sr-users January 2004

sr-users@lists.kamailio.org

142 participants
338 discussions

[Serusers] append_hf before sl_send_reply?
by Ulrich Abend 22 Jan '04

22 Jan '04

Hi, could someone please give a statement on this: If you use append_hf before a sl_send_reply (code 302), the appended HF is not included in the message. Is this behaviour intented or is it a bug? Thanks, Uli. ---------- Forwarded Message ---------- Subject: Re: Sorry to bother you Date: Wednesday 21 January 2004 16:47 From: Marcello Lupo <lupo(a)itspecialist.it> To: Ulrich Abend <ullstar(a)iptel.org> Hio Ulrich, thank you for the answer. YEs you get the problem. For the original header field i was making a mistake, it only do not append addedd field (CC-Diversion) in this case. This is the answer of the SER with the Moved Temporarly: ***********************************************+ U 194.244.164.13:5060 -> 194.244.164.14:5060 SIP/2.0 302 Moved Temporarily..Via: SIP/2.0/UDP 194.244.164.14:5060..From: "anonymous" <sip:194.244.164.14>;tag=9B2BDCC 8-1BD2..To: <sip:0350545048@194.244.164.13>;tag=da45cd22c5a2c167d6e0afbf7b99da4a.4b46..Ca ll-ID: 62E20996-453F11D6-9D4FFB 9A-B058E7F8@194.244.164.14..CSeq: 101 INVITE..Contact: sip:390457156862@194.244.164.14:5060..Server: Sip EXpress router (0.8.11 (i386/linux))..Content-Length: 0..Warning: 392 194.244.164.13:5060 "Noisy feedback tells: pid=27841 req_src_ip= 194.244.164.14 req_src_port=50410 in_uri=sip:0350545048@194.244.164.13:5060 out_uri=sip:390457156862@194.244.164.14:5060 via_cnt==1".... ************************************************* YEs the CC-Diversion field is not appended even if i specify it in the configuration with append_urihf or append_hf . I uset this command syntax: append_hf("CC-Diversion:<sip:0350545048@390350545001.lea-d.net:5060>;reason=u nconditional\r\n"); When i use it in a new call it is appended, but with the sl_send_reply it is not. Thanks, Bye, MArcello On Wednesday 21 January 2004 16:09, you wrote: > Hi Marcello, > > On Tuesday 20 January 2004 22:13, Marcello Lupo wrote: > > Hi Ulrich, > > i hope you remember me... Marcello from Italy. > > Of course, I remember ;-) > > > today i wrote an e-mail on the mailing list but don't received any > > answer. > > You will receive an answer for shure, but sometimes it may take a little. > > A short look into the sources showed me, that the sl_send_reply function > _should_ include the original header fields for any code between 300 and > 400. From your mail I understand it does not, right? > > From your mail in serusers I understand that you only have the problem > left, that the CC-diversion field is not present, even if you specify > append-HF in the script? > > If that is the problem, you probably have to add the required functionality > in the sl module. Maybe this is also a bug. (~ line 175 in > modules/sl/sl_funcs.c) > > Please send your problem again, if I misunderstood you, I will take a > closer look at it tomorrow. > > Uli. > > > I'm crushing my head to find out a way to let SER to answer to a call > > with a 302 Moved temporarly and the request appended to it with the > > addendum of the CC-Diversion field so our gateway (cisco 3725) can > > correctly take the CC-Diversion field and put the number in the > > Redirecting Number in the ISDN setup. > > I tried to do it on a normal call but seems that the cisco ignore teh > > CC-DIversion field. > > From the documentatio i read it take in consideration the CC-Diversion > > only if it receive a 3xx answer. > > I tried to use the sl_send_reply function but it answer to cisco without > > appending the original request and any field i specify with append_hf or > > append_urihf. > > Can you help me? > > Thanks a lot, > > Bye, > > MArcello -------------------------------------------------------

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[Fwd: Re: [Serusers] append_hf before sl_send_reply?]
by Bogdan-Andrei IANCU 22 Jan '04

22 Jan '04

Hi Ulrich, append_hf() appends header to the current processed request and it will not be reflected on the replies; only in the forwarded request. If you want to add something to the reply, use append_to_reply() function. All changes introduced by this function will be reflected only on all replies for the current request. Regards, Bogdan Ulrich Abend wrote: >Hi, > >could someone please give a statement on this: > >If you use append_hf before a sl_send_reply (code 302), the appended HF is not >included in the message. Is this behaviour intented or is it a bug? > >Thanks, > >Uli. > >---------- Forwarded Message ---------- > >Subject: Re: Sorry to bother you >Date: Wednesday 21 January 2004 16:47 >From: Marcello Lupo >To: Ulrich Abend > >Hio Ulrich, >thank you for the answer. >YEs you get the problem. > >For the original header field i was making a mistake, it only do not append >addedd field (CC-Diversion) in this case. > >This is the answer of the SER with the Moved Temporarly: > >***********************************************+ > >U 194.244.164.13:5060 -> 194.244.164.14:5060 > SIP/2.0 302 Moved Temporarily..Via: SIP/2.0/UDP 194.244.164.14:5060..From: >"anonymous" ;tag=9B2BDCC > 8-1BD2..To: >;tag=da45cd22c5a2c167d6e0afbf7b99da4a.4b46..Ca >ll-ID: 62E20996-453F11D6-9D4FFB > 9A-B058E7F8@194.244.164.14..CSeq: 101 INVITE..Contact: >sip:390457156862@194.244.164.14:5060..Server: Sip EXpress router > (0.8.11 (i386/linux))..Content-Length: 0..Warning: 392 194.244.164.13:5060 >"Noisy feedback tells: pid=27841 req_src_ip= > 194.244.164.14 req_src_port=50410 in_uri=sip:0350545048@194.244.164.13:5060 >out_uri=sip:390457156862@194.244.164.14:5060 > via_cnt==1".... >************************************************* > >YEs the CC-Diversion field is not appended even if i specify it in the >configuration with append_urihf or append_hf . >I uset this command syntax: > >append_hf("CC-Diversion:;reason=u >nconditional\r\n"); > >When i use it in a new call it is appended, but with the sl_send_reply it is >not. > >Thanks, >Bye, >MArcello > >On Wednesday 21 January 2004 16:09, you wrote: > > >>Hi Marcello, >> >>On Tuesday 20 January 2004 22:13, Marcello Lupo wrote: >> >> >>>Hi Ulrich, >>>i hope you remember me... Marcello from Italy. >>> >>> >>Of course, I remember ;-) >> >> >> >>>today i wrote an e-mail on the mailing list but don't received any >>>answer. >>> >>> >>You will receive an answer for shure, but sometimes it may take a little. >> >>A short look into the sources showed me, that the sl_send_reply function >>_should_ include the original header fields for any code between 300 and >>400. From your mail I understand it does not, right? >> >>From your mail in serusers I understand that you only have the problem >>left, that the CC-diversion field is not present, even if you specify >>append-HF in the script? >> >>If that is the problem, you probably have to add the required functionality >>in the sl module. Maybe this is also a bug. (~ line 175 in >>modules/sl/sl_funcs.c) >> >>Please send your problem again, if I misunderstood you, I will take a >>closer look at it tomorrow. >> >>Uli. >> >> >> >>>I'm crushing my head to find out a way to let SER to answer to a call >>>with a 302 Moved temporarly and the request appended to it with the >>>addendum of the CC-Diversion field so our gateway (cisco 3725) can >>>correctly take the CC-Diversion field and put the number in the >>>Redirecting Number in the ISDN setup. >>>I tried to do it on a normal call but seems that the cisco ignore teh >>>CC-DIversion field. >>>From the documentatio i read it take in consideration the CC-Diversion >>>only if it receive a 3xx answer. >>>I tried to use the sl_send_reply function but it answer to cisco without >>>appending the original request and any field i specify with append_hf or >>>append_urihf. >>>Can you help me? >>>Thanks a lot, >>>Bye, >>>MArcello >>> >>> > >------------------------------------------------------- > >_______________________________________________ >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

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Re: [Serusers] SEMS and g.729
by steve 22 Jan '04

22 Jan '04

Jan, Thank you for the offer. I will ask the people responsible for my funding if that can be done. They are talking to their carrier now to see if he is willing to add g711 support for us. Steve -----Original Message----- From: Jan Janak <jan(a)iptel.org> Date: Thu, 22 Jan 2004 01:17:53 To:"Steven R. Bunin" <steve(a)solaas.com> Cc:serusers@lists.iptel.org Subject: Re: [Serusers] SEMS and g.729 Let's split the work -- you pay the license, we intergrate it. Jan. On 21-01 16:42, Steven R. Bunin wrote: > Hi Guys, > > I have a number or users who can only do g.729 for a number of reasons > but want voicemail. Has anyone integrated SEMS with a g.729 codec to > record messages for missed incomming calls. If not does anyone have any > suggestions on how to go about integrating g.729 into SEMS? > > Thank you all in advance. > > Steve > > -- > Steven R. Bunin - Managing Partner > > SOLAAS LLC & Paketalk LLC > 212 W35th Street > 2nd Floor > New York, NY 10001 > (+001) 212-244-8300 > Cellular: 646-739-7000 > > http://www.solaas.com > > -- > > This e-mail may contain confidential and/or privileged information. If > you are not the intended recipient (or have received this e-mail in > error) please notify the sender immediately and destroy this e-mail. Any > unauthorized copying, disclosure or distribution of the material in this > e-mail is strictly forbidden. > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers -- Steven R. Bunin - Managing Partner SOLAAS LLC & Paketalk LLC 212 W35th Street 2nd Floor New York, NY 10001 (+001) 212-244-8300 Cellular 646-739-7000 http://www.solaas.com

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[Serusers] ispbs questions
by Arnd Vehling 22 Jan '04

22 Jan '04

Hello, i kinda installed ispbs and have a couple of questions: - most programm functions are not in the menu, i need to start them manually (i.e. look at the scripts and type full path name into URL) - I dont get how ispbs gets the accunting data from ser mysql database. Or does isps on work if i use radius accounting with ser? regards, Arnd

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[Serusers] Please help with PSTN
by Andy Singh 22 Jan '04

22 Jan '04

All, Please take a look at my ser.cfg . My sip domain is sipserv.foo.com. I can dial 3603(a)foo.com <mailto:3603@foo.com> which is a PSTN number off of a PBX but can't dial 3603(a)sipserv.foo.com <mailto:3603@sipserv.foo.com> . I get 404 not found. What can I change in my config file so I can dial 3603(a)sipserv.foo.com <mailto:3603@sipserv.foo.com> . # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=yes # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/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" # 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") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("sipserv.foo.com", "subscriber")) { www_challenge("sipserv.foo.com", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; attempt handoff to PSTN if (uri=~"^sip:3[0-9]*") { ## This assumes that the caller is log("Forwarding to PSTN\n"); ## registered in our realm forward( 156.151.96.253, 5060 ); ## Our Cisco router break; }; } Please help Thanks

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[Serusers] please HELP CISCO ATA not re-registers
by Humberto Atristain 22 Jan '04

22 Jan '04

Dear Pals. I have a 10 point VPN in the central I have a SER working fine , and we have CISCO ATA´s in the branches and also in the central site, my trouble is that if for some reason the vpn comes down , when the VPN comes up again , the SER do not registers the ATAs , got no idea what to do here, take in note I´m newbie and please be a little explicit. I have tried with in ATA SIPREG: 1 SIPRegInterval: 360 Please Help Regards Humberto Atristain

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[Serusers] SEMS and g.729
by Steven R. Bunin 22 Jan '04

22 Jan '04

Hi Guys, I have a number or users who can only do g.729 for a number of reasons but want voicemail. Has anyone integrated SEMS with a g.729 codec to record messages for missed incomming calls. If not does anyone have any suggestions on how to go about integrating g.729 into SEMS? Thank you all in advance. Steve -- Steven R. Bunin - Managing Partner SOLAAS LLC & Paketalk LLC 212 W35th Street 2nd Floor New York, NY 10001 (+001) 212-244-8300 Cellular: 646-739-7000 http://www.solaas.com -- This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden.

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[Serusers] voicemail config
by Srbislav Cvetkovic 22 Jan '04

22 Jan '04

Hi to all, I am new to ser and sems. I was able to setup ser on openbsd and make a call from one user agent to another. Then I installed sems and after some playing around i was able to make it work with ser. However now the call is being directed to my voicemail system and the phone does not ring at all. Is there a setting that will allow me to set up the number of rings before it gets redirected to my voicemail? Below are my ser.cfg and sems.conf files. I have mysql version of ser working. Also there is one problem. I have no log files for some reason. From the documentation that I 've read it says that it should be in system log file. I have no system log file in /var/log . Isn't that strange? I successfuly receive all of my voice messages though. ////////////////////// ser.cfg 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 (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 loose_route(); # Make MSN Messenger happy... if (method=="REGISTER") { sl_send_reply("200","ok"); break; }; if (uri == myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("myhost.com", "subscriber")) { www_challenge("myhost.com", "0"); break; }; save("location"); break; }; /* # if i comment this out then the call is being redirected to my voicemail # if it's not commented out, then I get 404 User Not Found error while dialing # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; */ # Voicemail specific configuration - begin if(method=="ACK" || method=="INVITE" || method=="BYE"){ if(t_newtran()){ t_reply("100","Trying -- just wait a minute !"); if(method=="INVITE"){ log("**************** vm start - begin * *****************\n"); if(!vm("/tmp/am_fifo","voicemail")){ log("could not contact the answe r machine\n"); t_reply("500","could not contact the answer machine"); }; log("**************** vm start - end *** ***************\n"); break; }; if(method=="BYE"){ log("**************** vm end - begin *** ***************\n"); if(!vm("/tmp/am_fifo","bye")){ log("could not contact the answe r machine\n"); t_reply("500","could not contact the answer machine"); }; log("**************** vm end - end ***** *************\n"); break; }; } else { log("could not create new transaction\n"); sl_send_reply("500","could not create new transacti on"); }; }; # Voicemail specific configuration - end }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } //////////////////////// sems.conf # # 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=yes # optional parameter: stderr={yes|no} # # - debug mode: do not fork and log to stderr stderr=no # optional parameter: loglevel={0|1|2|3} # # - sets log level (error=0, warning=1, info=2, debug=3) loglevel=1 # optional parameter: fifo_name=<filename> # # - path and file name of our fifo file fifo_name=/tmp/am_fifo # optional parameter: ser_fifo_name=<filename> # # - path and file name of Ser's fifo file 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/ ################################## # voicemail specific parameters # ################################## # optional parameter: announce_path=<path> # # - sets the path where announce files are searched for announce_path=/usr/local/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=localhost # optional parameter: smtp_port=<port> # # - sets port of smtp server smtp_port=25 Thanks, Srbo Cvetkovic | CityNet, Inc. srbo(a)city-net.com | Pittsburgh, PA voice: 412.481.5406 | fax: 412.431.1315

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[Serusers] ser/radius accounting not happening
by Madan 21 Jan '04

21 Jan '04

Hi List, I 'am facing a problem with ser/radius configuration which is working fine except for the accounting part. In freeradius accounting details I see that both start and stop time is same, can somebody help me on how to fix this. here is my ser.cfg # # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) listen=10.0.0.1 /* 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" # ------------------ 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/uri.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/group_radius.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 2) modparam("auth_radius", "radius_config","/usr/local/etc/radiusclient/radiusclient.conf") #modparam("auth_radius", "service_type", 15) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("acc", "log_level", 1) modparam("acc", "radius_flag", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (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 # t_relay(); # break; # }; if (loose_route()) { if (method=="BYE" || method=="CANCEL") { log (1, "BYE or CANCEL\n"); setflag(1); }; 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 (method=="REGISTER") { log(1, "REGISTER: Authenticating user\n"); if (!radius_www_authorize("")) { log(1, "REGISTER: challenging user\n"); www_challenge("", "0"); break; }; save("location"); break; }; ########################################################################################################## if (uri=~"^sip:44[127][0-9]+@.*$") { if (rewritehostport("10.0.0.8:5060")) { if (!t_relay()) { sl_reply_error(); }; } else { sl_reply_error (); }; break; } else if (method=="INVITE") { sl_send_reply("403", "Call cannot be served here"); }; # account completed transactions via syslog setflag(1); ############################################################################################################### if (method=="INVITE") { log(1, "INVITE\n"); setflag(1); /* set for accounting (the same value as in log_flag!) */ }; if (method=="MESSAGE") { log(1, "MESSAGE\n"); setflag(1); /* set for accounting (the same value as in log_flag!) */ }; if (method=="BYE" || method=="CANCEL") { log (1, "BYE or CANCEL\n"); setflag(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; if (!t_relay()) { sl_reply_error(); break; }; }

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[Serusers] sip323 source anyone?
by Arnd Vehling 21 Jan '04

21 Jan '04

Hello, while doing a search for open source SIP<>H323 proxies on the net i ran over a software called sip323 from the columbia unviversity. Unluckily it seems that this software has been taken offline. If anyone of you has a copy of it around i would greatly appreciate a copy of the source. Only if the software is open source, that is. best regards, Arnd

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sr-users January 2004