sr-users January 2004

sr-users@lists.kamailio.org

142 participants
338 discussions

[Serusers] multiple phone rings
by Srbislav Cvetkovic 26 Jan '04

26 Jan '04

Hi guys, thanks to you all I have my voicemail up and running now. But as people are greedy and always want more, i guess i have another question. Is it possible to have multiple sip phones ring at the same time and who ever answers first gets the call? It's for our technical support group. I have a call coming from PSTN onto a VEGA gateway that does all the sip conversion and talks to our ser server. thanks. Srbo Cvetkovic | CityNet, Inc. srbo(a)city-net.com | Pittsburgh, PA voice: 412.481.5406 | fax: 412.431.1315

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SV: [Serusers] Re: Peering SER and Cisco Call Manager
by Kerker Staffan 26 Jan '04

26 Jan '04

Hi I guess you have to wait a few more months. Next version of Cisco Call Manager is supposed to support SIP on the "outside" I understand. Or maybe use Asterisk as a H.323/SIP gateway? Anyone tried? rgds, /Staffan Kerker Sweden -----Ursprungligt meddelande----- Från: Jiri Kuthan [mailto:jiri@iptel.org] Skickat: den 26 januari 2004 10:55 Till: Raymond May; serusers(a)lists.iptel.org Ämne: Re: [Serusers] Re: Peering SER and Cisco Call Manager Cisco Call Manager does not speak SIP. SIP interconnection to a call manager does not work. -jiri At 07:25 PM 1/23/2004, Raymond May wrote: >Hi SER Gurus, > My SER server has been running just >perfect for the last 9 months and I am now trying to >connect my private VoIP network to my Employer's. They >have a Cisco Call Manager and I am trying to figure >out how to configure SER so I can make calls from my >private network to users that are registered to the >Corporate ( Cisco Call Manager ). Ideally I would like >to be able to make calls from my private network to >the users on the Corporate and have users on the >Corporate be able to reach users on my VoIp Network ( >SER Server ). Any suggestions would be appreciated. >Below is current connectivity between my private >network and our corporate network. > >SER Users --- SER Server --- Internet --- Cisco Call >Manager --- Corporate Users > > >Brgds, >Ray May > >__________________________________ >Do you Yahoo!? >Yahoo! SiteBuilder - Free web site building tool. Try it! >http://webhosting.yahoo.com/ps/sb/ > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/ _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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RE: [Serusers] nathelper
by Jiri Kuthan 26 Jan '04

26 Jan '04

Then make sure you are using the latest CVS stable branch. (see www.iptel.org/ser/cvs) I'm not sure that's going to solve your problem but as a matter of fact, older versions don't support NAT traversal extensively. -jiri At 05:20 PM 1/23/2004, Federico R. SANCHEZ Y NAPAL wrote: >Hi, I'm using the rpm packge: > >ser-0.8.12-0.i386.rpm on Linux RedHat 9.0 > >I tried to use a source distribution (ser-0.8.12_src.tar.gz), and got >the same result. > >Regards, >-fs > >-----Mensaje original----- >De: Jiri Kuthan [mailto:jiri@iptel.org] >Enviado el: viernes, 23 de enero de 2004 11:24 >Para: Federico R. SANCHEZ Y NAPAL; serusers(a)lists.iptel.org >Asunto: Re: [Serusers] nathelper > >What SER version are you using? (Unless you use stable CVS branch, >that's >a very likely explanation for the problem you experience.) > >-jiri > >At 08:19 PM 1/22/2004, Federico R. SANCHEZ Y NAPAL wrote: >>Hello: I'm using a NATed UA to terminate a call in the PSTN using a >Cisco 5300. I want to use the RTP relay capability. In the script, I >wrote: >> >> ... >> force_rtp_proxy(); >> rewritehost("200.60.XXX.XXX"); >> forward("200.60.XXX.XXX", 5060); >> ... >> >>Where 200.60.XXX.XXX is the GTW IP. >> >>I get an error in the log: >> "/usr/sbin/ser[7039]: ERROR: extract_mediaip: no 'c=' in SDP" >> >>The SIP Message sent to de GTW has a wrong content-length header, so >the GTW rejects it. Here I copy the message: >> >>INVITE sip:2345674284618@200.60.30.24 SIP/2.0 >>Record-Route: <sip:2345674284618@200.32.43.55;ftag=673501644;lr=on> >>Via: SIP/2.0/UDP 200.32.43.55;branch=0 >>Via: SIP/2.0/UDP >200.68.54.46:5060;rport=25127;branch=z9hG4bKFE5D22D6B6C4407884 >>2396149F64C3DE >>From: 1001 <sip:1001@200.32.43.55>;tag=673501644 >>To: <sip:2345674284618@200.32.43.55> >>Contact: <sip:1001@200.68.54.46:25127> >>Call-ID: 107D459F-7402-480D-988A-FC87455E1E5D(a)192.168.0.15 >>CSeq: 31862 INVITE >>Max-Forwards: 69 >>Content-Type: application/sdp >>voip-gtw04# >>User-Agent: X-Lite build 1101 >>Content-Length: 315296 >>v=0 >>o=1001 23178458 23178458 IN IP4 200.68.54.46 >>s=X-Lite >>c=IN IP4 200.32.43.55 >>t=0 0 >>m=audio 35052 RTP/AVP 0 8 3 98 97 101 >>a=rtpmap:0 pcmu/8000 >>a=rtpmap:8 pcma/8000 >>a=rtpmap:3 gsm/8000 >>a=rtpmap:98 iLBC/8000 >>a=rtpmap:97 speex/8000 >>a=rtpmap:101 telephone-event/8000 >>a=fmtp:101 0-15 >>a=direction:active >>I would appreciate any help about this. >>Best Regards, >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers > >-- >Jiri Kuthan http://iptel.org/~jiri/ -- Jiri Kuthan http://iptel.org/~jiri/

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Re: RE: [Serusers] NEW USER...
by CURRO_DOMINGUEZ 26 Jan '04

26 Jan '04

Hello Sebastian, As Klaus says, ask your questions. One month ago, I started to configure SER, and this list have helped me a lot. So, I'll be delighted to help you with SER as much as I can. Bye Curro ----- Mensaje Original ----- De: "Klaus Darilion" <darilion(a)ict.tuwien.ac.at> Fecha: Lunes, Enero 26, 2004 1:10 am Asunto: RE: [Serusers] NEW USER... > hi sebastion! > > ask your questions to the mailing list, than you get the help of > several users und other may benefit too. > > regards, > klaus > > > -----Original Message----- > From: Sebastian Nocetti [mailto:snocetti@fibertel.com.ar] > Sent: Fri 23.01.2004 18:58 > To: serusers(a)lists.iptel.org > Cc: > Subject: [Serusers] NEW USER... > Hi everybody, I am new on this of SER and I am really confused... > I want > to know if someone can contact me by MSN (gnocetti(a)hotmail.com) > and give > me some clues.... to configure and make it works. Really I tried a lot > but never succesfully, please if someone can introduce me to this > excelent tool, becaus I want to implement SER like a PROXY SIP and > ROUTING... for Carrier.... > > > TX > > Sebastian. > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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RE: [Serusers] NEW USER...
by Klaus Darilion 26 Jan '04

26 Jan '04

hi sebastion! ask your questions to the mailing list, than you get the help of several users und other may benefit too. regards, klaus -----Original Message----- From: Sebastian Nocetti [mailto:snocetti@fibertel.com.ar] Sent: Fri 23.01.2004 18:58 To: serusers(a)lists.iptel.org Cc: Subject: [Serusers] NEW USER... Hi everybody, I am new on this of SER and I am really confused... I want to know if someone can contact me by MSN (gnocetti(a)hotmail.com) and give me some clues.... to configure and make it works. Really I tried a lot but never succesfully, please if someone can introduce me to this excelent tool, becaus I want to implement SER like a PROXY SIP and ROUTING... for Carrier.... TX Sebastian.

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RE: [Serusers] multiple phone rings
by Klaus Darilion 26 Jan '04

26 Jan '04

yes, there is a limit. its called "max branches" or similar and can be configured before compilation in one of the config files (search the archive). the maximum is around 30. klaus -----Original Message----- From: Arnd Vehling [mailto:av@nethead.de] Sent: Fri 23.01.2004 17:47 To: Jiri Kuthan Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] multiple phone rings Jiri Kuthan wrote: > > register multiple phones with the same address, that's it. -jiri Yupp, very nice feature. Is there a limit to the number of how many sip-clients can subscribe with the same sip-address? -- Arnd _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] ser with Windowsce.net client
by Annie Sasidar 25 Jan '04

25 Jan '04

Hi, I just want to know if anyone has a windowsce.net client application working with ser-0.8.12? Please let me know. Thanks, Annie

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[Serusers] sems + Freebsd
by Atle Samuelsen 25 Jan '04

25 Jan '04

hey. I downloaded the CVS version of sems.. and Im running FreeBSD (uname -a ) FreeBSD voip4.7-RELEASE FreeBSD 4.7-RELEASE #0: Wed Oct 9 15:08:34 GMT 2002 root@builder.freebsdmall.com:/usr/obj/usr/src/sys/GENERIC I compile sems with gmake I get this : AmRequest.cpp: In method `void AmRequestUAC::execute()': AmRequest.cpp:568: `req' undeclared (first use this function) AmRequest.cpp:568: (Each undeclared identifier is reported only once AmRequest.cpp:568: for each function it appears in.) gmake[1]: *** [AmRequest.o] Error 1 gmake[1]: Leaving directory `/root/sems/answer_machine' gmake: [all] Error 2 (ignored) anybody know why this is happening? I need to use the CVS version of sems , because Im running SER CVS version. - Atle

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Re: [Serusers] VoiceMail
by Raphael Coeffic 25 Jan '04

25 Jan '04

Sorry for this, i tought you had already the lastest patchs. Just change the module name back to 'voicemail' and verify your database connection string. You should also verify the access permission within mysql. -Raphael. PS: don't forget to CC the list... ----- Original Message ----- From: "Atle Samuelsen" <clona(a)cyberhouse.no> To: "Raphael Coeffic" <rco(a)iptel.org> Sent: Sunday, January 25, 2004 1:50 PM Subject: Re: [Serusers] VoiceMail > > Now I got : > Jan 25 12:49:22 voip ser: set_mod_param_regex: No module matching vm > found | > Jan 25 12:49:22 voip ser: parse error (54,63-64): Can't set module > parameter > > > .. this is my version : > > voip# ser -V > version: ser 0.8.12 (i386/freebsd) > flags: STATS:Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, DNS_IP_HACK, > SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > @(#) $Id: main.c,v 1.168 2003/10/12 15:09:08 andrei Exp $ > main.c compiled on 13:19:53 Nov 21 2003 with gcc 2.95 > > > > - Atle: > > * Raphael Coeffic <rco(a)iptel.org> [040125 13:35]: > > Hello, > > > > the module name has been changed from 'voicemail' to 'vm'. > > > > That means that you should change following line: > > > modparam("voicemail", "db_url","mysql://ser:heslo@localhost:3306/ser") > > to: > > modparam("vm", "db_url","mysql://ser:heslo@localhost:3306/ser") > > > > -Raphael. > > > > ----- Original Message ----- > > From: "Atle Samuelsen" <clona(a)cyberhouse.no> > > To: <serusers(a)lists.iptel.org> > > Sent: Sunday, January 25, 2004 12:34 PM > > Subject: [Serusers] VoiceMail > > > > > > > > > > Hey guys. > > > Im trying to configure the voicemail module. tho im haveing some weird > > > problems.. > > > > > > Im getting these errors: > > > > > > Jan 25 11:27:17 voip ser[97088]: ERROR: vm_mod_init: unable to bind db > > > Jan 25 11:27:17 voip ser[97088]: init_mod(): Error while initializing > > > module voicemail > > > > > > Tanks for the help > > > > > > - Atle > > > > > > Here's my config file.. > > > > > > # $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) > > > port=5060 > > > children=4 > > > fifo="/tmp/ser_fifo" > > > > > > alias="voip.domain.com" > > > > > > # ------------------ module loading ---------------------------------- > > > > > > > > > loadmodule "/usr/local/lib/ser/modules/nathelper.so" > > > loadmodule "/usr/local/lib/ser/modules/textops.so" > > > loadmodule "/usr/local/lib/ser/modules/vm.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" > > > > > > > > > # ----------------- setting module-specific parameters --------------- > > > # -- usrloc params -- > > > # modparam("usrloc", "db_mode", 0) > > > > > > > > > # -- rr params -- > > > # add value to ;lr param to make some broken UAs happy > > > modparam("rr", "enable_full_lr", 1) > > > > > > # ----------------- setting module-specific parameters --------------- > > > > > > modparam("voicemail", "db_url","mysql://ser:heslo@localhost:3306/ser") > > > > > > > > > # ------------------------- 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 > > > 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") { > > > save("location"); > > > break; > > > }; > > > > > > # native SIP destinations are handled using our USRLOC DB > > > if (!lookup("location")) { > > > sl_send_reply("404", "Not Found"); > > > break; > > > }; > > > }; > > > > > > #inserted by klaus > > > if (method=="INVITE") { > > > record_route(); > > > force_rtp_proxy(); > > > /* set up reply processing */ > > > t_on_reply("1"); > > > }; > > > > > > > > > # 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(); > > > }; > > > > > > > > > > > > # 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 answer 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 answer 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 transaction"); > > > }; > > > }; > > > > > > # Voicemail specific configuration - end > > > } > > > > > > > > > #inserted by klaus > > > # all incoming replies for t_onrepli-ed transactions enter here > > > onreply_route[1] { > > > if (status=~"[12][0-9][0-9]") > > > force_rtp_proxy(); > > > > > > } > > > > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > >

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[Serusers] VoiceMail
by Atle Samuelsen 25 Jan '04

25 Jan '04

Hey guys. Im trying to configure the voicemail module. tho im haveing some weird problems.. Im getting these errors: Jan 25 11:27:17 voip ser[97088]: ERROR: vm_mod_init: unable to bind db Jan 25 11:27:17 voip ser[97088]: init_mod(): Error while initializing module voicemail Tanks for the help - Atle Here's my config file.. # $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) port=5060 children=4 fifo="/tmp/ser_fifo" alias="voip.domain.com" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/vm.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" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- # modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ----------------- setting module-specific parameters --------------- modparam("voicemail", "db_url","mysql://ser:heslo@localhost:3306/ser") # ------------------------- 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 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") { save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; #inserted by klaus if (method=="INVITE") { record_route(); force_rtp_proxy(); /* set up reply processing */ t_on_reply("1"); }; # 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(); }; # 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 answer 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 answer 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 transaction"); }; }; # Voicemail specific configuration - end } #inserted by klaus # all incoming replies for t_onrepli-ed transactions enter here onreply_route[1] { if (status=~"[12][0-9][0-9]") force_rtp_proxy(); }

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sr-users January 2004