sr-users March 2004

sr-users@lists.kamailio.org

145 participants
286 discussions

[Serusers] Changing of Realname
by Atle Samuelsen 02 Mar '04

02 Mar '04

Hey... When you call a cisco 7960 f.eks. you see a text string with the name of a user. .. Is it possible "hook" ser upto a database, witch contains the whole phonecatalog. then make ser lookup a number, then put it into the realname feeld ? - Atle

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[Serusers] Minu How To SER/RTPPROXY
by Joao Carlos Moura 02 Mar '04

02 Mar '04

Hi all I have seen much doubt on using SER with RTPPROXY, also mine. Already it sends some questions and I did not obtain to decide my problem. UA(192.x.x.x) -----> SIP --------> UA 10.0x0x0 My archive of log (var/log/message) returns the error: ++++++++++++++++++++ ERROR +++++++++++++++++++++++++++++ /var/log/message Feb 27 18:09:47 blacksip ser[3588 ]: ERROR: send_rtpp_command: can't read reply from the RTP proxy Feb 27 18:09:58 blacksip ser[3586 ]: ERROR: send_rtpp_command: can't read reply from the RTP proxy Feb 27 18:09:58 blacksip ser[3586 ]: ERROR: on_reply processing failed ++++++++++++++++++++ ERROR +++++++++++++++++++++++ I am certain that it has a good will of the members in helping those that they are initiating. I suggest that the Mrs Jiri and Klaus, write one mini How You of as to place SER/RTPPROXY being functioned. Thank You João Carlos Moura NiNeTel Telecommunications +55 85 264-9039

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[Serusers] Display uri
by Alan Litster 02 Mar '04

02 Mar '04

Hi, In the way that it is possible to use log(1, "message..."), is it possible to log the value of the uri variable ?? Regards, Alan ------------------------------------------------------------------------------------------------------- This email, and any files transmitted with it, is copyright and may contain confidential information. The contents are intended for the use of the addressee(s) only. Unauthorized use may be unlawful. If you receive this email by mistake, please advise sender immediately. The views of the author may not necessarily constitute the views of Telco Electronics Limited. Nothing in this mail shall bind Telco Electronics Limited in any contract or obligation. Telco Electronics Limited 6-8 Oxford Court Brackley Northants NN13 7XY Tel 07000 701999 Fax 07000 701777

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[Serusers] Accounting is only logging INVITE to my mysql database.
by Atle Samuelsen 02 Mar '04

02 Mar '04

Hey guys. anybody had the problem that the acc.so module just logs the INVITE to the database? when I try to dial 0.. etc.. - Atle #ACC stuff modparam("acc", "log_level", 3) modparam("acc", "log_flag", 12) modparam("acc", "log_missed_flag", 22) #ACC DB STUFF modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 31) # ------------------------- request routing logic ------------------- # main routing logic route{ if (uri=~"sip:0[0-9][3.5]*") { # ... it is for the peer site strip(1); setflag(1); rewritehostport("192.168.11.5"); t_relay(); break; };

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[Serusers] A problem with SIP-PSTN Gateway.please help me.
by 黄谦 02 Mar '04

02 Mar '04

Dear sir: I am using a Cisco 3660 router as a SIP-PSTN gateway with my SER server.When I have configure all the thing has the guideline,I meet a problem.If I use a telephone call my MSN Messager or use MSN Messager call the telephone,the receiver could ring,and the MSN Messager could send and receive signal,but the telephone could only receive.In other words,the RTP packages could not send to UA at computers from the router.Does anyone can help me? This is the configure at SER server: # ------------------ module loading ---------------------------------- loadmodule "modules/sl/sl.so" loadmodule "modules/tm/tm.so" loadmodule "modules/acc/acc.so" loadmodule "modules/rr/rr.so" loadmodule "modules/maxfwd/maxfwd.so" loadmodule "modules/mysql/mysql.so" loadmodule "modules/auth/auth.so" loadmodule "modules/auth_db/auth_db.so" loadmodule "modules/group/group.so" loadmodule "modules/uri/uri.so" # ----------------- setting module-specific parameters --------------- modparam("auth_db", "db_url","sql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") # -- acc params -- modparam("acc", "log_level", 1) # that is the flag for which we will account -- don't forget to # set the same one :-) modparam("acc", "log_flag", 1 ) # ------------------------- request routing logic ------------------- # main routing logic route{ /* ********* ROUTINE CHECKS ********************************** */ # filter too old messages if (!mf_process_maxfwd_header("10")) { log("LOG: Too many hops\n"); sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Wow -- Message too large"); break; }; /* ********* RR ********************************** */ /* grant Route routing if route headers present */ if (loose_route()) { t_relay(); break; }; /* record-route INVITEs -- all subsequent requests must visit us */ if (method=="INVITE") { record_route(); }; # now check if it really is a PSTN destination which should be handled # by our gateway; if not, and the request is an invitation, drop it -- # we cannot terminate it in PSTN; relay non-INVITE requests -- it may # be for example BYEs sent by gateway to call originator if (!uri=~"sip:\+?[0-9]+@.*") { if (method=="INVITE") { sl_send_reply("403", "Call cannot be served here"); } else { forward(uri:host, uri:port); }; break; }; # account completed transactions via syslog setflag(1); # free call destinations ... no authentication needed if ( is_user_in("Request-URI", "free-pstn") /* free destinations */ | uri=~"sip:[79][0-9][0-9][0-9]@.*" /* local PBX */ | uri=~"sip:98[0-9][0-9][0-9][0-9]") { log("free call"); } else if (src_ip==192.168.0.10) { # our gateway doesn't support digest authentication; # verify that a request is coming from it by source # address log("gateway-originated request"); } else { # in all other cases, we need to check the request against # access control lists; first of all, verify request # originator's identity if (!proxy_authorize( "gateway" /* realm */, "subscriber" /* table name */)) { proxy_challenge( "gateway" /* realm */, "0" /* no qop */ ); break; }; # authorize only for INVITEs -- RR/Contact may result in weird # things showing up in d-uri that would break our logic; our # major concern is INVITE which causes PSTN costs if (method=="INVITE") { # does the authenticated user have a permission for local # calls (destinations beginning with a single zero)? # (i.e., is he in the "local" group?) if (uri=~"sip:0[1-9][0-9]+@.*") { if (!is_user_in("credentials", "local")) { sl_send_reply("403", "No permission for local calls"); break; }; # the same for long-distance (destinations begin with two zeros") } else if (uri=~"sip:00[1-9][0-9]+@.*") { if (!is_user_in("credentials", "ld")) { sl_send_reply("403", " no permission for LD "); break; }; # the same for international calls (three zeros) } else if (uri=~"sip:000[1-9][0-9]+@.*") { if (!is_user_in("credentials", "int")) { sl_send_reply("403", "International permissions needed"); break; }; # everything else (e.g., interplanetary calls) is denied } else { sl_send_reply("403", "Forbidden"); break; }; }; # INVITE to authorized PSTN }; # authorized PSTN # if you have passed through all the checks, let your call go to GW! rewritehostport("192.168.0.10:5060"); # forward the request now if (!t_relay()) { sl_reply_error(); break; }; } And this is the configure at the Cisco 3660: ! version 12.2 service timestamps debug uptime service timestamps log uptime no service password-encryption ! hostname SIP ! enable secret 5 $1$Ov6R$SoCYqYuzY7a7spMUSEz2e. enable password 123456 ! ip subnet-zero no ip routing ! ! ip name-server 162.105.170.66 ip name-server 162.105.129.27 ! ! voice rtp send-recv ! voice service pots ! voice class codec 1 codec preference 1 g729r8 codec preference 2 g729br8 codec preference 13 g711ulaw codec preference 14 g711alaw ! voice class codec 13 ! ! ! ! ! ! fax interface-type fax-mail mta receive maximum-recipients 0 ! ! ! ! interface FastEthernet0/0 ip address 192.168.0.200 255.255.255.0 no ip route-cache no ip mroute-cache speed auto half-duplex no cdp enable ! interface FastEthernet0/1 no ip address no ip route-cache no ip mroute-cache shutdown duplex auto speed auto no cdp enable ! interface Serial3/0 no ip address no ip route-cache no ip mroute-cache shutdown serial restart_delay 0 no cdp enable ! interface Serial3/1 no ip address no ip route-cache no ip mroute-cache shutdown serial restart_delay 0 no cdp enable ! interface Serial3/2 no ip address no ip route-cache no ip mroute-cache shutdown serial restart_delay 0 no cdp enable ! interface Serial3/3 no ip address no ip route-cache no ip mroute-cache shutdown serial restart_delay 0 no cdp enable ! interface FastEthernet5/0 no ip address no ip route-cache no ip mroute-cache shutdown duplex auto speed auto no cdp enable ! interface Serial5/0 no ip address no ip route-cache no ip mroute-cache shutdown clockrate 2000000 no cdp enable ! interface FastEthernet5/1 no ip address no ip route-cache no ip mroute-cache shutdown duplex auto speed auto no cdp enable ! interface Serial5/1 no ip address no ip route-cache no ip mroute-cache shutdown clockrate 2000000 no cdp enable ! interface FastEthernet6/0 no ip address no ip route-cache no ip mroute-cache shutdown duplex auto speed auto no cdp enable ! interface Serial6/0 no ip address no ip route-cache no ip mroute-cache shutdown clockrate 2000000 no cdp enable ! interface FastEthernet6/1 no ip address no ip route-cache no ip mroute-cache shutdown duplex auto speed auto no cdp enable ! interface Serial6/1 no ip address no ip route-cache no ip mroute-cache shutdown clockrate 2000000 no cdp enable ! interface Dialer1 no ip address ! ip default-gateway 192.168.0.1 ip classless no ip http server ip pim bidir-enable ! ! vc-group 1 ! snmp-server community public RO ! call rsvp-sync ! voice-port 1/0/0 cptone CN description 3078 supervisory disconnect dualtone mid-call ! voice-port 1/0/1 ! voice-port 1/1/0 cptone CN description 3078 supervisory disconnect dualtone mid-call ! voice-port 1/1/1 ! voice-port 2/0/0 ! voice-port 2/0/1 ! voice-port 2/1/0 ! voice-port 2/1/1 ! ! mgcp profile default ! dial-peer cor custom ! ! ! dial-peer voice 86100 pots application session destination-pattern 8610T port 1/0/0 ! dial-peer voice 1000 voip application session destination-pattern .T session protocol sipv2 session target sip-server session transport udp codec g711ulaw ! dial-peer voice 100 pots destination-pattern 010T port 1/0/0 ! dial-peer voice 86000 pots destination-pattern 8600T port 1/1/0 ! sip-ua timers disconnect 200 sip-server dns:sipproxy.icst.pku.edu.cn ! ! line con 0 line aux 0 line vty 0 4 password 123456 login ! ! end

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[Serusers] Ser script.
by Laurent BURGY 02 Mar '04

02 Mar '04

hi, I want to know if there is any complete documentation about SER scripts... Type definition, pointers, arrays ( to declare an array with several uris ) or struct.... I've read the cfg.y file but i didn't understand much... Maybe the best way to process these kinds of types is using a module... thx, Laurent

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[Serusers] I have a problem with SIP-PSTN Gateway.please help me.
by 黄谦 02 Mar '04

02 Mar '04

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FW: [Serusers] DNS SRV
by Toni Barata 01 Mar '04

01 Mar '04

Hi Again, Does SER support queries to DNS SRV?Yes?No? If yes.How can we do to configure it that way? Thanks, Toni -----Original Message----- From: Toni Barata [mailto:toni-r-barata@ptinovacao.pt] Sent: quarta-feira, 25 de Fevereiro de 2004 17:21 To: 'serusers(a)lists.iptel.org' Subject: [Serusers] DNS SRV Hi All, In case we pretend to proxy the messages from SER to multiple Media Servers, it is possible to use DNS SRV for load balancing and fault tolerance? We have already one implementation of load balancing e fault tolerance for SER (using DNS SRV) but now we want to implement the same configuration for ours Media Servers. Best regards, Toni

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[Serusers] rtpproxy error
by Joao Carlos Moura 01 Mar '04

01 Mar '04

Hi, I have that to implement in my company a solution VOIP. I am testing SER/RTPProxy. My problem is in the call of the equipment with IP 192.168.0.20 for another side with IP 200.2xx.xx.xx. In the archive /var/log/message, has the error message ++++++++++++++++++++ ERROR +++++++++++++++++++++++++++++ /var/log/message Feb 27 18:09:47 blacksip ser[3588 ]: ERROR: send_rtpp_command: can't read reply from the RTP proxy Feb 27 18:09:58 blacksip ser[3586 ]: ERROR: send_rtpp_command: can't read reply from the RTP proxy Feb 27 18:09:58 blacksip ser[3586 ]: ERROR: on_reply processing failed ++++++++++++++++++++ ERROR +++++++++++++++++++++++ The project that I intend to implant is similar to the FWD. The version that I am using is: CVS SER 0.8.12-TCP_nonb (i386/linux) CVS rtpproxy 20040107 Athlon 1900+ / 256Mb / 40GB HD REDHat 9.0 Thank You João Carlos Moura

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[Serusers] Re: Pardon, really more necessary of help
by Klaus Darilion 01 Mar '04

01 Mar '04

Hi Joao! You should ask your questions to the mailing list which were setup for these problems. You can subscribe at: http://lists.iptel.org/mailman/listinfo/serusers And then you should send your questions to serusers(a)lists.iptel.org These versions should work fine. Are you using the original 0.8.12 release or have you used CVS to retrieve 0.8.12? There were some updates to the nathelper module which are not in the original 0.8.12 release, but in the CVS rel_0_8_12. You should donwload from cvs, compile ser again and try again. Klaus Joao Carlos Moura wrote: > Hi Mr. Klaus, > > Thank you for helping me > > Pardon, really more necessary of help > > The version that I am using is: SER 0.8.12-TCP_nonb (i386/linux) > rtpproxy 20040107 > > Athlon 1900+ / 256Mb / 40GB HD > REDHat 9.0 > > My archive of log (var/log/message) returns the error: > ++++++++++++++++++++ ERROR +++++++++++++++++++++++++++++ /var/log/message > Feb 27 18:09:47 blacksip ser[3588 ]: ERROR: send_rtpp_command: can't read > reply from the RTP proxy > Feb 27 18:09:58 blacksip ser[3586 ]: ERROR: send_rtpp_command: can't read > reply from the RTP proxy > Feb 27 18:09:58 blacksip ser[3586 ]: ERROR: on_reply processing failed > ++++++++++++++++++++ ERROR +++++++++++++++++++++++ > > Thank You > > João Carlos Moura > NiNeTel Telecommunications > +55 85 264-9039 > > >

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sr-users March 2004