Aug 2004 14:52:19 G MT..Proxy-Authorization: DIGEST USERNAME="test3", REALM="sip.dev.inmarsat.com", NONCE="412b574f5e 5a864751e3e96fb32f3f7a4929a17a", URI="sip:1000@sip.dev.inmarsat.com", RESPONSE="520ee043cedf1c3d2 ccbc0fa05101d2c"..Via: SIP/2.0/UDP 161.30.94.136;branch=z9hG4bK4773.683f9521.0..Via: SIP/2.0/UDP 161.30.94.151..CC-Diversion: sip:1000@sip.dev.inmarsat.com....v=0..o=Pingtel 5 5 IN IP4 161.30.94 .151..s=phone-call..c=IN IP4 161.30.94.151..t=0 0..m=audio 8766 RTP/AVP 96 97 0 8 18 98..a=rtpmap :96 eg711u/8000/1..a=rtpmap:97 eg711a/8000/1..a=rtpmap:0 pcmu/8000/1..a=rtpmap:8 pcma/8000/1..a=r tpmap:18 g729/8000/1..a=fmtp:18 annexb=no..a=rtpmap:98 telephone-event/8000/1.. # U 161.30.94.136:5060 -> 161.30.94.137:5060 INVITE sip:1000@asterisk.dev.inmarsat.com:5060 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:1000 @161.30.94.136;ftag=2c24948;lr=on>..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..To: sip:100 0@sip.dev.inmarsat.com..Call-Id: call-1093359137-28@161.30.94.151..Cseq: 2 INVITE..Contact: <sip: test3(a)161.30.94.151>..Content-Type <mailto:test3@161.30.94.151%3e..Content-Type> : application/sdp..Content-Length: 308..Accept-Language: en..Al low: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE..Supported: sip-cc, si p-cc-01, timer, replaces..User-Agent: Pingtel/2.1.11 (VxWorks)..Date: Tue, 24 Aug 2004 14:52:19 G MT..Proxy-Authorization: DIGEST USERNAME="test3", REALM="sip.dev.inmarsat.com", NONCE="412b574f5e 5a864751e3e96fb32f3f7a4929a17a", URI="sip:1000@sip.dev.inmarsat.com", RESPONSE="520ee043cedf1c3d2 ccbc0fa05101d2c"..Via: SIP/2.0/UDP 161.30.94.136;branch=z9hG4bK4773.683f9521.1..Via: SIP/2.0/UDP 161.30.94.151..CC-Diversion: sip:1000@sip.dev.inmarsat.com....v=0..o=Pingtel 5 5 IN IP4 161.30.94 .151..s=phone-call..c=IN IP4 161.30.94.151..t=0 0..m=audio 8766 RTP/AVP 96 97 0 8 18 98..a=rtpmap :96 eg711u/8000/1..a=rtpmap:97 eg711a/8000/1..a=rtpmap:0 pcmu/8000/1..a=rtpmap:8 pcma/8000/1..a=r tpmap:18 g729/8000/1..a=fmtp:18 annexb=no..a=rtpmap:98 telephone-event/8000/1.. # U 161.30.94.137:5060 -> 161.30.94.136:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 161.30.94.136;branch=z9hG4bK4773.683f9521.0..Via: SIP/2.0/UD P 161.30.94.151..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..To: sip:1000@sip.dev.inmarsat. com;tag=as78a37589..Call-ID: call-1093359137-28@161.30.94.151..CSeq: 2 INVITE..User-Agent: Asteri sk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Contact: <sip:1000@161.30.94.137>..Conte nt-Length: 0.... # U 161.30.94.137:5060 -> 161.30.94.136:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 161.30.94.136;branch=z9hG4bK4773.683f9521.1..Via: SIP/2.0/UD P 161.30.94.151..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..To: sip:1000@sip.dev.inmarsat. com;tag=as78a37589..Call-ID: call-1093359137-28@161.30.94.151..CSeq: 2 INVITE..User-Agent: Asteri sk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Contact: <sip:1000@161.30.94.137>..Conte nt-Length: 0.... # U 161.30.94.137:5060 -> 161.30.94.136:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 161.30.94.136;branch=z9hG4bK4773.683f9521.0..Via: SIP/2.0/UDP 16 1.30.94.151..Record-Route: <sip:1000@161.30.94.136;ftag=2c24948;lr=on>..From: sip:test3@sip.dev.i nmarsat.com;tag=2c24948..To: sip:1000@sip.dev.inmarsat.com;tag=as78a37589..Call-ID: call-10933591 37-28@161.30.94.151..CSeq: 2 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTION S, BYE, REFER..Contact: <sip:1000@161.30.94.137>..Content-Type: application/sdp..Content-Length: 288....v=0..o=root 2063 2063 IN IP4 161.30.94.137..s=session..c=IN IP4 161.30.94.137..t=0 0..m=au dio 10248 RTP/AVP 18 0 97 3 98..a=rtpmap:18 G729/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:97 iLBC/800 0..a=rtpmap:3 GSM/8000..a=rtpmap:98 telephone-event/8000..a=fmtp:98 0-16..a=silenceSupp:off - - - -.. # U 161.30.94.136:5060 -> 161.30.94.151:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 161.30.94.151..Record-Route: <sip:1000@161.30.94.136;ftag=2c2494 8;lr=on>..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..To: sip:1000@sip.dev.inmarsat.com;tag =as78a37589..Call-ID: call-1093359137-28@161.30.94.151..CSeq: 2 INVITE..User-Agent: Asterisk PBX. .Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Contact: <sip:1000@161.30.94.137>..Content-Type : application/sdp..Content-Length: 288....v=0..o=root 2063 2063 IN IP4 161.30.94.137..s=session.. c=IN IP4 161.30.94.137..t=0 0..m=audio 10248 RTP/AVP 18 0 97 3 98..a=rtpmap:18 G729/8000..a=rtpma p:0 PCMU/8000..a=rtpmap:97 iLBC/8000..a=rtpmap:3 GSM/8000..a=rtpmap:98 telephone-event/8000..a=fm tp:98 0-16..a=silenceSupp:off - - - -.. # U 161.30.94.136:5060 -> 161.30.94.137:5060 CANCEL sip:1000@asterisk.dev.inmarsat.com:5060 SIP/2.0..Via: SIP/2.0/UDP 161.30.94.136;branch=z9h G4bK4773.683f9521.1..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..Call-Id: call-1093359137-2 8@161.30.94.151..To: sip:1000@sip.dev.inmarsat.com..Cseq: 2 CANCEL..User-Agent: Sip EXpress route r(0.8.14 (i386/linux))..Content-Length: 0.... # U 161.30.94.137:5060 -> 161.30.94.136:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 161.30.94.136;branch=z9hG4bK4773.683f9521.1..From: sip:test3@sip .dev.inmarsat.com;tag=2c24948..To: sip:1000@sip.dev.inmarsat.com;tag=as78a37589..Call-ID: call-10 93359137-28@161.30.94.151..CSeq: 2 CANCEL..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Contact: <sip:1000@161.30.94.137>..Content-Length: 0.... # U 161.30.94.151:2394 -> 161.30.94.136:5060 ACK sip:1000@161.30.94.136;ftag=2c24948;lr=on SIP/2.0..Route: <sip:1000@161.30.94.137>..Contact: sip:test3@161.30.94.151..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..To: sip:1000@sip.dev.i nmarsat.com;tag=as78a37589..Call-Id: call-1093359137-28@161.30.94.151..Cseq: 2 ACK..Accept-Langua ge: en..User-Agent: Pingtel/2.1.11 (VxWorks)..Date: Tue, 24 Aug 2004 14:52:19 GMT..Via: SIP/2.0/U DP 161.30.94.151..Content-Length: 0.... # U 161.30.94.136:5060 -> 161.30.94.137:5060 ACK sip:1000@161.30.94.137 SIP/2.0..Max-Forwards: 10..Contact: sip:test3@161.30.94.151..From: sip :test3@sip.dev.inmarsat.com;tag=2c24948..To: sip:1000@sip.dev.inmarsat.com;tag=as78a37589..Call-I d: call-1093359137-28@161.30.94.151..Cseq: 2 ACK..Accept-Language: en..User-Agent: Pingtel/2.1.11 (VxWorks)..Date: Tue, 24 Aug 2004 14:52:19 GMT..Via: SIP/2.0/UDP 161.30.94.136;branch=0..Via: SI P/2.0/UDP 161.30.94.151..Content-Length: 0.... exit

20 years

[Serusers] new user help appreciated

by Dimitri Georganas

Hi everybody, I'm new to this list and currently trying to get things working with ser. We have a ser server behind a cisco soho router (DSL) and I enabled udp and tcp on ports 5060, but clients outside cannot register with the server. When I dialin with a vpn connection, registration is no problem. Did anyone experience this problem? Isn't opening up 5060 enough or is the problem in host name resolution. I'm seeing some syslog entries with ser complaining that it cannot resolve 'dg' which isn't a hostname, but my username. Thanks in advance for any help. Dimitri Georganas MITC

20 years

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sr-users August 2004