sr-users January 2004

sr-users@lists.kamailio.org

142 participants
338 discussions

Re: [Serusers] SIP client and serweb
by CURRO_DOMINGUEZ 14 Jan '04

14 Jan '04

For example, from serweb user_interface I can add new contacts with their sip addresses But when I open my SIP client (I'm using Nortel SIP Client and Windows Messenger, not phones) I can't see my contacts. I have been watching the network whit tcpdump, and I can see the following messages: REGISTER from the client 200 OK from the server You are right, Jiri (and when not?). In 200 OK message, SIP Message Header, I can see several Contact Fields with my contacts. So, it should be that Windows Messenger can't understand this. I'll try another clients. Thanks very much. Curro ----- Mensaje Original ----- De: Jiri Kuthan <jiri(a)iptel.org> Fecha: Miercoles, Enero 14, 2004 4:27 pm Asunto: Re: [Serusers] SIP client and serweb > At 04:23 PM 1/14/2004, CURRO_DOMINGUEZ wrote: > >Hello everybody > > > >Well, we have been able to make a Voice call between two Nortel > SIP > >clients using SER. It's great! Thanks for your help. > > > >I have a question about user's contacts. In one hand, I can use > serweb > >to login with my username and add sip contacts or add phonebook > entries.>In other hand, I can use my client to add new contacts. > Moreover, the > >client can tell me if my contacts are online or offline. > > > >But the two ways don't display the same information, it means, > from > >serweb I add several sip address (I check that are recorded at ser > >database) but this addresses are not imported when I start the SIP > >client. > > > >I need my client import sip contacts ( in a very similar way like > MSN > >Messenger) from server. Is there any way to do this? Does the > client > >have to make a request to the server or is the server who has to > send > >this information when it receives a REGISTER? > > SER sends all registered contacts in replies to REGISTER. If you wish > your phone to display them, ask your phone's vendor. Look otherwise > in serweb. > > -jiri > > > >Or maybe SER is not intended to do this? > > > >Thank you very much for your help. > > > >Curro > > > > > >_______________________________________________ > >Serusers mailing list > >serusers(a)lists.iptel.org > >http://lists.iptel.org/mailman/listinfo/serusers > > -- > Jiri Kuthan http://iptel.org/~jiri/ > >

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14 Jan '04

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Re: [Serusers] Error: sorry --cannot open write fifo
by CURRO_DOMINGUEZ 14 Jan '04

14 Jan '04

Thanks Jiri, Jesus You were right, it was an issue about permissions. ----- Mensaje Original ----- De: Jiri Kuthan <jiri(a)iptel.org> Fecha: Miercoles, Enero 14, 2004 12:41 pm Asunto: Re: [Serusers] Error: sorry --cannot open write fifo > make sure that your web server has write permissions to access the > fifo. > At 11:45 AM 1/14/2004, CURRO_DOMINGUEZ wrote: > >Hello > > > >I continue with my test of SER. SER is running almost OK, but I > have > >some problems. Sorry if this is a newbie question, but is really > >important to me configure SER correctly. > > > >When I try to enter to serweb user_interface, I have this error: > > > >sorry --cannot open write fifo > > > >I can add, delete contacts with serweb, but I can't send Instant > >Messages(nor whit Windows Messenger). I have read fifo is the > >communication way between ser and serweb, and responsible for > instant > >messages. > > > >In config.php, the configuration is > >$this->fifo_server="/tmp/ser_fifo" > >And in ser.cfg the configuration is > >fifo="/tmp/ser_fifo" > > > >I think is OK, isn't it? > > > >What can be wrong? > > > >One more time, thank you very much for your time an knowledge. > > > >Curro > > > > > >_______________________________________________ > >Serusers mailing list > >serusers(a)lists.iptel.org > >http://lists.iptel.org/mailman/listinfo/serusers > > -- > Jiri Kuthan http://iptel.org/~jiri/ > >

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[Serusers] can SCS-Client work with SER?
by Kevin Chu 14 Jan '04

14 Jan '04

Hi, There is a SIP client from Siemens called SCS-Client: http://mysip.ch/index.html Has anybody ever tried to use the SCS-Client with SER? According to its specification, it can provide audio, video, presence and IM functions much similar to MS's Windows Messenger 5.0. Since the IM part of Messenger 5.0 can only work correctly with MS's Live Communications server, I was just trying to find another client with similar features that can work with SER. Thanks, Kevin

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[Serusers] Error: sorry --cannot open write fifo
by CURRO_DOMINGUEZ 14 Jan '04

14 Jan '04

Hello I continue with my test of SER. SER is running almost OK, but I have some problems. Sorry if this is a newbie question, but is really important to me configure SER correctly. When I try to enter to serweb user_interface, I have this error: sorry --cannot open write fifo I can add, delete contacts with serweb, but I can't send Instant Messages(nor whit Windows Messenger). I have read fifo is the communication way between ser and serweb, and responsible for instant messages. In config.php, the configuration is $this->fifo_server="/tmp/ser_fifo" And in ser.cfg the configuration is fifo="/tmp/ser_fifo" I think is OK, isn't it? What can be wrong? One more time, thank you very much for your time an knowledge. Curro

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[Serusers] SER forwinding to h323 gatekeeper
by Fábio Silvestri 14 Jan '04

14 Jan '04

Hi. Can I forward SER calls to a h323 gatekeeper, or how to forward SER calls to vocal sip server? -- |o |o |o Fabio Silvestri |o fabio(a)informatec.com.br |o ICQ: 1667351 |o

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Re: [Serusers] BYE missage is missed
by jerk face 13 Jan '04

13 Jan '04

--- Jiri Kuthan <jiri(a)iptel.org> wrote: > in front of the t_relay which forwards the > transaction you wish to report on. -jiri > The only place I have t_relay is the loose forward section. if (loose_route()) { t_relay(); break; }; Should I place setflag(3) before t_relay() in this section? Or should it be in the section in which the call is made? In my ser.cfg (entire file is at the bottom of this page) the call is made with the following lines: if (uri=~"sip:1[0-9]+@.*") { rewritehostport("sip.provider.net:5060"); setflag(2); if(!t_relay()) { sl_reply_error(); }; break; }; ::My ser.cfg :: # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode #debug=7 #fork=no #log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" alias=mysip.mydomain.com # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.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/acc.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("acc", "log_level", 2) modparam("acc", "db_url", "sql://ser:heslo@localhost/ser") modparam("acc", "db_flag", 2) # ------------------------- 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; }; # 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) # 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 (uri=~"sip:1[0-9]+@.*") { rewritehostport("switch.myprovider.net:5060"); setflag(2); if(!t_relay()) { sl_reply_error(); }; break; }; if (uri=~"sip:011[0-9]+@.*") { rewritehostport("switch.myprovider.net:5060"); setflag(2); if(!t_relay()) { sl_reply_error(); }; break; }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("mysip.mydomain.com", "subscriber")) { www_challenge("mysip.mydomain.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(); }; lookup("aliases"); save("aliases"); } ===== Asterisk is my lover, and IAX2 is our scented lubricant __________________________________ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/

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[Serusers] Forward calls to a second proxy
by Fábio Silvestri 13 Jan '04

13 Jan '04

Hi to all, I have on my ser.cfg many dial plans forwarding calls to several Cisco PSTN, now I need to configure, if the call could not find a specific dial plan the forward to a second SIP proxy! How can I do this? Regards. -- |o |o |o Fabio Silvestri |o fabio(a)informatec.com.br |o ICQ: 1667351 |o

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[Serusers] SER + B2BUA
by Edgar Barbosa 13 Jan '04

13 Jan '04

Hi guys, I need to setup a system, based on SER that can handle, for a particular class of users, a prepaid call scenario. I'm aware that I do need a B2BUA for this particular case. I've been trying to mount a scheme like this, routing some calls from SER to Vovida's B2BUA, but I'm stuck in the authentication process. I can't arrange a way of authenticating SER users on B2BUA... Has anyone tried a setup like this? Has anyone other suggestions of accomplishing this? I believe Maxim is developing a B2BUA, but I need a short term solution... Thanks, Edgar

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[Serusers] BYE missage is missed
by jerk face 13 Jan '04

13 Jan '04

I recently set up database accounting with SER and I was testing it out to see if there were any situations in which the logging might fail. I found that if I unplug my SIP device during a call, then a SIP BYE message will not be created in my database. Is there any configuration change I can make to fix this problem, or is this due to the design of the SIP protocol? Thank you for your time. Just in case it is a configuration problem, I have included my ser.cfg #ser.cfg ### # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode #debug=7 #fork=no #log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" alias=sip.mydomain.com # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.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/acc.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("acc", "log_level", 2) modparam("acc", "db_url", "sql://ser:heslo@localhost/ser") modparam("acc", "db_flag", 2) # ------------------------- 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; }; # 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) # 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()) { if (method == "BYE") { setflag(2); }; t_relay(); break; }; if (uri=~"sip:1[0-9]+@.*") { rewritehostport("sip.provider.net:5060"); setflag(2); if(!t_relay()) { sl_reply_error(); }; break; }; if (uri=~"sip:011[0-9]+@.*") { rewritehostport("sip.provider.net:5060"); setflag(2); if(!t_relay()) { sl_reply_error(); }; break; }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("sip.mydomain.com", "subscriber")) { www_challenge("sip.mydomain.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(); }; lookup("aliases"); save("aliases"); } __________________________________ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/

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sr-users January 2004