sr-users September 2005

sr-users@lists.kamailio.org

231 participants
507 discussions

by harry gaillac

Dear Cesc, Is it easier to be amazed than helping people ? Harry --- Cesc <cesc.santa(a)gmail.com> a écrit : > Hi harry, > i can't wait to see what prank you pull with your > next email ... > not that i care, but at least i get a good laugh. > Sincerely amazed, > Cesc > > On 9/8/05, harry gaillac <gaillacharry(a)yahoo.fr> > wrote: > > > > Hello, > > > > Forget this mail ! > > > http://lists.iptel.org/pipermail/serusers/2005-September/023710.html > > > > Harry > > > > > > > > > > > > > > > > > ___________________________________________________________________________ > > Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > > Téléchargez cette version sur > http://fr.messenger.yahoo.com > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

19 years, 2 months

[Serusers] NEED HELP !!!

by harry gaillac

Hello, I've been spending much time to understand why SER don't reply to caller with public ip in SDP message. I try both rtpproxy and mediaproxy. when caller/callee are both registered in registrar server they are no problem. I can't solve it !!!! Harry My ser.cfg # $Id: ser.cfg,v 1.27 2005/03/10 14:16:25 Exp $ # # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) #memlog=5 # memory debug log level #log_facility=LOG_LOCAL0 # sets the facility used for logging (see syslog(3)) /* Uncomment these lines to enter debugging mode #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" fifo_db_url="mysql://ser:heslo@serveur1.home.net/ser" user=root group=ser fifo_user=root # owner of the ser fifo fifo_group=ser fifo_mode=0660 # fifo's permissions #disable_core=yes #disables core dumping #open_fd_limit=1024 # sets the open file descriptors limit #mhomed=yes # usefull for multihomed hosts, small performance penalty #disable_tcp=yes #tcp_accept_aliases=yes # accepts the tcp alias via option (see NEWS) # listen=80.119.5.100 # ------------------ module loading ---------------------------------- 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" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/uri_db.so" loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/mediaproxy.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/permissions.so" loadmodule "/usr/lib/ser/modules/domain.so" loadmodule "/usr/lib/ser/modules/group.so" loadmodule "/usr/lib/ser/modules/acc.so" loadmodule "/usr/lib/ser/modules/msilo.so" loadmodule "/usr/lib/ser/modules/avpops.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params-- modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) # -- autdb_params-- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", 1) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- nathelper params -- modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "rtpproxy_disable", 1) # mediaproxy params -- modparam("mediaproxy", "natping_interval", 10) modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") # -- registrar params -- modparam("registrar", "nat_flag", 4) modparam("registrar", "use_domain", 1) # -- tm params -- modparam("tm", "fr_inv_timer", 15) modparam("tm", "fr_timer", 10) # --domain params -- modparam("domain", "db_url", "mysql://ser:heslo@serveur1.home.net/ser") modparam("domain", "domain_table", "domain") modparam("domain", "db_mode", 1) # Use caching # -- permissions params -- modparam("permissions", "db_url", "mysql://ser:heslo@serveur1.home.net/ser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "default_deny_file", "/etc/ser/permissions.deny") modparam("permissions", "default_allow_file", "/etc/ser/permissions.allow") # -- acc params -- modparam("acc", "db_url", "mysql://ser:heslo@serveur1.home.net/ser") modparam("acc", "log_fmt", "miocfsp") modparam("acc", "failed_transactions", 1) modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 3) # -- msilo params -- modparam("msilo", "db_url", "mysql://ser:heslo@serveur1.home.net/ser") modparam("msilo", "check_time", 10) # -- avpos params -- modparam("avpops","avp_url","mysql://ser:heslo@serveur1.home.net/ser") modparam("avpops","avp_table","usr_preferences") # ------------------------- 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 if (!method=="REGISTER") { record_route(); }; if (method=="CANCEL" || method=="BYE") { end_media_session(); }; # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { if (has_totag() && method=="INVITE" || method=="ACK") { if (client_nat_test("3")) { setflag(4); force_rport(); fix_contact(); }; use_media_proxy(); }; route(1); break; }; # if the request is for our domains use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (is_uri_host_local()) { if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; } else if (method=="MESSAGE") { route(7); break; } else if (method=="SUBSCRIBE" || method=="NOTIFY") { route(8); break; }; }; # Requests to others domains if (!is_uri_host_local()) { # Checking Authorization if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("407", "Proxy Authentication Required"); break; }; if (client_nat_test("3")) { setflag(4); }; if (isflagset(4)) { force_rport(); fix_contact(); use_media_proxy(); }; # Message Handler to others domains # if (method=="MESSAGE") { route(1); break; }; # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; } route[1] { ########################### # Default Message Handler # ########################### t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } route[2] { ############################ # Register Message Handler # ############################ if (client_nat_test("3")) { setflag(4); force_rport(); fix_contact(); }; if (!www_authorize("", "subscriber")) { www_challenge("", "0"); sl_send_reply("401", "Unauthorized"); break; }; if (!is_from_local()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); break; }; m_dump(); } route[3] { ############################# # CANCEL and INVITE handler # ############################# # SIP destinations to PSTN if (uri=~"^sip:[0-9]{4}@.*$" || uri=~"^sip:[0-9]{10}@.*$" || uri=~"^sip:06[0-9]{8}@.*$" || uri=~"^sip:00.*@.*$") { route(4); break; }; # SIP destinations to IPBX if (uri=~"^sip:8[0-3]@.*$") { route(5); break; } # Check AVPs for Voicemail if ((avp_db_load("$ruri","s:fw_voicemail/usr_preferences") && avp_check("fw_voicemail", "eq/i:1"))) { route(6); route(1); break; }; # accounting db flags setflag(2); setflag(3); acc_db_request("404 missed call", "missed_calls"); # Check Nated User Agents if (client_nat_test("3")) { setflag(4); }; if (isflagset(4)) { force_rport(); fix_contact(); use_media_proxy(); }; # Check ACL for Voicemail if (is_user_in("Request-URI","voicemail")) { setflag(5); }; # native SIP destinations are handled using our USRLOC DB lookup("aliases"); if (!lookup("location")) { if (isflagset(5)) { route(6); route(1); break; } else { sl_send_reply("404", "User Not Found"); break; }; }; t_on_failure("1"); route(1); break; } route[4] { ########################## # Outgoing PSTN Handler # ########################## # Checking Authorization to PSTN if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("407", "Proxy Authentication Required"); break; }; if (uri=~"^sip:[0-9]{4}@.*$") { if (!is_user_in("From", "local")) { sl_send_reply("403", "Forbidden"); break; }; } else if (uri=~"^sip:[0-9]{10}@.*$") { if (!is_user_in("From", "local")) { sl_send_reply("403", "Forbidden"); break; }; } else if (uri=~"^sip:06[0-9]{8}@.*$") { if (!is_user_in("From", "cellphone")) { sl_send_reply("403", "Forbidden"); break; }; } else if (uri=~"^sip:00.@.*$") { if (!is_user_in("From", "int")) { sl_send_reply("403", "Forbidden"); break; }; }; prefix("0"); rewritehostport("asterisk.home.net:5050"); route(1); break; } route[5] { ########################## # Asterisk Handler # ########################## # Checking Authorization to Asterisk if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("407", "Proxy Authentication Required"); break; }; if (uri=~"^sip:8[0-3]@.*$") { if (!is_user_in("From", "local")) { sl_send_reply("403", "Forbidden"); break; }; }; rewritehostport("asterisk.home.net:5050"); route(1); break; } route[6] { ########################## # Voicemail Handler # ########################## revert_uri(); # resend to voicemail with original request URI prefix("u"); rewritehostport("asterisk.home.net:5050"); break; } route[7] { ########################## # Message Handler # ########################### # Checking Authorization if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("407", "Proxy Authentication Required"); break; }; lookup("aliases"); if (!lookup("location")) { if (method == "MESSAGE") { if (!t_newtran()) { sl_reply_error(); break; }; if (m_store("0")) { t_reply("202", "Accepted for Later Delivery"); break; }; t_reply("503", "Service Unavailable"); break; }; }; route(1); } route[8] { ################################ # SUBSCRIBE/NOTIFY Handler # ################################ # Checking Authorization if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("407", "Proxy Authentication Required"); break; }; lookup("aliases"); if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } onreply_route[1] { ########################## # Statefull Handler # ########################## if (status=~"(180) || (183) || 2[0-9][0-9]") { if(search("^Content-Type: .*$")) { use_media_proxy(); }; }; } failure_route[1] { ########################## # Voicemail Handler # ########################## if (t_check_status("486")) { revert_uri(); # resend to voicemail with original request URI prefix("b"); rewritehostport("asterisk.home.net:5050"); append_branch(); t_relay_to_udp("asterisk.home.net","5050"); break; }; if (t_check_status("408")) { revert_uri(); # resend to voicemail with original request URI prefix("u"); rewritehostport("asterisk.home.net:5050"); append_branch(); t_relay_to_udp("asterisk.home.net","5050"); break; }; } ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

19 years, 2 months

[Serusers] Voip and Firewalls

by harry gaillac

Hello, What are the policies to apply in order to secure a network with Voip context? Which ports must be openned for signalling and which ports must be openned for rtp traffic incoming/outgoing ? Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

19 years, 2 months

[Serusers] ser mysql database sync

by Jack Wei

i'm not familiar w/ ser enough to make the following changes, but does anyone know how to make ser update entries in table location that are present in the cache only. I have 2 ser servers and i want to use 1 database. Jack Wei __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

19 years, 2 months

[Serusers] one way voice on public IP...ngrep

by Iqbal

Hi All the IP adds here are public, but I get one way voice, i.e from UA to pstn I can get voice, but not from pstn to UA box UA ---SER---PSTN a.b.c.d - UA x.x.x.x = SER y.y.y.y = pstn gw a.a.a.a = pstn gw 2 The c= and all seem correct, and natting problems shouldn't come into this...all ports are open, there is a FW, but that doesnt show anything, any thoughts on howto debug this Iqbal -------- U 2005/09/12 14:20:26.349238 a.b.c.d:5060 -> x.x.x.x:5060 INVITE sip:0123456789@sip.domain.com SIP/2.0. Content-Type: application/sdp. Via: SIP/2.0/UDP a.b.c.d:5060. To: <sip:0123456789@sip.domain.com>. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257cfc-1c12c480-4823. CSeq: 1 INVITE. Call-ID: c7ec14d4-13c4-43257cfc-1c12c480-29-15a7ee8. Contact: <sip:047132@a.b.c.d:5060>. Content-Length: 241. . v=0. o=voxeo 0 0 IN IP4 a.b.c.d. s=voxeo. t=0 0. m=audio 10198 RTP/AVP 0 121 101. c=IN IP4 a.b.c.d. a=rtpmap:121 frf-dialed-digit/8000. a=fmtp:121 0-15. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=rtpmap:0 PCMU/8000. ## U 2005/09/12 14:20:26.393835 x.x.x.x:5060 -> a.b.c.d:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP a.b.c.d:5060. To: <sip:0123456789@sip.domain.com>. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257cfc-1c12c480-4823. CSeq: 1 INVITE. Call-ID: c7ec14d4-13c4-43257cfc-1c12c480-29-15a7ee8. Content-Length: 0. Warning: 392 x.x.x.x:5060 "Noisy feedback tells: pid=31505 req_src_ip=a.b.c.d req_src_port=5060 in_uri=sip:0123456789@sip.domain.com out_uri=sip:44123456789@y.y.y.y:5060 via_cnt==1". . # U 2005/09/12 14:20:26.394104 x.x.x.x:5060 -> y.y.y.y:5060 INVITE sip:44123456789@y.y.y.y:5060 SIP/2.0. Max-Forwards: 10. Record-Route: <sip:x.x.x.x;ftag=c7ec14d4-13c4-43257cfc-1c12c480-4823;lr=on>. Content-Type: application/sdp. Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bKc628.0cf5ca27.0. Via: SIP/2.0/UDP a.b.c.d:5060. To: <sip:0123456789@sip.domain.com>. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257cfc-1c12c480-4823. CSeq: 1 INVITE. Call-ID: c7ec14d4-13c4-43257cfc-1c12c480-29-15a7ee8. Contact: <sip:047132@a.b.c.d:5060>. Content-Length: 241. . v=0. o=voxeo 0 0 IN IP4 a.b.c.d. s=voxeo. t=0 0. m=audio 10198 RTP/AVP 0 121 101. c=IN IP4 a.b.c.d. a=rtpmap:121 frf-dialed-digit/8000. a=fmtp:121 0-15. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=rtpmap:0 PCMU/8000. ## U 2005/09/12 14:20:26.401238 y.y.y.y:5060 -> x.x.x.x:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bKc628.0cf5ca27.0. Via: SIP/2.0/UDP a.b.c.d:5060. To: <sip:0123456789@sip.domain.com>. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257cfc-1c12c480-4823. CSeq: 1 INVITE. Call-ID: c7ec14d4-13c4-43257cfc-1c12c480-29-15a7ee8. Server: Sip EXpress router (0.8.99-dev (i386/linux)). Content-Length: 0. . ###### U 2005/09/12 14:20:30.161123 y.y.y.y:5060 -> x.x.x.x:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bKc628.0cf5ca27.0,SIP/2.0/UDP a.b.c.d:5060. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257cfc-1c12c480-4823. To: <sip:0123456789@sip.domain.com>;tag=38EED020-1D04. Date: Mon, 12 Sep 2005 13:04:45 gmt. Call-ID: c7ec14d4-13c4-43257cfc-1c12c480-29-15a7ee8. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow-Events: telephone-event. Contact: <sip:0044123456789@a.a.a.a:5060>. Record-Route: <sip:y.y.y.y;ftag=c7ec14d4-13c4-43257cfc-1c12c480-4823;lr=on>,<sip:x.x.x.x;ftag=c7ec14d4-13c4-43257cfc-1c12c480-4823;lr=on>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 182. . v=0. o=CiscoSystemsSIP-GW-UserAgent 5590 7529 IN IP4 a.a.a.a. s=SIP Call. c=IN IP4 a.a.a.a. t=0 0. m=audio 17896 RTP/AVP 0. c=IN IP4 a.a.a.a. a=rtpmap:0 PCMU/8000. # U 2005/09/12 14:20:30.163775 x.x.x.x:5060 -> a.b.c.d:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP a.b.c.d:5060. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257cfc-1c12c480-4823. To: <sip:0123456789@sip.domain.com>;tag=38EED020-1D04. Date: Mon, 12 Sep 2005 13:04:45 gmt. Call-ID: c7ec14d4-13c4-43257cfc-1c12c480-29-15a7ee8. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow-Events: telephone-event. Contact: <sip:0044123456789@a.a.a.a:5060>. Record-Route: <sip:y.y.y.y;ftag=c7ec14d4-13c4-43257cfc-1c12c480-4823;lr=on>,<sip:x.x.x.x;ftag=c7ec14d4-13c4-43257cfc-1c12c480-4823;lr=on>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 182. . v=0. o=CiscoSystemsSIP-GW-UserAgent 5590 7529 IN IP4 a.a.a.a. s=SIP Call. c=IN IP4 a.a.a.a. t=0 0. m=audio 17896 RTP/AVP 0. c=IN IP4 a.a.a.a. a=rtpmap:0 PCMU/8000. ## U 2005/09/12 14:20:31.368083 y.y.y.y:5060 -> x.x.x.x:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bKc628.0cf5ca27.0,SIP/2.0/UDP a.b.c.d:5060. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257cfc-1c12c480-4823. To: <sip:0123456789@sip.domain.com>;tag=38EED020-1D04. Date: Mon, 12 Sep 2005 13:04:45 gmt. Call-ID: c7ec14d4-13c4-43257cfc-1c12c480-29-15a7ee8. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. Allow-Events: telephone-event. Contact: <sip:0044123456789@a.a.a.a:5060>. Record-Route: <sip:y.y.y.y;ftag=c7ec14d4-13c4-43257cfc-1c12c480-4823;lr=on>,<sip:x.x.x.x;ftag=c7ec14d4-13c4-43257cfc-1c12c480-4823;lr=on>. Content-Type: application/sdp. Content-Length: 182. . v=0. o=CiscoSystemsSIP-GW-UserAgent 5590 7529 IN IP4 a.a.a.a. s=SIP Call. c=IN IP4 a.a.a.a. t=0 0. m=audio 17896 RTP/AVP 0. c=IN IP4 a.a.a.a. a=rtpmap:0 PCMU/8000. # U 2005/09/12 14:20:31.370673 x.x.x.x:5060 -> a.b.c.d:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP a.b.c.d:5060. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257cfc-1c12c480-4823. To: <sip:0123456789@sip.domain.com>;tag=38EED020-1D04. Date: Mon, 12 Sep 2005 13:04:45 gmt. Call-ID: c7ec14d4-13c4-43257cfc-1c12c480-29-15a7ee8. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. Allow-Events: telephone-event. Contact: <sip:0044123456789@a.a.a.a:5060>. Record-Route: <sip:y.y.y.y;ftag=c7ec14d4-13c4-43257cfc-1c12c480-4823;lr=on>,<sip:x.x.x.x;ftag=c7ec14d4-13c4-43257cfc-1c12c480-4823;lr=on>. Content-Type: application/sdp. Content-Length: 182. . v=0. o=CiscoSystemsSIP-GW-UserAgent 5590 7529 IN IP4 a.a.a.a. s=SIP Call. c=IN IP4 a.a.a.a. t=0 0. m=audio 17896 RTP/AVP 0. c=IN IP4 a.a.a.a. a=rtpmap:0 PCMU/8000. # U 2005/09/12 14:20:31.392583 a.b.c.d:5060 -> x.x.x.x:5060 ACK sip:x.x.x.x;ftag=c7ec14d4-13c4-43257cfc-1c12c480-4823;lr=on SIP/2.0. Via: SIP/2.0/UDP a.b.c.d:5060. To: <sip:0123456789@sip.domain.com>;tag=38EED020-1D04. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257cfc-1c12c480-4823. CSeq: 1 ACK. Call-ID: c7ec14d4-13c4-43257cfc-1c12c480-29-15a7ee8. Contact: <sip:047132@a.b.c.d:5060>. Route: <sip:y.y.y.y;ftag=c7ec14d4-13c4-43257cfc-1c12c480-4823;lr=on>. Route: <sip:0044123456789@a.a.a.a:5060>. Content-Length: 0. . # U 2005/09/12 14:20:31.394335 x.x.x.x:5060 -> y.y.y.y:5060 ACK sip:0044123456789@a.a.a.a:5060 SIP/2.0. Max-Forwards: 10. Record-Route: <sip:x.x.x.x;ftag=c7ec14d4-13c4-43257cfc-1c12c480-4823;lr=on>. Via: SIP/2.0/UDP x.x.x.x;branch=0. Via: SIP/2.0/UDP a.b.c.d:5060. To: <sip:0123456789@sip.domain.com>;tag=38EED020-1D04. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257cfc-1c12c480-4823. CSeq: 1 ACK. Call-ID: c7ec14d4-13c4-43257cfc-1c12c480-29-15a7ee8. Contact: <sip:047132@a.b.c.d:5060>. Route: <sip:0044123456789@a.a.a.a:5060>. Content-Length: 0. . ############################################################################################################################################################################################################################################################################################################################### U 2005/09/12 14:22:25.033315 a.b.c.d:5060 -> x.x.x.x:5060 INVITE sip:0123456789@sip.domain.com SIP/2.0. Content-Type: application/sdp. Via: SIP/2.0/UDP a.b.c.d:5060. To: <sip:0123456789@sip.domain.com>. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257d73-1c14942f-4823. CSeq: 1 INVITE. Call-ID: c7ec14d4-13c4-43257d73-1c14942f-29-15a7ee8. Contact: <sip:047132@a.b.c.d:5060>. Content-Length: 241. . v=0. o=voxeo 0 0 IN IP4 a.b.c.d. s=voxeo. t=0 0. m=audio 10198 RTP/AVP 0 121 101. c=IN IP4 a.b.c.d. a=rtpmap:121 frf-dialed-digit/8000. a=fmtp:121 0-15. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=rtpmap:0 PCMU/8000. # U 2005/09/12 14:22:25.067537 x.x.x.x:5060 -> a.b.c.d:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP a.b.c.d:5060. To: <sip:0123456789@sip.domain.com>. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257d73-1c14942f-4823. CSeq: 1 INVITE. Call-ID: c7ec14d4-13c4-43257d73-1c14942f-29-15a7ee8. Content-Length: 0. Warning: 392 x.x.x.x:5060 "Noisy feedback tells: pid=31508 req_src_ip=a.b.c.d req_src_port=5060 in_uri=sip:0123456789@sip.domain.com out_uri=sip:441560485332@y.y.y.y:5060 via_cnt==1". . # U 2005/09/12 14:22:25.068063 x.x.x.x:5060 -> y.y.y.y:5060 INVITE sip:441560485332@y.y.y.y:5060 SIP/2.0. Max-Forwards: 10. Record-Route: <sip:x.x.x.x;ftag=c7ec14d4-13c4-43257d73-1c14942f-4823;lr=on>. Content-Type: application/sdp. Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bK7719.a95a2e74.0. Via: SIP/2.0/UDP a.b.c.d:5060. To: <sip:0123456789@sip.domain.com>. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257d73-1c14942f-4823. CSeq: 1 INVITE. Call-ID: c7ec14d4-13c4-43257d73-1c14942f-29-15a7ee8. Contact: <sip:047132@a.b.c.d:5060>. Content-Length: 241. . v=0. o=voxeo 0 0 IN IP4 a.b.c.d. s=voxeo. t=0 0. m=audio 10198 RTP/AVP 0 121 101. c=IN IP4 a.b.c.d. a=rtpmap:121 frf-dialed-digit/8000. a=fmtp:121 0-15. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=rtpmap:0 PCMU/8000. # U 2005/09/12 14:22:25.078167 y.y.y.y:5060 -> x.x.x.x:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bK7719.a95a2e74.0. Via: SIP/2.0/UDP a.b.c.d:5060. To: <sip:0123456789@sip.domain.com>. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257d73-1c14942f-4823. CSeq: 1 INVITE. Call-ID: c7ec14d4-13c4-43257d73-1c14942f-29-15a7ee8. Server: Sip EXpress router (0.8.99-dev (i386/linux)). Content-Length: 0. . ############ U 2005/09/12 14:22:28.963663 y.y.y.y:5060 -> x.x.x.x:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bK7719.a95a2e74.0,SIP/2.0/UDP a.b.c.d:5060. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257d73-1c14942f-4823. To: <sip:0123456789@sip.domain.com>;tag=38F09FAC-544. Date: Mon, 12 Sep 2005 13:06:43 gmt. Call-ID: c7ec14d4-13c4-43257d73-1c14942f-29-15a7ee8. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow-Events: telephone-event. Contact: <sip:0044123456789@a.a.a.a:5060>. Record-Route: <sip:y.y.y.y;ftag=c7ec14d4-13c4-43257d73-1c14942f-4823;lr=on>,<sip:x.x.x.x;ftag=c7ec14d4-13c4-43257d73-1c14942f-4823;lr=on>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 182. . v=0. o=CiscoSystemsSIP-GW-UserAgent 3344 8781 IN IP4 a.a.a.a. s=SIP Call. c=IN IP4 a.a.a.a. t=0 0. m=audio 16636 RTP/AVP 0. c=IN IP4 a.a.a.a. a=rtpmap:0 PCMU/8000. # U 2005/09/12 14:22:28.966645 x.x.x.x:5060 -> a.b.c.d:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP a.b.c.d:5060. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257d73-1c14942f-4823. To: <sip:0123456789@sip.domain.com>;tag=38F09FAC-544. Date: Mon, 12 Sep 2005 13:06:43 gmt. Call-ID: c7ec14d4-13c4-43257d73-1c14942f-29-15a7ee8. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow-Events: telephone-event. Contact: <sip:0044123456789@a.a.a.a:5060>. Record-Route: <sip:y.y.y.y;ftag=c7ec14d4-13c4-43257d73-1c14942f-4823;lr=on>,<sip:x.x.x.x;ftag=c7ec14d4-13c4-43257d73-1c14942f-4823;lr=on>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 182. . v=0. o=CiscoSystemsSIP-GW-UserAgent 3344 8781 IN IP4 a.a.a.a. s=SIP Call. c=IN IP4 a.a.a.a. t=0 0. m=audio 16636 RTP/AVP 0. c=IN IP4 a.a.a.a. a=rtpmap:0 PCMU/8000. # U 2005/09/12 14:22:30.228227 y.y.y.y:5060 -> x.x.x.x:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bK7719.a95a2e74.0,SIP/2.0/UDP a.b.c.d:5060. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257d73-1c14942f-4823. To: <sip:0123456789@sip.domain.com>;tag=38F09FAC-544. Date: Mon, 12 Sep 2005 13:06:43 gmt. Call-ID: c7ec14d4-13c4-43257d73-1c14942f-29-15a7ee8. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. Allow-Events: telephone-event. Contact: <sip:0044123456789@a.a.a.a:5060>. Record-Route: <sip:y.y.y.y;ftag=c7ec14d4-13c4-43257d73-1c14942f-4823;lr=on>,<sip:x.x.x.x;ftag=c7ec14d4-13c4-43257d73-1c14942f-4823;lr=on>. Content-Type: application/sdp. Content-Length: 182. . v=0. o=CiscoSystemsSIP-GW-UserAgent 3344 8781 IN IP4 a.a.a.a. s=SIP Call. c=IN IP4 a.a.a.a. t=0 0. m=audio 16636 RTP/AVP 0. c=IN IP4 a.a.a.a. a=rtpmap:0 PCMU/8000. # U 2005/09/12 14:22:30.230960 x.x.x.x:5060 -> a.b.c.d:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP a.b.c.d:5060. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257d73-1c14942f-4823. To: <sip:0123456789@sip.domain.com>;tag=38F09FAC-544. Date: Mon, 12 Sep 2005 13:06:43 gmt. Call-ID: c7ec14d4-13c4-43257d73-1c14942f-29-15a7ee8. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. Allow-Events: telephone-event. Contact: <sip:0044123456789@a.a.a.a:5060>. Record-Route: <sip:y.y.y.y;ftag=c7ec14d4-13c4-43257d73-1c14942f-4823;lr=on>,<sip:x.x.x.x;ftag=c7ec14d4-13c4-43257d73-1c14942f-4823;lr=on>. Content-Type: application/sdp. Content-Length: 182. . v=0. o=CiscoSystemsSIP-GW-UserAgent 3344 8781 IN IP4 a.a.a.a. s=SIP Call. c=IN IP4 a.a.a.a. t=0 0. m=audio 16636 RTP/AVP 0. c=IN IP4 a.a.a.a. a=rtpmap:0 PCMU/8000. # U 2005/09/12 14:22:30.253088 a.b.c.d:5060 -> x.x.x.x:5060 ACK sip:x.x.x.x;ftag=c7ec14d4-13c4-43257d73-1c14942f-4823;lr=on SIP/2.0. Via: SIP/2.0/UDP a.b.c.d:5060. To: <sip:0123456789@sip.domain.com>;tag=38F09FAC-544. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257d73-1c14942f-4823. CSeq: 1 ACK. Call-ID: c7ec14d4-13c4-43257d73-1c14942f-29-15a7ee8. Contact: <sip:047132@a.b.c.d:5060>. Route: <sip:y.y.y.y;ftag=c7ec14d4-13c4-43257d73-1c14942f-4823;lr=on>. Route: <sip:0044123456789@a.a.a.a:5060>. Content-Length: 0. . # U 2005/09/12 14:22:30.254767 x.x.x.x:5060 -> y.y.y.y:5060 ACK sip:0044123456789@a.a.a.a:5060 SIP/2.0. Max-Forwards: 10. Record-Route: <sip:x.x.x.x;ftag=c7ec14d4-13c4-43257d73-1c14942f-4823;lr=on>. Via: SIP/2.0/UDP x.x.x.x;branch=0. Via: SIP/2.0/UDP a.b.c.d:5060. To: <sip:0123456789@sip.domain.com>;tag=38F09FAC-544. From: <sip:047132@a.b.c.d:5060>;tag=c7ec14d4-13c4-43257d73-1c14942f-4823. CSeq: 1 ACK. Call-ID: c7ec14d4-13c4-43257d73-1c14942f-29-15a7ee8. Contact: <sip:047132@a.b.c.d:5060>. Route: <sip:0044123456789@a.a.a.a:5060>. Content-Length: 0.

19 years, 2 months

[Serusers] help!how to make ser+rtpproxy can only realize nat traversal?

by 皮面

Hi,list I have found a configure file which can make ser+rtpproxy to support NAT traversal.But I don't want to use mysql and auth or some other modules which are not concerned with nathelper.In another words,I just want to use nathelper module to realize NAT traversal as a full proxy.I did some changes on this configure file now it can be registered but still cannot realize transaction between two client:( Could sombody help me or tell me which parameter should be change! thankes! http://mianpi.blogchina.com/inc/nat-rtpproxy.3.05.cfg --------------------------------- DO YOU YAHOO!? 雅虎邮箱超强增值服务－2G超大空间、pop3收信、无限量邮件提醒

19 years, 2 months

[Serusers] Looking for a way to register many numbers with just one Register Method

by Rodrigue ESTEVE

Hi all, I'd like to know if SER offers a way to register many phone numbers with just one REGISTER request. For example I've an IP gateway connected to the PBX and to the PSTN (for backup): Phones-----> PBX--------> IP GW --------> PSTN Backup | | IP Backbone ------> SER This PBX has a lot of phone numbers associated and a main phone number (receptionist for example). I would like to make this phone numbers registered on the proxy when the IP GW is running fine. But if the PBX has 100 numbers, the IP GW has to send 100 register requests to SER, moreover I'd like to register every 5-6 minutes so that the IP GW can use PSTN backup in case of failover of the WAN link. The problem is that I'm not sure that the IP GW will scale very well for a huge quantity of phone numbers (some customers have 700 phones numbers), and all these requests will consume bandwidth. For the moment I've done a Perl Script running each time an IP GW register one number (the receptionist number for example) using the exec_dset function (exec_dset("/root/sip_registration.pl $SIP_HF_TO $SIP_SRCIP");). This perl script looks into a database all the numbers related to the "receptionist" number and sends as many register information to the SER FIFO. But I'm not very satisfied because I've to create (not yet implemented) as many temporary FIFO receipt as I receive "receptionist" register to maintain the relationship between a register request and the FIFO output (SER FIFO receipt). Moreover it's not easy to be sure that all the information passed to the FIFO have been correctly used (I'm for sure not a good programmer). So I would like to know if there is a way in SER to do this (perhaps AVP ???) so I don't have to program a crappy perl script (see below for a first release of this perl script, if somebody know how I can create a temporary unix fifo in perl without using the system command: mkifo). Thanks, Rodrigue, Here's my script: #!/usr/bin/perl -w \ use DBI; use strict; use Fcntl; #Database arguments my $db = 'DBI:mysql:telephone:localhost'; my $username = 'root'; my $passwd = 'mo1000$$'; my $table = 'association'; #Error Log my $error_log = "/var/log/sip_registration.log"; #SER FIFO arguments my $fifo_method = "ul_add"; my $fifo_receipt = "ser_receipt_fifo"; my $fifo_table = "location"; my $fifo_expiration_timer = 60; my $fifo_qvalue = 0; my $fifo_replicate = 0; my $fifo_flags = 0; #Get the uri and ip to register my $uri = $ARGV[0]; my $ip = $ARGV[1]; #Get the phone number from the uri my ($phone_number)= $uri =~ /sip:(.*)@/; #Retrieve from database the numbers associated to the uri my $dbh = DBI->connect($db, $username, $passwd); my $sql = qq{ SELECT associated FROM $table where initnumber='$phone_number' }; my $sth = $dbh->prepare($sql); $sth->execute; #Bulk registration using FIFO open(my $wr,">", "/tmp/ser_fifo"); select((select($wr), $|=1)[0]); sysopen(my $read, "/tmp/$fifo_receipt", O_RDONLY | O_NONBLOCK); my $i=1; my $associated_phone; $sth->bind_col(1,\$associated_phone); while($sth->fetch()) { if ($associated_phone != $phone_number) { print $wr ":$fifo_method:$fifo_receipt\n$fifo_table\n$associated_phone\nsip:$associate d_phone\@$ip\n$fifo_expiration_timer\\ n$fifo_qvalue\n$fifo_replicate\n$fifo_flags\n" or die; my @read_fifo; while(<$read>){ $read_fifo[0]=$_; #print "$i $read_fifo[0]"; } $i++; } } #print "ASSOCIATED PHONE : $associated_phone"; # Wait till the loop see the last associated phone, the receipt FIFO has to #be completely synchronised between Input (SER FIFO) and Ouput (Temporary #receipt FIFO) , not sure that SER works in this way. while(<$read> !~ /sip:$associated_phone\@$ip/){ #DO NOTHING } close($read); close($wr); $sth->finish(); $dbh->disconnect(); #Return uri to STDOUT to be compliant with SER exec_dset function print "$uri";

19 years, 2 months

[Serusers] How to account for call forward?

by Chris St Denis

When I user has call forwarding enabled, I want them to be able to forward to a long distance number - at their own expense. Because of this, I want to account call forwarded numbers as originating as the callee and going to the destination long distance number. I would like to log this for local calls as well. How can I make this happen?

19 years, 2 months

[Serusers] Module creation

by sendman

Hi folks! I'm writing a new module for SER, which wil do routing task; well now I have a little problem. The module is do routing very well, using t_relay_to_udp function, but I don't now how to setup t_on_failure route directly inside the module... So is possible do something like: cmd_function t_on_failure_f; ... if (!(t_on_failure_f = find_export("t_on_failure",1,0))) { LOG(L_NOTICE, "mod_init(): This module requires tm module\n"); return -5; } ... t_on_failure_f(msg,failure); t_relay_to_udp_f(msg,gw,foo)); Best regards. -fs

19 years, 2 months

[Serusers] call tests

by harry gaillac

Please, Call sip:84@nxs.yi.org or 85 86 87 to debug my config file. Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

19 years, 2 months

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sr-users September 2005