sr-users October 2003

sr-users@lists.kamailio.org

105 participants
230 discussions

by Sergio Gerardo Festa

hello I have problems to make to operate the voicemail. when it is accomplished the call not lets me to record the message and is lost the ans_machine. Español hola tengo problemas para hacer funcionar el voicemail. cuando se realiza la llamada no me deja grabar el mensaje y se pierde el ans_machine. estos son mis archivos de configuracion. Log messages Oct 28 17:14:09 chitara /usr/sbin/ser[21057]: ACC: call missed: from=Save1<sip:102@mydomain.com>;tag=28778-a725, i-uri=sip:100@mydomain.com, method=INVITE, o-uri=sip:100@262.61.161.129:5060;user=phone, code=488 Not Acceptable Here Oct 28 17:14:09 chitara /usr/sbin/ser[21057]: failure_route[1]:jump to route[4]:vm Oct 28 17:14:09 chitara /usr/sbin/ser[21057]: submit_query(): You have an error in your SQL syntax near ''' at line 1 Oct 28 17:14:09 chitara /usr/sbin/ser[21057]: submit_query(): Error while submitting query Oct 28 17:14:09 chitara /usr/sbin/ser[21057]: ERROR: vm: db_query() failed. Oct 28 17:14:09 chitara /usr/sbin/ser[21057]: ERROR: vm: vm_get_user_info failed Oct 28 17:14:09 chitara /usr/sbin/ser[21057]: route[3]:vm:voicemail failed Oct 28 17:14:09 chitara /usr/sbin/ser[21047]: route[]:eof Oct 28 17:14:10 chitara /usr/sbin/ser[21073]: route[]:eof ---------------------------------------------------------------------------- ------------------------------ SER.CFG # # $Id: ser.cfg,v 1.20 2003/05/31 21:12:19 jiri Exp $ # # config script with voicemail, PSTN dial-out functionality # # ----------- global configuration parameters ------------------------ debug=1 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) # Uncomment these lines to enter debugging mode /* debug=8 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" sip_warning=no listen=262.69.20.92 alias=sip.boxip.com.ar # # ------------------ module loading ---------------------------------- # # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" # loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/vm.so" loadmodule "/usr/lib/ser/modules/pa.so" loadmodule "/usr/lib/ser/modules/msilo.so" loadmodule "/usr/lib/ser/modules/acc.so" loadmodule "/usr/lib/ser/modules/textops.so" # loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/group.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" # # ----------------- 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") # # modparam("tm", "fr_inv_timer", 30) #INVITE timeout modparam("tm", "fr_timer", 15) #negative INVITE reply or no #final reply for a request for ACK # modparam("voicemail", "db_url", "sql://ser:heslo@localhost/ser") # modparam("acc", "db_url", "sql://ser:heslo@localhost/ser") modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 3) # modparam("acc", "log_level", 1) modparam("acc", "log_flag", 2) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_fmt", "fimos") # # ------------------------- request routing logic ------------------- # # main routing logic # alias=262.69.20.92 #sip server IP address alias=mydomain.com #sip server FQDN # route{ # log(1,"entering main route"); setflag(2); #set flag for accounting # 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 (len_gt( 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") { # digest authentication # log(1,"request for registration"); # if (!www_authorize("mydomain.com", "subscriber")) { # www_challenge("mydomain.com", "0"); # break; # }; save("location"); break; }; /* ********** Dial out to PSTN logic ************* */ #forward 411[information] and 911[emergency] requests to gateway if(uri=~"^sip:(9|2)11@(boxip\.com.ar|200\.69\.20\.92)"){ log(1,"411/911 expression match"); route(2); break; }; #forward numerical 7 digit requests to gateway if(uri=~"^sip:[0-9]{7}@(mydomain\.com|262\.69\.20\.92)"){ log(1,"7 digit expression match"); route(2); break; }; # strip 650 and forward to GW if user dials 650 before phone no. if(uri=~"^sip:650[0-9]{7}@mydomain\.com|262\.69\.20\.92)"){ strip(3); log(1,"650 area code dialed, 650 stripped"); route(2); break; }; #forward numerical 10 digit requests to gateway, append a 1 first if(uri=~"^sip:[0-9]{10}@(mydomain\.com|262\.69\.20\.92)"){ prefix("1"); log(1,"10 digit expression match, prefix 1"); route(2); break; }; #forward numerical 11 digit requests that start with a 1 to GW if(uri=~"^sip:1[0-9]{10}@(mydomain\.com|262\.69\.20\.92)"){ log(1,"10 digit exp match w/leading 1"); route(2); break; }; #forward international N digit requests to gateway if(uri=~"^sip:011[0-9]+@(mydomain\.com|262\.69\.20\.92)"){ log(1,"international expression match"); route(2); break; }; /* ********** VOICEMAIL logic ************* */ if (uri=~"^sip:voicemail\+@"){ log(1,"sip:voicemail uri match"); route(3); break; }; /* ****** Find Aliases and Locations of users ********* */ #lookup "aliases" before looking up "location" lookup("aliases"); # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { route(3); break; # this section needs help. I need to differentiate between users that # exits buy aren't online (send these requests to voicemail) # requests for users that aren't in my subscriber database # (send these requests a 404 reply) sl_send_reply("404", "User Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP t_on_failure("1"); if (!t_relay()) { sl_reply_error(); }; log(1,"route[]:eof"); } route[2]{ log(1,"route[2]:SIP-to-PSTN call routed"); rewritehostport("262.69.20.92:5060"); if(!t_relay()){ sl_reply_error(); }; } route[3]{ log(1,"route[3]:vm:1"); if (method=="INVITE" || method=="BYE" || method=="REFER"){ log(1,"route[3]:vm:2"); if(t_newtran()){ t_reply("100","Trying -- just a second"); if(method=="INVITE" || method=="REFER"){ log(1,"route[3]:method==INVITE || REFER"); if(uri =~ "conference" ){ if(!vm("/tmp/am_fifo","conference")){ log(1,"route[3]:vm:conference failed"); t_reply("500","could not contact conference server"); }; } else if (uri =~"echo"){ if(!vm("/tmp/am_fifo","echo")){ log(1,"route[3]:vm:echo failed"); t_reply("500","could not contact echo"); }; } else{ if(!vm("/tmp/am_fifo","voicemail")){ log(1,"route[3]:vm:voicemail failed"); t_reply("500", "voicemail error"); }; }; break; }; if(method=="BYE"){ log(1,"route[3]:vm:method==BYE"); if(!vm("/tmp/am_fifo","bye")){ log(1,"route[3]:vm:bye failed"); t_reply("500" , "could not contact the media server"); }; break; }; } else{ log(1,"route[3]:vm:new transaction failed"); sl_send_reply("500", "new transaction failed"); break; }; }; } route[4]{ # this should be voicemail logic that is specific to a failure_route # i.e. line busy, or timeout after a certain period with no answer if(method=="INVITE" || method=="REFER"){ if(!vm("/tmp/am_fifo","voicemail")){ log(1,"route[3]:vm:voicemail failed"); t_reply("500", "voicemail error"); }; }else if(method=="BYE"){ log(1,"route[3]:vm:method==BYE"); if(!vm("/tmp/am_fifo","bye")){ log(1,"route[3]:vm:bye failed"); t_reply("500" , "could not contact the media server"); }; }; } failure_route[1]{ log(1,"failure_route[1]:jump to route[4]:vm"); # append_branch("sip:info at mydomain.com"); route(4); } ---------------------------------------------------------------------------- ------------------ SEMS.CFG # optional parameter: announce_path=<path> # # - sets the path where announce files are searched for announce_path=/usr/lib/sems/audio/ # optional parameter: default_announce=<filename> # # - sets the name of the default announce WAV file default_announce=default_en.wav # optional parameter: max_record=<seconds> # # - maximum record time max_record=30 # optional parameter: smtp_server=<hostname> # # - sets address of smtp server smtp_server=262.61.161.129 # optional parameter: smtp_port=<port> # # - sets port of smtp server smtp_port=25 ################################## # module specific parameters # ################################## # add more module configurations here (inline or external): # # config.mymodule=<filename> # or # config.mymodule=inline # ... # config.mymodule=end ---------------------------------------------------------------------------- --------------------------- Thanks Gracias Saludos A su Disposición. Sergio Gerardo Festa

21 years

[Serusers] exec module question

by Andres

Hi, I am testing the exec module by outputting some variables to a flat file using: SIP_HF_<hf_name> So for example I put something like exec_msg("echo `date` - From: $SIP_HF_FROM To: $SIP_HF_TO Contact: $SIP_HF_CONTACT Src_IP: $SIP_SRCIP User-Agent: $SIP_HF_USER-AGENT >> /var/log/ser/ser_register.log"); but the User-Agent header field does not get passed even though a sniffer trace shows it in the SIP REGISTER request. Can somebody tell me why not? Thanks, Andres

21 years

[Serusers] Support for offline messages in an IM conference

by Madhuri garigipati

Hello All, I was going through the msilo package in SER modules. It says - "It stores received messages for an offline user and sends them when the user becomes online." I am creating a J2ME IM client for mobile devices using SIP at the backend. And the problem with mobile devices is connection loss. So I am planning to provide a feature to my IM client where in if more than 2 users are in a conference chat room, and one user gets disconnected for some reason, I want to store the messages during the user's absense and send it back to the user when they reconnect. Is this possible with SIP and SER? Does SER module msilo or any other module support offline messages for a conference chat? Thanks, Madhuri. --------------------------------- Do you Yahoo!? Exclusive Video Premiere - Britney Spears

21 years

[Serusers] Messenger5.0 481 Error continued

by Rork, Joseph (J.P.)

I've followed the instructions: Attention: instant messaging model has changed from 4.7 to 5.0 dramatically and in a non-standard way. 5.0 takes session establishement using INVITE and drops out-of-dialog MESSAGEs on the floor, whereas sending "481 Call Leg/Transaction...". To permit them, apply the following registry setting: HighSecurityMode HKLM\Software\Policies\Microsoft\Messenger\Client\{83D4679F-B6D7-11D2-BF36-0 0C04FB90A03}\_Default\EnableSIPHighSecurityMode DWORD= 0 -- Low Security 1 -- High Security 2(Default, same as not set) --- Medium security You need set it to 0 in order to make peer-to-peer IM. Since WM4.7 client is not registered in the LCS server, I guess that WM5 treats it as peer-to-peer call. And I'm still getting the following error: <<messenger5dump>> Any ideas? --joe

21 years

Re: [Serusers] SER Contact rewrite for Softswitch

by Jiri Kuthan

If you need to achieve security, why don't you let the proxy server authenticate callers using digest identity and the gateway trust the proxy by ip address? That's the most common practice today, we use it with our Cisco gateway too. -jiri At 04:27 PM 10/23/2003, Harry Behrens wrote: >Hello, > >we have the following setup in our "Unified Messaging lab": > >SUAs <-> LAN <-> SER <-> Softswitch <-> PSTN > >The SER is provisioned as a SIP Server (essentially a SUA) from the Softswich point of view. > >We have provisioned a number block of 20 numbers (from a 30-channel ISDN PRI) for direct inward dialling (DID) to the SER. >E.g. 12345620-12345639 are routed to the SER. > >On the SER we have a number of users <user>@dai-labor.de, where the first 20 users have an alias >of 123456[23][0-9] >Each user also has his number added to his permanent contacts to enable (on the cheap) a "call by PSTN if not online" poliy. > >The aim obviously being to have a 1-1 mapping between URI and allocated telephone number (we are still in the proof-of-concept phase and will move to DNS and ENUM based architecture at a later stage). > >Scenario 1: OK > PSTN user Up calls 12345621 - which is mapped to harry(a)dai-labor.de. > If harry(a)dai-labor.de is registered, his SIP UA rings and the call is established. > >Scenario 2: problem > harry(a)dai-labor.de wants to use his UA to place a call through the Softswitch. > The Softswitch "authenticates" (if you can call this authentication ;-( by looking at the > Contact header and comparing it to its database of numbers assigned to provisioned SIP Servers. > I therefore need to be able to specify a rewrite of the Contact header to > replace harry@<Contact IP:port> with 12345621@<Contact IP:port> > when the callee is of the form <number>@<whatever> > >I am aware that this would be strictly speaking not proxy but rather B2BUA behaviour, but would still be >interested to know whether anyone has a solution to this problem or whether we need to hack one ourselves. > >Many thanks in advance, > > Harry > >Dr. Harry Behrens >Projektleitung BIB3R >DAI Labor - Technische Universität Berlin >Sekretariat GOR 1-1, Franklinstrasse 28/29, 10587 Berlin > >Fon: +49 30 314 23383 >Fax: +49 30 314 21799 > >Email: harry.behrens(a)dai-labor.de > >http://www.dai-labor.de > > > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

21 years

[Serusers] noch Voicemail

by Chen Youcheng

Hallo, laut Beschreibung von http://iptel.org/ser/doc/modules/txt/vm.txt muss Mann viele Parameters einstellen. Aber das Beispiel von http://www.iptel.org/ser/doc/prerelease/x1294.html ist anders. Nur ein Parameter : modparam("voicemail","db_url","modules/vm/db"). Kann jemand einmal erkl�ren? dayong1973 __________________________________ Do you Yahoo!? Exclusive Video Premiere - Britney Spears http://launch.yahoo.com/promos/britneyspears/

21 years

[Serusers] append_rpid_hf() and search()

by Alexander Mayrhofer

Hi, i'm adding remote party id information to certain SIP messages (to those destined to the PSTN gateway). After doing that, i try to search() the "Remote-Party-ID" header field for a specific string, and it seems to me that search() only works on the initial (received) SIP message rather than on the "current" modified message. Is that assumption correct? (i hope it's not :) If yes, is there another way to search SER's rpid information for a specific string? config snippet: if (!is_rpid_user_e164()) { append_rpid_hf(); xlog("L_INFO", "rpid added"); if (search("Remote-Party-ID:.*@1005")) { xlog("L_INFO", "rpid search succeeded"); }; ... thanx axelm

21 years

[Serusers] SIP Instant Messaging UA

by dhiraj.2.bhuyan＠bt.com

Hello List, Can anyone recommend some SIP Instant Messaging UA (linux and windows) that follows the specification word by word? I tried MSN Messenger 4.6 and 5 - both appears to have some problems (version 5 tries to send a MESSAGE with the INVITE method and gets back a BYE in response!!!!!!!!!!!!!). Thanks in advance, Dhiraj Bhuyan Security Research Engineer BT Exact Tel: +44 1473 643932 Mob: +44 7962 012145 Email: dhiraj.2.bhuyan(a)bt.com

21 years

RE: [Serusers] DHCP and SER

by Jiri Kuthan

At 03:13 AM 10/28/2003, Iizuka, Hirotaka, VF-JP wrote: >I'm sorry to reply back to you, I know you are busy. >How does SER query DNS? using DNS protocol. It takes DNS names from SIP messages and resolves them to IP addresses using DNS. >(Query with SIP message form?) >Where do I set the DNS's address in SER? What DNS address? If you mean the address at which SER is reachable, see Dan Austin's howto http://iptel.org/ser/doc/ser-howto/ser-Howto.html#AEN132 -jiri

21 years

RE: [Serusers] DHCP and SER

by Iizuka, Hirotaka, VF-JP

I'm sorry to reply back to you, I know you are busy. How does SER query DNS?(Query with SIP message form?) Where do I set the DNS's address in SER? -----Original Message----- From: Jan Janak [mailto:jan@iptel.org] Sent: Monday, October 27, 2003 4:20 PM To: Iizuka, Hirotaka, VF-JP Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] DHCP and SER Of course. Jan. On 27-10 10:04, Iizuka, Hirotaka, VF-JP wrote: > Hi, > > Does SER send query to the DNS to get SIP clients' address? Thank you. > > Hiro > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

21 years

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sr-users October 2003