sr-users March 2004

sr-users@lists.kamailio.org

145 participants
286 discussions

RE: [Serusers] t_relay_to not found
by Jiri Kuthan 16 Mar '04

16 Mar '04

oh, thank you -- I was looking at wrong place. Indeed, the documentation did not keep the pace with development. t_relay was split in t_relay_to_udp and t_relay_to_tcp when tcp was introduced. I updated CVS documentation. I will update the documentation. Thanks for reporting. -jiri At 01:08 PM 3/16/2004, Stephen James wrote: >This would be the one, reached through the main ser webpage >clicking on ser modules and then html version. > >http://iptel.org/ser/doc/modules/html/tm.html > > >-----Original Message----- >From: Jiri Kuthan [mailto:jiri@iptel.org] >Sent: Monday, March 15, 2004 7:20 PM >To: Stephen James >Subject: RE: [Serusers] t_relay_to not found > > >which tm.html document are you referring to? -jiri > >At 02:04 AM 3/16/2004, Stephen James wrote: >>Hmm. This code segement was cut straight from the tm.html document? >>Is this a documentation problem or a deprecated function? >> >>-----Original Message----- >>From: Jiri Kuthan [mailto:jiri@iptel.org] >>Sent: Monday, March 15, 2004 6:50 PM >>To: Stephen James; serusers(a)lists.iptel.org >>Subject: Re: [Serusers] t_relay_to not found >> >> >>There is no such a command in SER. There is t_relay_to_udp though. >>-jiri >> >>At 04:44 PM 3/15/2004, Stephen James wrote: >>>I attempted to implement the code in the example for PSTN gateway >>>where >> >>>after matching the enum in the user portion of the uri it uses >>>t_relay_to. When I >>>implement this I get unable to find t_relay_to in the startup of ser. >>It >>>find t_relay so I know that the tm module is loaded and exists, when I > >>>do nm on the module I see the symbol t_relay_to in there as well??? >>> >>># attempt handoff to PSTN >>>if (uri=~"^sip:9[0-9]*@.*") { >>> log("Forwarding to PSTN\n"); >>> t_relay_to( "192.168.0.2", "5060"); >>> break; >>>}; >>> >>> >>>Carrius Technologies >>>Stephen James >>>Principal MTS >>>sjames(a)carriustech.com >>>2140 Lake Park Blvd. >>>Suite 500 >>>Richardson TX, 75080 >>> >>>Signature powered by Plaxo ... Want a signature like this? >>> >>>_______________________________________________ >>>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/ -- Jiri Kuthan http://iptel.org/~jiri/

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Re: [Serusers] SER and rtpproxy
by Klaus Darilion 16 Mar '04

16 Mar '04

Hi Fabio! Just replace your condition in the reply route with my condition. No, you don't need to any special commands for the rtpproxy. Yes I have sample cfg - i already postet is ome time ago - search the archive. regards, klaus PS: cc to the list Fabio Silvestri wrote: > Well, where do I need to include that code if { } on my ser.cfg? > > Do you have a sample cfg? > > How do I need to run rtpproxy, have some special command line? > > On Mon, 1 Mar 2004, Klaus Darilion wrote: > > >>Date: Mon, 01 Mar 2004 09:35:12 +0100 >>From: Klaus Darilion <klaus.mailinglists(a)pernau.at> >>To: Fábio Silvestri <fabio(a)informatec.com.br> >>Cc: serusers(a)lists.iptel.org >>Subject: Re: [Serusers] SER and rtpproxy >> >> >> >>Fábio Silvestri wrote: >> >> >>>Hi! >>> >>>After a long search on maillinst, I'm trying to run SER with rtpproxy, >>>for resolve problems with SIP UA (ata186) behind a nat, but I'm still >>>get in trouble. >>> >>>I have setup ser.cfg to use rtpproxy, like this: >>> >>>... >>> if (method=="INVITE") { >>> log(1,"INVITE\n"); >>> fix_nated_sdp("1"); >>> record_route(); >>> force_rport(); >>> setflag(1); >>> force_rtp_proxy(); >>> t_on_reply("1"); >>> } >>>... >>>onreply_route[1] { >>> log(1, ">=====> reply 1\n"); >>> if ((status=~"[12][0-9][0-9]")) { >>> force_rtp_proxy(); >>> } >>>} >>>... >>> >>>But I get this error on syslog: >>> >>> 0(5463) ERROR: extract_mediaip: no `c=' in SDP >>> >> >>This will happen when you try to rewrite a response without sdp, e.g. >>100 trying, 180 ringing.... >> >>I'm unsing: >> if ( (status=~"200" || status=~"183") ) { >> force_rtp_proxy(); >> >> >>>I'm running rtpproxy, on linux shell without any parameters. >>> >>>Does anybody knows what this means that message, and how to solve, a >>>ser.cfg for rtpproxy, and others tips will be helpfull... >>> >>>Regards. >>> >> >

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[Serusers] SER & freeradius !
by Anton 16 Mar '04

16 Mar '04

Hello List, Please help me! I've installed Ser & followed the RADIUS Howto in ser website. Every thing seems to run, but radius is idle all the time, SIPPS cannot registre with SER. Here is my ser.cfg file What I'm wrong ? Any info is appreciated, Best regards, Anton =============================ser.cfg file =============== debug=7 fork=yes log_stderror=no 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" #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_radius.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/auth_radius.so" #loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" modparam("usrloc", "db_mode", 0) #modparam("uri_radius", "service_type", 11) modparam("rr", "enable_full_lr", 1) modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") #modparam("acc", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") #modparam("xlog", "buf_size", 8192) ### #modparam("acc", "log_level", 1) #modparam("acc", "log_flag", 1) #modparam("acc", "radius_flag", 1) modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "fr_timer", 10 ) modparam("auth", "secret", "testing123") modparam("auth_radius", "service_type", 15) ### route{ 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; # }; record_route(); if (loose_route()) { t_relay(); break; }; if (uri=~"fptnet.vn") { xlog("L_ERR", " method <%rm> <%fu> <%tu>\n"); if (method=="REGISTER") { # if (!www_authorize("mydomain", "subscriber")) { if (!radius_www_authorize("fptnet.com.vn")) { www_challenge("fptnet.com.vn", "0"); break; }; save("location"); break; }; setflag(1); if (method=="INVITE") record_route(); if (method=="REGISTER") { log(1, "REGISTER received\n"); } else { log(1, "non-REGISTER received\n"); }; if (uri=~"sip:.*[@:]fptnet.vn") { log(1, "request for fptnet.vn\n"); } else { log(1, "request for other domain received\n"); }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; #}; if (!t_relay()) { sl_reply_error(); }; }; }

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[Serusers] t_relay_to not found
by Stephen James 16 Mar '04

16 Mar '04

I attempted to implement the code in the example for PSTN gateway where after matching the enum in the user portion of the uri it uses t_relay_to. When I implement this I get unable to find t_relay_to in the startup of ser. It find t_relay so I know that the tm module is loaded and exists, when I do nm on the module I see the symbol t_relay_to in there as well??? # attempt handoff to PSTN if (uri=~"^sip:9[0-9]*@.*") { log("Forwarding to PSTN\n"); t_relay_to( "192.168.0.2", "5060"); break; }; Carrius Technologies Stephen James Principal MTS sjames(a)carriustech.com 2140 Lake Park Blvd. Suite 500 Richardson TX, 75080 Signature powered by Plaxo ... Want a signature like this?

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[Serusers] Questions about sems/answer_machine IVR plugin
by Arun Dharankar 15 Mar '04

15 Mar '04

Greetings... From the CVS repositories I picked up ser and sems. Nicely done! Tried to execute some scenarios and ser/sems seem to work good. Also tried the sems' IVR feature using Python. 1. The playAndDetect method/function seems to play the announcement but the return value seems to be invalid. The return values are some very large numbers when I print them. What could I possibly being doing wrong? The calling phone is Linux based softphone from Zultsys (and it seems to work quite well). 2. The timeout argument to playAndDetect does not seem to have any effect - it seems to return almost immediately. What could I possibly be doing wrong? Best regards, -Arun. -- Email: ArunD(a)Users.SourceForge.Net GnuPG: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x36E0C61D

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Re: [Serusers] ENUM anyone?
by Jadylson Bomfim 15 Mar '04

15 Mar '04

Hi all. How can I start to provides Numbering Services into my own network ? What tool set I need to ? Thank You all Jadylson da Rocha Passos Bomfim Redevox Telecom Uberlandia +55 34 3234-7813 São Paulo +55 11 5055-6888 Móvel +55 34 9103-6854 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.618 / Virus Database: 397 - Release Date: 9/3/2004

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[Serusers] How to handle ACK after stateless replies?
by Klaus Darilion 15 Mar '04

15 Mar '04

Hi! If the proxy replies stateless, how should the proxy handle the incoming ACK request? e.g. an INVITE to an offline user: if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; } The SIP client will respond with ACK to the 404 response. How shall I handle this ACK - just ignoring it? BTW: Is it allowed to send stateless replies on existing transactions? e.g: t_newtran(); ... sl_send_reply(..); thanks, Klaus

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[Serusers] prohibit gateway access - question
by Klaus Darilion 15 Mar '04

15 Mar '04

Hi! I've got a small problem and don't know how to solve it best, so I would appreciate your comments: I want to allow my ser users to call any IP destination for free and always, but I want to restrict access to the PSTN. Therefore, I authenticate, account and check if user is in PSTN group before forwarding to the gateway (local GW or the GW of an PSTN termination provider). But, for example if one of my users call 1234567(a)anydomain.com and this domain resolvs to the IP address of the gateway, the request would be forwarded to the gateway, and the GW would accept the call as it comes from a trusted SIP proxy. How can I prevent this? regards, Klaus

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[Serusers] exec_msg
by M.I. Deen 15 Mar '04

15 Mar '04

Hi, I am trying to create an external script using exec_msg, such that the request uri is written based on the SIP_HF_FROM. I have appended this below. Is it possible to have a rewriteuri within an exec_msg. Please give your comments on the script below. Deen if (!exec_msg(' QUERY="select ouser from subscriber1 where user=\"$SIP_HF_FROM\""; OUSER1=`mysql -Bsuser -pheslo -e "$QUERY" ser`; Rewriteuri("sip:$OUSER1@192.168.0.1")')) { sl_send_reply("404", "User does not exist"); } else { sl_send_reply("600", "user exists"); };

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[Serusers] ANNOUNCE: IPv6 now fully supported in nathelper/RTPProxy [Fwd: [Serdev] CVS:commitlog: sip_router/modules/nathelper nathelper.c sip_router/modules/nathelper/examples 4to6.cfg alg.cfg]
by Maxim Sobolev 15 Mar '04

15 Mar '04

With the great pleasure I'm announcing immediate availability of full IPv6 support in RTPproxy/nathelper, which allows to relay IPv6 RTP streams as well as create IPv4<->IPv6 gateways. I've committed a short example, due to absense of the IPv6-capable phone so far it has been tested in the following configuration: UA1->IPv4->SER/RTPproxy->IPv6->SER/RTPproxy->IPv4->UA2 I've committed short example, which illustrates usage of this feature. Enjoy! I'd like to thank Armin Brunner for his donation which made this happen. Regards, Maxim P.S. For those curious, following is the log of one of my test calls: rtpproxy: new session 1e0d3524-4f79dacc(a)192.168.0.100, tag 79b50153b98e6976 requested rtpproxy: new session on a port 35016 created, tag 79b50153b98e6976 rtpproxy: pre-filling caller's address with 192.168.0.100:16412 rtpproxy: lookup on a ports 35016/35022, session timer restarted rtpproxy: pre-filling callee's address with 2002:c3ea:d4b2::3:35030 rtpproxy: lookup on a ports 35016/35022, session timer restarted rtpproxy: forcefully deleting session on ports 35016/35022 rtpproxy: RTP stats: 2061 in from callee, 972 in from caller, 3033 relayed, 0 dropped rtpproxy: RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped rtpproxy: session on ports 35016/35022 is cleaned up rtpproxy: new session 1e0d3524-4f79dacc(a)192.168.0.100, tag 79b50153b98e6976 requested rtpproxy: new session on a port 35008 created, tag 79b50153b98e6976 rtpproxy: pre-filling caller's address with 2002:c3ea:d4b2::1:35016 rtpproxy: lookup on a ports 35008/35030, session timer restarted rtpproxy: pre-filling callee's address with 192.168.0.105:16384 rtpproxy: lookup on a ports 35008/35030, session timer restarted rtpproxy: forcefully deleting session on ports 35008/35030 rtpproxy: RTP stats: 2062 in from callee, 972 in from caller, 3034 relayed, 0 dropped rtpproxy: RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped rtpproxy: session on ports 35008/35030 is cleaned up -------- Original Message -------- Subject: [Serdev] CVS:commitlog: sip_router/modules/nathelper nathelper.c sip_router/modules/nathelper/examples 4to6.cfg alg.cfg Date: Sat, 13 Mar 2004 20:57:11 +0100 From: Maxim Sobolev <sobomax(a)portaone.com> To: serdev(a)lists.iptel.org sobomax 2004/03/13 20:57:11 CET SER CVS Repository Modified files: modules/nathelper nathelper.c modules/nathelper/examples alg.cfg Added files: modules/nathelper/examples 4to6.cfg Log: Added support for extracting/rewriting IPv6 addresses in SDPs. Particularly, force_rtp_proxy now can work with IPv6-aware RTP proxy, replacing IPv4 address in SDP with IPv6 one and vice versa. This allows creating full-fledged IPv4<->IPv6 gateway. See 4to6.cfg file for example. Two new options added into force_rtp_proxy: `f' - instructs nathelper to ignore marks inserted by another nathepler in transit to indicate that the session is already goes through another proxy. Allows creating chain of proxies. `d' - flags that IP address in SDP should be trusted. Without this flag, nathelper ignores address in the SDP and uses source address of the SIP message as media address which is passed to the RTP proxy. Revision Changes Path 1.1 +113 -0 sip_router/modules/nathelper/examples/4to6.cfg (new) http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/modules/nathelper/… 1.3 +6 -6 sip_router/modules/nathelper/examples/alg.cfg http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/modules/nathelper/… 1.47 +216 -76 sip_router/modules/nathelper/nathelper.c http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/modules/nathelper/… _______________________________________________ Serdev mailing list serdev(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serdev

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sr-users March 2004