sr-users

sr-users@lists.kamailio.org

19 participants
37529 discussions

[Users] permission.allow example
by Ion Minzu 05 Feb '07

05 Feb '07

Hello , Can someone describe me a real example how to write rules in permission.allow, permission.deny. I use the example from file permission.* and it doesn't work, permission is set ALL to ALL. Or maybe exist other method to make call restrictions? #openser.cfg .... loadmodule "/usr/lib/openser/modules/permissions.so" .... modparam("permissions", "default_deny_file", "/etc/openser/permissions.deny") modparam("permissions", "default_allow_file", "/etc/openser/permissions.allow") .... -- Best regards, Ion Minzu, mailto:ion.minzu@cts.md

2 1

[Serusers] Cisco AS 5300 can't send BYE for SER
by Sahria Hao 05 Feb '07

05 Feb '07

Hi all. I have a problem for the following scenario. When I make a call for PSTN and on hook by PSTN side, Cisco As can't send BYE for SER. 01. [UA via Asterisk] dialing "08022223333" -> [SER] 02. [SER] prefix("0333") and rewritehostport("my.cisco.ip.address:5060") -> [Cisco] 03. [Cisco] dial-peer voice 6000 pots, translate-outgoing called from "033308022223333" to "008022223333" 04. [Cisco] process an outbound call to "008022223333" -> [e.g. Mobile] 05. [e.g. Mobile] Catch call 06. [SER] log CDR start 07. [Cisco] talking 08. [e.g. Mobile] On hook and call disconnect 09. [Cisco] can't send BYE for SER *****why??***** 10. [UA via Asterisk] On hook 11. [UA via Asterisk] Send BYE for SER 12. [SER] log CDR End [Cisco] Call finished But another scenario, if make a call from PSTN to Asterisk and on hook by PSTN side, Cisco As send BYE to SER. 01. [e.g. Mobile] dialing "0377771111(Asterisk user number)" 02. [Cisco] receive "77771111" call number 03. [Cisco] dial-peer voice 5000 voip, session target ipv4: my.ser.ip.address -> [SER] 04. [SER] process an incoming call to "0377771111" -> [UA via Asterisk] 05. [UA via Asterisk] Catch call 06. [SER] log CDR start 07. [UA via Asterisk] talking 08. [e.g. Mobile] On hook and call disconnect 09. [Cisco] Send BYE to SER 10. [SER] log CDR End [Cisco] Call finished 11. [UA via Asterisk] receive BYE from SER And sorry for my diffucult example. Why Cisco AS 5300 can't send BYE to SER When PSTN call is disconnected by PSTN side? My ser.cfg as follows: # -------------------------------------------------------------------------- # global configuration parameters # -------------------------------------------------------------------------- fork=no log_stderror=yes check_via=no dns=no rev_dns=no listen=my.ser.ip.address port=5060 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" # -------------------------------------------------------------------------- # module loading # -------------------------------------------------------------------------- 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/textops.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/exec.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("usrloc", "db_url", " mysql://ser:heslo@localhost/ser") modparam("auth_db", "db_url", "mysql://ser:heslo@localhost/ser") modparam("permissions", "db_url", " mysql://ser:heslo@localhost /ser") modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 3) # -------------------------------------------------------------------------- # route pattern # -------------------------------------------------------------------------- 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()) { if (method=="ACK") { acc_db_request("01:CallStart\n", "acc"); }; if (method=="BYE" || method=="CANCEL") { acc_db_request("02:CallEnd\n", "acc"); }; t_relay(); break; }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; save("location"); break; }; if (search("^(f|From): .*(a)(my\.cisco\.ip\.address<.*(a)(my%5C.cisco%5C.ip%5C.address>)")) { #PSTN Incoming call from Cisco AS 5300 e.g. 0377771111 rewritehost("my.asterisk.ip.address "); }; lookup("aliases"); if (!lookup("location")) { if (method=="INVITE" && !search("^(f|From): .*(a)(my\.cisco\.ip\.address<.*(a)(my%5C.cisco%5C.ip%5C.address>)")) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; }; if (uri=~"^sip:0[0-9]{10}@") { # PSTN Outgoing call to Cisco AS 5300 e.g. 08022223333 prefix("0333"); rewritehostport("my.cisco.ip.address:5060"); avp_write("i:45", "inv_timeout"); } else { sl_send_reply("404", "Not Found"); break; }; consume_credentials(); }; }; }; if (!t_relay()) { sl_reply_error(); }; } And my Cisco AS 5300 config as follows: voice call send-alert voice rtp send-recv voice service pots fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco voice service voip fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco sip min-se 60 translation-rule 50 Rule 0 0333 0 Rule 1 ^7777 037777 voice class codec 2 codec preference 1 g711ulaw codec preference 2 g711alaw dial-peer voice 5000 voip tone ringback alert-no-PI description ser-asterisk-cisco-test huntstop destination-pattern 77771111$ translate-outgoing called 50 voice-class codec 2 session protocol sipv2 session target ipv4:my.ser.ip.address dtmf-relay rtp-nte max-conn 1 dial-peer voice 6000 pots application session max-conn 2 destination-pattern 0333T progress_ind alert enable 8 translate-outgoing called 50 port 0:D Thanks, Sahria

1 0

[Users] OpenSER devel & SNOM 360
by mika.saari＠wipsl.com 05 Feb '07

05 Feb '07

Hi, I do have OpenSER 1.1.0 with TLS support on Debian Sarge working nicely, and my SNOM 360 registers against it nicely. When using OpenSER 1.2 (2 days old CVS build) with TLS on Debian Etch with SNOM 360 and own signed certificates (with tls/tools & own machine names) I get debug info (below) from OpenSER. Any ideas with that "peer did not send sertificate" ? Thank you so many times, -Mika --- clip --- Feb 2 13:15:10 localhost openser[20855]: tcp_receive_timeout: 0xb59e1900 expired (296, 297) lt=0 Feb 2 13:15:10 localhost openser[20855]: DBG: io_watch_del (0x8122980, 18, -1, 0x10) fd_no=2 called Feb 2 13:15:10 localhost openser[20855]: releasing con 0xb59e1900, state 0, fd=18, id=2 Feb 2 13:15:10 localhost openser[20855]: extra_data (nil) Feb 2 13:15:10 localhost openser[20859]: handle_tcp_child: reader response= b59e1900, 0 from 0 Feb 2 13:15:10 localhost openser[20859]: DBG: io_watch_add(0x8122820, 23, 2, 0xb59e1900), fd_no=15 Feb 2 13:15:10 localhost openser[20859]: handle_tcp_child: CONN_RELEASE 0xb59e1900 refcnt= 0 Feb 2 13:15:10 localhost openser[20859]: tcpconn_new: new tcp connection to: 193.65.183.233 Feb 2 13:15:10 localhost openser[20859]: tcpconn_new: on port 2063, type 3 Feb 2 13:15:10 localhost openser[20859]: tls_tcpconn_init: Entered: Creating a whole new ssl connection Feb 2 13:15:10 localhost openser[20859]: tls_tcpconn_init: Looking up socket based TLS server domain [193.65.183.13:5061] Feb 2 13:15:10 localhost openser[20859]: tls_find_server_domain: virtual TLS server domain not found, Using default TLS server domain settings Feb 2 13:15:10 localhost openser[20859]: tls_tcpconn_init: Found socket based TLS server domain [0.0.0.0:0] Feb 2 13:15:10 localhost openser[20859]: tls_tcpconn_init: Setting in ACCEPT mode (server) Feb 2 13:15:10 localhost openser[20859]: tcpconn_add: hashes: 462, 9 Feb 2 13:15:10 localhost openser[20859]: handle_new_connect: new connection: 0xb5a06c88 24 flags: 0002 Feb 2 13:15:10 localhost openser[20859]: send2child: to tcp child 0 6(20855), 0xb5a06c88 Feb 2 13:15:10 localhost openser[20855]: received n=4 con=0xb5a06c88, fd=18 Feb 2 13:15:10 localhost openser[20855]: DBG: io_watch_add(0x8122980, 18, 2, 0xb5a06c88), fd_no=1 Feb 2 13:15:11 localhost openser[20855]: tls_update_fd: New fd is 18 Feb 2 13:15:11 localhost openser[20855]: tls_update_fd: New fd is 18 Feb 2 13:15:11 localhost openser[20855]: tls_accept: Error in SSL: Feb 2 13:15:11 localhost openser[20855]: tls_error: error:140890C7:SSL routines:SSL3_GET_CLIENT_CERTIFICATE:peer did not return a certificate Feb 2 13:15:11 localhost openser[20855]: DBG: io_watch_del (0x8122980, 18, -1, 0x10) fd_no=2 called Feb 2 13:15:11 localhost openser[20855]: releasing con 0xb5a06c88, state -2, fd=18, id=9 Feb 2 13:15:11 localhost openser[20855]: extra_data 0xb59f1a18 Feb 2 13:15:11 localhost openser[20859]: handle_tcp_child: reader response= b5a06c88, -2 from 0 Feb 2 13:15:11 localhost openser[20859]: tcpconn_destroy: destroying connection 0xb5a06c88, flags 0002 Feb 2 13:15:11 localhost openser[20859]: tls_close: Closing SSL connection Feb 2 13:15:11 localhost openser[20859]: tls_update_fd: New fd is 24 Feb 2 13:15:11 localhost openser[20859]: tls_shutdown: Shutdown successful Feb 2 13:15:11 localhost openser[20859]: tls_tcpconn_clean: Entered -- clip ---

2 2

[Users] Asterisk OpenSER and Exchange 2007 Unified Messaging
by Jon Webster 05 Feb '07

05 Feb '07

Greetings list, I'm in the process of configuring Asterisk <-> OpenSER <-> Microsoft Exchange 2007 Unified Messaging. I have a windows based proxy that works great, and want to accomplish the same feat using OpenSER. Below are network captures from the working proxy and OpenSER. From comparing the working call and the OpenSER call it looks like OpenSER needs to modify the CONTACT header in its last "OK" response. I'm not sure that's right, but more importantly I'm not sure how to do this in the config. Any help or direction is greatly appreciated. Thanks, -jon p.s. Please cc me on any replies, I'm not currently subscribed to the list. openSER unsuccessfull call negotiation =========== # U +9.215930 asterisk:5060 -> openser:5060 INVITE sip:8885555@openser SIP/2.0. Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK629bdb92;rport. From: "Jon Webster" <sip:3149@asterisk>;tag=as43593b57. To: <sip:8885555@openser>. Contact: <sip:3149@asterisk>. Call-ID: 46109877480231241e3b8e9d57036e69@asterisk. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Date: Sat, 03 Feb 2007 18:01:48 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 216. . v=0. o=root 381 381 IN IP4 asterisk. s=session. c=IN IP4 asterisk. t=0 0. m=audio 13614 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. # U +0.000305 openser:5060 -> asterisk:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK629bdb92;rport=5060. From: "Jon Webster" <sip:3149@asterisk>;tag=as43593b57. To: <sip:8885555@openser>. Call-ID: 46109877480231241e3b8e9d57036e69@asterisk. CSeq: 102 INVITE. Server: OpenSer (1.1.1-tls (i386/linux)). Content-Length: 0. Warning: 392 openser:5060 "Noisy feedback tells: pid=6890 req_src_ip=asterisk req_src_port=5060 in_uri=sip:8885555@openser out_uri=sip:5555@openser via_cnt==1". . # U +0.025313 openser:5060 -> asterisk:5060 SIP/2.0 180 Ringing. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as43593b57. TO: <sip:8885555@openser>;epid=BD-70-82-06-F9;tag=e38b6cdb9b. CSEQ: 102 INVITE. CALL-ID: 46109877480231241e3b8e9d57036e69@asterisk. MAX-FORWARDS: 70. VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK629bdb92;rport=5060. CONTENT-LENGTH: 0. SERVER: RTCC/2.0.6017.0. . # U +0.016314 openser:5060 -> asterisk:5060 SIP/2.0 200 OK. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as43593b57. TO: <sip:8885555@openser>;epid=BD-70-82-06-F9;tag=e38b6cdb9b. CSEQ: 102 INVITE. CALL-ID: 46109877480231241e3b8e9d57036e69@asterisk. MAX-FORWARDS: 70. VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK629bdb92;rport=5060. CONTACT: <sip:exchangeUM:5065;transport=Tcp;maddr=exchangeUM>. CONTENT-LENGTH: 197. CONTENT-TYPE: application/sdp. SERVER: RTCC/2.0.6017.0. . v=0. o=- 0 0 IN IP4 exchangeUM. s=Microsoft Exchange Speech Engine. c=IN IP4 exchangeUM. t=0 0. m=audio 59544 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. working proxy =================== # U +8.036801 asterisk:5060 -> goodproxy:5060 INVITE sip:5555@goodproxy SIP/2.0. Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK42383fff;rport. From: "Jon Webster" <sip:3149@asterisk>;tag=as07e96b4e. To: <sip:5555@goodproxy>. Contact: <sip:3149@asterisk>. Call-ID: 7d0450601533753042ce2d835a3f98d2@asterisk. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Date: Sat, 03 Feb 2007 18:02:40 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 216. . v=0. o=root 381 381 IN IP4 asterisk. s=session. c=IN IP4 asterisk. t=0 0. m=audio 17160 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. # U +0.003872 goodproxy:5060 -> asterisk:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK42383fff;rport;received=goodproxy. From: "Jon Webster" <sip:3149@asterisk>;tag=as07e96b4e. To: <sip:5555@goodproxy>. Call-ID: 7d0450601533753042ce2d835a3f98d2@asterisk. CSeq: 102 INVITE. User-Agent: M-Networks USR/1.0. Allow: INVITE, INFO, ACK, CANCEL, BYE, NOTIFY, BENOTIFY, SUBSCRIBE. Content-Length: 0. . # U +0.139238 goodproxy:5060 -> asterisk:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK42383fff;rport. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as07e96b4e. TO: <sip:5555@exchangeUM>;epid=BD-70-82-06-F9;tag=eb4f3cf30. CSEQ: 102 INVITE. CALL-ID: 7d0450601533753042ce2d835a3f98d2@asterisk. MAX-FORWARDS: 70. CONTENT-LENGTH: 0. SERVER: RTCC/2.0.6017.0. . # U +0.010224 goodproxy:5060 -> asterisk:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK42383fff;rport. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as07e96b4e. TO: <sip:5555@exchangeUM>;epid=BD-70-82-06-F9;tag=eb4f3cf30. CSEQ: 102 INVITE. CALL-ID: 7d0450601533753042ce2d835a3f98d2@asterisk. MAX-FORWARDS: 70. CONTACT: <sip:goodproxy:5060>. CONTENT-LENGTH: 197. CONTENT-TYPE: application/sdp. SERVER: RTCC/2.0.6017.0. . v=0. o=- 0 0 IN IP4 exchangeUM. s=Microsoft Exchange Speech Engine. c=IN IP4 exchangeUM. t=0 0. m=audio 32458 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. . # U +0.000894 asterisk:5060 -> goodproxy:5060 ACK sip:goodproxy:5060 SIP/2.0. Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK440651a7;rport. From: "Jon Webster" <sip:3149@asterisk>;tag=as07e96b4e. To: <sip:5555@goodproxy>;tag=eb4f3cf30. Contact: <sip:3149@asterisk>. Call-ID: 7d0450601533753042ce2d835a3f98d2@asterisk. CSeq: 102 ACK. User-Agent: Asterisk PBX. Max-Forwards: 70. Content-Length: 0. . # U +5.475898 asterisk:5060 -> goodproxy:5060 BYE sip:goodproxy:5060 SIP/2.0. Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK5fabb75e;rport. From: "Jon Webster" <sip:3149@asterisk>;tag=as07e96b4e. To: <sip:5555@goodproxy>;tag=eb4f3cf30. Call-ID: 7d0450601533753042ce2d835a3f98d2@asterisk. CSeq: 103 BYE. User-Agent: Asterisk PBX. Max-Forwards: 70. Content-Length: 0. . # U +0.028752 goodproxy:5060 -> asterisk:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK5fabb75e;rport. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as07e96b4e. TO: <sip:5555@exchangeUM>;tag=eb4f3cf30;epid=BD-70-82-06-F9. CSEQ: 103 BYE. CALL-ID: 7d0450601533753042ce2d835a3f98d2@asterisk. MAX-FORWARDS: 70. CONTENT-LENGTH: 0. SERVER: RTCC/2.0.6017.0. . # # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=4 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) /* 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) listen=openser port=5060 children=4 fifo="/tmp/openser_fifo" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify = 1 #tls_require_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/home/darilion/software/openser-1.0.1-cvs/sip-server/debian/openser/etc /openser/tls/user/user-cert.pem" #tls_private_key = "/home/darilion/software/openser-1.0.1-cvs/sip-server/debian/openser/etc /openser/tls/user/user-privkey.pem" #tls_ca_list = "/home/darilion/software/openser-1.0.1-cvs/sip-server/debian/openser/etc /openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/lib/openser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/textops.so" loadmodule "/usr/lib/openser/modules/uri.so" loadmodule "/usr/lib/openser/modules/nathelper.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/openser/modules/auth.so" #loadmodule "/usr/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- 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"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if (method=="INVITE" && uri=~"^sip:888[0-9]*@") { log("\n\nSIP invite received from exchange12\n\n"); strip(3); t_relay("tcp:exchangeUM:5065"); exit; }; # 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(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (uri=="sip:openser:5060;transport=TCP") { sl_send_reply("200", "OK"); exit; } if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay_to_tls("IP_domain1","port_domain1"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay_to_tls("IP_domain2","port_domain2"); # exit; #} route(1); }; # 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") { # Uncomment this if you want to use digest authentication #if (!www_authorize("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); # exit; #}; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } route[2] { t_on_reply("1"); }

2 1

[Users] Change a Status Code From 182 to 180
by Marc Schaefer 05 Feb '07

05 Feb '07

Hello all, I've a small problem that causes a lot of problems and is in some way specific: I've a Openser 1.1.1 Server running on a public rootserver. User are registered with FritzBox ( DSL-Router with Voip-Stack) or a Soft/Hardphone being Nated. For the PSTN connectivity i've access to a PSTN Gateway from the BT. Everything works very nice with just one problem remaining. When one User gets a call from PSTN and picks it up, no problems. But the problem comes when a second call is made to the same User. His phone tells him that somebody is knocking and he can pickup the phone without any problem. The problem is on the callers side because he doesn't get any feedback that he is ringing/knocking, so most of the time he will just cancel the call. I did some research about it and here are my conclusions: The INVITE from the second call isn't answered with a "180 Ringing.." but with a "182 Queued.." which doesn't trigger a Ringing for the PSTN Call. When I talked to the BT support they did not really know about the status code 182 and could not give me an answer how their mediagateway will handle it ( They promised me to get the informtion :-) ). On the other side I can't change the Firmware so there's no way of persuading the DSL-Router to use 180 instead of 182. So my questions are: Can i change the status code at the proxy? If so, in which part of the routing logic do I have to change it? I tried a lot of different subst() and replace() commands but none had any effect on the sip messages it always kept being 182. Below you can see my configuration which is based on the template from ixcall ( http://www.ixcall.net/?q=node/32 ) with a lot of changes and support from the "getting started" guide from onsip.org, of course adapted to the openser 1.1.1 version I have attached my config. Keep in mind that this config will get through a m4 parser before being ready to be used ( mechanism can be seen at ixcall ). I hope my english is good enough to have my problem explained in a understandable manner Have fun Marc /* ==================================================================== Schnuffle Openser Configuration V6.3 Based on IxCall template V2.4 (http://www.ixcall.net/?q=node/32) And OnSip.org "Getting Started" Guid (http://siprouter.onsip.org/doc/gettingstarted/) ==================================================================== list of flags used 1 - Accounting 2 - Missed calls Accouting 3 - NAT SIP ping flag 4 - NAT invite detected 5 - registrar module nat_flag 6 - Enable NAT support 7 - Enable Trunk (pstn, asterisk or yate) 13 - Enable debugging -------------------------------------------------------------------- Routes 1 - Relay with NAT helper and no proxy authentication 2 - Method Register handler 3 - Method Invite handler (call routing) 4 - Trunk and PSTN gateway routing 5 - TLS routing ( not really used but still in config to keep the given message flow intact ) 6 - Relay with no NAT support and no proxy authentication 7 - User verify 10 - Relay with NAT helper and proxy authentication Reply 1 - NAT helper Fail 1 - Voicemail and message store ( stripped from all the voicemail and message storage stuff, so in fact just empty ) -------------------------------------------------------------------- */ #Global settings debug=3 log_facility=LOG_LOCAL7 fork=yes log_stderror=no mhomed=no # Multihomed - set this to no if you set a "listen" addres listen=OPENSER_IP # only want to bind to this ip port=OPENSER_PORT children=4 tcp_children=4 tcp_accept_aliases=yes alias=OPENSER_REALM # your realm alias=OPENSER_HOST # alias domains of your SRV records alias=OPENSER_IP # and for the die hard ip users. # uncomment to override config values for debugging /* debug=7 # debug level (cmd line: -ddd) fork=no port=OPENSER_PORT log_stderror=yes # (cmd line: -E) fifo="OPENSER_FIFO" */ dns=no rev_dns=no fifo="OPENSER_FIFO" fifo_mode=OPENSER_FIFO_MODE #TLS specific settings disable_tls = DISABLE_TLS #listen = tls:OPENSER_TLS_IP:OPENSER_TLS_PORT #tls_verify = 1 #tls_require_certificate = 0 #tls_method = SSLv23 #tls_certificate = "DEFAULT_TLS_CERTIFICATE" #tls_private_key = "DEFAULT_TLS_PRIVATE_KEY" #tls_ca_list = "DEFAULT_TLS_CA_LIST" mpath="OPENSER_MODULE_PATH" #path to openser modules loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "acc.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "alias_db.so" loadmodule "uri.so" loadmodule "uri_db.so" loadmodule "mediaproxy.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "avpops.so" loadmodule "domain.so" loadmodule "permissions.so" loadmodule "group.so" loadmodule "xlog.so" loadmodule "enum.so" loadmodule "lcr.so" fifo_db_url="mysql://OPENSER_DB_USER:OPENSER_DB_PW@OPENSER_DB_HOST/OPENSER_DB" modparam("avpops|auth_db|alias_db|permissions|uri_db|usrloc|acc|msilo|group|domain|lcr|pdt","db_url", "mysql://OPENSER_DB_USER:OPENSER_DB_PW@OPENSER_DB_HOST/OPENSER_DB") modparam("auth_db", "calculate_ha1", CALCULATE_HA1) modparam("auth_db", "password_column", "PASSWORD_COLUMN") modparam("auth_db", "password_column_2", "ha1b") modparam("auth", "secret", "OPENSER_DIGEST_PW") #nathelper, mediaproxy or both can be used #we will only be using mediaproxy modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "MEDIAPROXY_SOCKET") #modparam("mediaproxy","sip_asymmetrics","MEDIAPROXY_SIP_ASYMMETRICS") #modparam("mediaproxy","rtp_asymmetrics","MEDIAPROXY_RTP_ASYMMETRICS") # -- lcr params -- modparam("lcr","gw_table","gw") modparam("lcr","gw_name_column","gw_name") modparam("lcr","ip_addr_column","ip_addr") modparam("lcr","port_column","port") modparam("lcr","uri_scheme_column","uri_scheme") modparam("lcr","transport_column","transport") modparam("lcr","grp_id_column","grp_id") modparam("lcr","db_mode",0) # -- group params -- modparam("group", "table", "grp") modparam("group", "use_domain", 0) # might be usefull for future expansions modparam("group", "user_column", "username") modparam("group", "domain_column", "domain") modparam("group", "group_column", "grp") # -- usrloc params -- modparam("usrloc", "db_mode", 1) modparam("usrloc", "desc_time_order", 1) # -- registrar params -- modparam("registrar", "sip_natping_flag", 3) modparam("registrar", "nat_flag", 5) modparam("registrar", "desc_time_order", 1) # -- rr params -- modparam("rr", "enable_full_lr", 1) # -- tm params -- #modparam("tm", "fr_timer", 20 ) modparam("tm", "fr_inv_timer", 90 ) #modparam("tm", "wt_timer", 20 ) modparam("tm", "via1_matching", 1) modparam("tm", "fr_inv_timer_avp", "s:inv_timeout") # -- permission params -- modparam("permissions", "db_mode", 0) modparam("permissions", "trusted_table", "trusted") # -- acc params -- modparam("acc", "db_flag", 1 ) modparam("acc", "db_missed_flag", 2) modparam("acc", "report_cancels", 1) modparam("acc", "log_flag", 1) # -- alias_db params -- modparam("alias_db", "user_column", "username") modparam("alias_db", "domain_column", "domain") modparam("alias_db", "alias_user_column", "alias_username") modparam("alias_db", "alias_domain_column", "alias_domain") #--enum params -- #modparam("enum", "domain_suffix", "e164.arpa.") # -- domain params -- modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") # -- avpops params -- # avpops is used to store the local prefix from each customer in subscriber table # and append it as soon as a local call is done (meaning a call matching [^0][0-9]*) modparam("avpops","db_scheme","prefix-scheme:username_col=username;domain_col=domain;value_col=prefix;value_type=string;table=subscriber") modparam("avpops","avp_table","subscriber") modparam("avpops","use_domain",0) modparam("avpops","username_column","username") modparam("avpops","domain_column","domain") modparam("avpops","avp_aliases","prefix=s:800") # ------------------------- request routing logic ------------------- # main routing logic route { # ----------------------------------------------------------------- # Configuration flags # ----------------------------------------------------------------- NAT_SUPPORT_FLAG_6 TRUNK_ENABLED_FLAG_7 DEBUG_FLAG_13 # changing return code 182 to 180 so that the calling phone will ring # does not work at all #if (status=="182") { # xlog("Changing Status Code from 182 to 180"); # subst('/182 Queued/180 Ringing/ig'); #} #xlog("Replacing Status 1820 with 180"); #replace_all("182 Queued","180 Ringing"); if (isflagset(13) && !is_method("ACK") && !is_method("REGISTER")) { #debug route route(13); }; # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { if (method!="ACK") { sl_send_reply("483", "Too Many Hops"); }; return; }; if (msg:len > max_len) { if (method!="ACK") { sl_send_reply("513", "Message Overflow"); }; return; }; if (status=="482") { #loop detection xlog("L_ERR", "OPENSER_NAME: $mi $rm $fu -> $ru status 482 Loop Detected\n"); return; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE") { record_route(); if (client_nat_test("3") && isflagset(6)) { add_rr_param(";nat=yes"); }; }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (isflagset(6)) { if (has_totag() && (method=="INVITE" || method=="ACK" || method=="BYE")) { setflag(1); # not really clear which of the setflags triggered the "BYE" and "CANCEL" to be accounted setflag(2); if (client_nat_test("3")||search("^Route:.*;nat=yes")){ xlog("L_NOTICE","Loose Route: Using MediaProxy"); use_media_proxy(); }; }; }; route(6); return; }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (isflagset(6)) { if (method=="BYE" || method=="CANCEL") { setflag(1); setflag(2); end_media_session(); }; }; # ----------------------------------------------------------------- # Unauthorized relay # ----------------------------------------------------------------- if (!is_uri_host_local()) { xlog("L_NOTICE","Prevent unauthorized Access is_uri_host_local=false, URI: $u, From: $fu -> Request: $ru\n"); if (is_from_local() || allow_trusted()) { # We prevent unautherised relays xlog("is_from_local or allow_trusted is true\n"); append_hf("P-hint: outbound\r\n"); route(10); # need auth } else { xlog("L_NOTICE","Uauthorized Relay: Reply a Forbidden\n"); sl_send_reply("403", "Forbidden"); }; return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri=~"sip:daemon@" ) { sl_send_reply("410", "daemon has gone for a beer"); return; }; # Calls matching means it is a local call without prefix # So prefix needs to be added expect for special numbers if (uri=~"sip:[1-9][0-9]*@" ) { # Check if uri contains special numbers ( 110,112 and some other 11[0-9]+ numbers ) # that should not be changed if (!uri=~"sip:11.*") { # add prefix to number # load user prefix and store it in $avp(s:800) # the prefix is stored in the subscriber table in an supplement attribute prefix if(!avp_db_load("$fu","$avp(s:800)/$prefix-scheme")) { xlog("Failed to load User Prefix"); } else {; # prefix R-URI xlog("Adding Prefix: $avp(s:800) to $ru"); subst_uri('/^sip:([0-9]+)@(.*)$/sip:$avp(s:800)\1@\2;orig_uri=\1-\2/i'); xlog("R-URI: $ru"); } } } if (method=="INVITE") { route(3); return; } else if (method=="ACK") { route(1); return; } else if (method=="BYE" || method=="CANCEL") { setflag(1); setflag(2); route(1); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); #alias_db_lookup("dbaliases"); if (uri!=myself) { xlog("L_NOTICE", "Lookup Aliases failed"); route(10); return; }; if (!lookup("location")) { xlog("L_NOTICE", "Lookup Location failed"); route(8); return; }; route(1); # Do not proxy authenticate - needed by system to send messages return; } route[1] { # ----------------------------------------------------------------- # Default Route relay Handler with NAT ENABLED # ----------------------------------------------------------------- if (isflagset(6)) { if (client_nat_test("3")) { setflag(4); force_rport(); fix_contact(); }; if (isflagset(4) || isflagset(5)) { xlog("L_NOTICE","Route(1) using Mediaproxy"); use_media_proxy(); }; t_on_reply("1"); }; route(6); # Relay the call return; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- # This cool register script came from http://openser.org/docs/scripting.html sl_send_reply("100", "Trying"); if (!route(7)) { # verify the user return(0); }; if (!search("^Contact:[ ]*\*") && client_nat_test("7") && isflagset(6)) { setflag(3); setflag(5); force_rport(); fix_contact(); }; if (!is_user_in("From", "noauth")) { #no authentication required if (!www_authorize("", "subscriber")) { if (isflagset(13)) { xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $fu Authorization for $fu ($si) requested\n"); }; www_challenge("", "0"); return(0); } else { if (isflagset(13)){ xlog("L_NOTICE","OPENSER_NAME: $mi route[$rm][2] \ $fu Authorization OK\n"); }; }; # Prohibit attempts to grab someone else's To address using valid credentials if (!check_to()) { if (isflagset(13)){ xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $fu -> $ru REPLY 401 - Unauthorized check_to() failed\n"); }; sl_send_reply("401", "Unauthorized - account and username must match"); return(0); }; consume_credentials(); }; if (!save("location")) { if (isflagset(13)) { xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $fu save(location) Failed\n"); }; sl_reply_error(); return(0); }else { if (isflagset(13)) { xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $fu save(location)OK\n"); }; }; return; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- setflag(1); #turn on accounting setflag(2); #also missed calls lookup("aliases"); #in the event that an alias resolves to another domain. #alias_db_lookup("dbaliases"); if (uri!=myself) { xlog("L_NOTICE","No Local Call Check Auth I: r-uri: <$ru> from-uri:<$fu>\n"); route(10); #check authentication return; }; if (!lookup("location")) { xlog("L_NOTICE","Lookup Location Failed: r-uri: <$ru> from-uri:<$fu>\n"); if (src_ip==OPENSER_PBX_IP || !isflagset(7)) { # loopback prevention route(8); return; } else { # if it's not here, go hassle asterisk. xlog("L_NOTICE","Trunk Routing: r-uri: <$ru> from-uri:<$fu>\n"); route(4); return; }; }; t_on_failure("1"); #call waiting,busy,DND,voicemail route(1); # local normal routing return; } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- avp_write("$avp(i:120)", "$avp(s:inv_timeout)"); xlog("L_NOTICE","Loading Gateways form Database"); if (!load_gws()){ xlog("L_ERR","****************************** LoadGateways Failed **********************"); sl_send_reply("500", "Internal server error, unable to load phone gateways"); return; }; if (!next_gw()) { xlog("L_NOTICE","******************** Didn'T found Any Gateways *************************"); sl_send_reply("503", "Service not available - No gateways"); return; }; # check if we route through BT # if so, change From Header domain to IP ( @callway-voip.de -> @212.65.6.4 ) # BT Gateway is in GatewayGroup 1 if (to_gw("2")) { xlog("L_NOTICE","Relaying to External Gateway, so rewriting From-HEADER:$fu\n"); #subst('/182 Queued/180 Ringing/ig'); #replace_all("182 Queued","180 Ringing"); subst('/From:([^@]*)@callway-voip.de/From:\1@212.65.6.4/ig'); }; #if (isflagset(13)){ xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][4] \ time [$Tf] Next Gateway $ruri_user_avp method <$rm> r-uri <$ru> 2nd via <$hdr(via[1])>\n"); #}; route(1); return; } route[5] { # ----------------------------------------------------------------- # TLS Handling # ----------------------------------------------------------------- # if (!is_uri_host_local()) { # if you have some interdomain connections via TLS # if(uri=~"@tls_domain1.net") { # t_relay_to_tls("IP_domain1","port_domain1"); # exit; # } else if(uri=~"@tls_domain2.net") { # t_relay_to_tls("IP_domain2","port_domain2"); # exit; # }; # }; if (!is_uri_host_local()) { remove_hf("Alert-Info"); if (method=="INVITE" && proto==tls) { append_hf("Alert-info: 1\r\n"); #cisco 7960 append_hf("Alert-info: Bellcore-dr4\r\n"); #cisco ATA # append_hf("Alert-info: http://foo.bar/x.wav\r\n"); #snom }; }; return; } route[6] { # ----------------------------------------------------------------- # Route relay Handler # ----------------------------------------------------------------- #subst('/182 Queued/180 Ringing/ig'); #replace_all("182 Queued","180 Ringing"); if (!t_relay()) { if (isflagset(6)) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; }; xlog("L_NOTICE","Route6: T_relay failed"); sl_reply_error(); }; return; } route[7] { # ----------------------------------------------------------------- # User verify # ----------------------------------------------------------------- # Check for known domain(s) if (!is_from_local()) { if (isflagset(13)){ xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $fu Unknown domain\n"); sl_send_reply("403", "Unknown domain"); }; return(0); }; # Check to see that the user has not been disabled if (is_user_in("From", "disabled")) { if (isflagset(13)){ xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $fu Your account has been disabled\n"); }; sl_send_reply("403", "Your account has been disabled"); return(0); }; if (is_user_in("To", "disabled")) { if (isflagset(13)) { xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $tu That account has been disabled\n"); }; sl_send_reply("403", "That account has been disabled"); return(0); }; return(1); } route[8] { # ----------------------------------------------------------------- # Handling of unavailable user # ----------------------------------------------------------------- # non-Voip -- just send "off-line" if (!(method=="INVITE" || method=="ACK" || method=="CANCEL")) { sl_send_reply("404", "Not Found"); return; }; t_newtran(); t_reply("404", "Not Found"); # we account missed incoming calls; previous statteful processing # guarantees that retransmissions are not accounted acc_db_request("404 missed call", "missed_calls"); return; } route[10] { # ----------------------------------------------------------------- # Default Message Handler with Proxy Authentication # ----------------------------------------------------------------- if(method=="ACK") { #these you never proxy authenticate route(1); return; }; if(method=="BYE" || method=="CANCEL") { #these you never proxy authenticate setflag(1); route(1); return; }; xlog("L_INFO", "Proxy auth $fd $dP destination:$du $dd $ds"); if (!route(7)) { #verify the user return(0); }; if (!is_user_in("From", "noauth")) { #no authentication required if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); }; # if (is_user_in("Credentials", "local")) { # Uncomment to use the group options route(1); # }else{ # sl_send_reply("403", "Denied, you are not allowed this route"); # }; return; } onreply_route[1] { # ----------------------------------------------------------------- # Return Route # ----------------------------------------------------------------- #replace_all("182 Queued","180 Ringing"); if ((isflagset(4) || isflagset(5)) && isflagset(6) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { xlog("L_NOTICE","OnReplyRoute(1) Using Mediaproxy"); use_media_proxy(); }; }; if (client_nat_test("1") && isflagset(6)) { fix_contact(); }; return; } failure_route[1] { # ----------------------------------------------------------------- # No answer, cancel, busy handling # ----------------------------------------------------------------- return; } route[13] { # ----------------------------------------------------------------- # Debuging info # ----------------------------------------------------------------- if (status=="100") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 100 Trying\n"); } else if (status=="180") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 180 Ringing\n"); } else if (status=="181") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 181 Call Is Being Forwarded\n"); } else if (status=="182") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 182 Queued\n"); } else if (status=="183") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 183 Session Progress\n"); } else if (status=="200") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 200 OK\n"); } else if (status=="202") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 202 Accepted\n"); } else if (status=="300") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 300 Multiple Choices $ct\n"); } else if (status=="301") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 301 Moved Permanently $ct\n"); } else if (status=="302") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 302 Moved Temporarily $ct\n"); } else if (status=="305") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 305 Use Proxy\n"); } else if (status=="380") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 380 Alternative Service\n"); } else if (status=="400") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 400 Bad Request\n"); } else if (status=="401") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 401 Unauthorized\n"); } else if (status=="402") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 402 Payment Required\n"); } else if (status=="403") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 403 Forbidden\n"); } else if (status=="404") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 404 Not Found\n"); } else if (status=="405") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 405 Method Not Allowed\n"); } else if (status=="406") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 406 Not Acceptable\n"); } else if (status=="407") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 407 Proxy Authentication Required\n"); } else if (status=="408") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 408 Request Timeout\n"); } else if (status=="409") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 409 Conflict\n"); } else if (status=="410") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 410 Gone\n"); } else if (status=="411") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 411 Length Required\n"); } else if (status=="412") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 412 Conditional Request Failed\n"); } else if (status=="413") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 413 Request Entity Too Large\n"); } else if (status=="414") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 414 Request-URI Too Long\n"); } else if (status=="415") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 415 Unsupported Media Type\n"); } else if (status=="416") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 416 Unsupported URI Scheme\n"); } else if (status=="420") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 420 Bad Extension\n"); } else if (status=="421") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 421 Extension Required\n"); } else if (status=="422") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 422 Session Interval Too Small\n"); } else if (status=="423") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 423 Interval Too Brief\n"); } else if (status=="429") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 429 Provide Referrer Identity\n"); } else if (status=="480") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 480 Temporarily Unavailable\n"); } else if (status=="481") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 481 Call Leg/Transaction Does Not Exist\n"); } else if (status=="482") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 482 Loop Detected\n"); } else if (status=="483") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 483 Too Many Hops\n"); } else if (status=="484") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 484 Address Incomplete\n"); } else if (status=="485") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 485 Ambiguous\n"); } else if (status=="486") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 486 Busy Here\n"); } else if (status=="487") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 487 Busy Here; Request Terminated\n"); } else if (status=="488") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 488 Not Acceptable Here\n"); } else if (status=="489") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 489 Bad Event\n"); } else if (status=="491") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 491 Request Pending\n"); } else if (status=="493") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 493 Undecipherable\n"); } else if (status=="494") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 494 Security Agreement Required\n"); } else if (status=="500") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 500 Server Internal Error\n"); } else if (status=="501") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 501 Not Implemented\n"); } else if (status=="502") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 502 Bad Gateway\n"); } else if (status=="503") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 503 Service Unavailable\n"); } else if (status=="504") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 504 Server Timeout\n"); } else if (status=="505") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 505 Version Not Supported\n"); } else if (status=="513") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 513 Message Too Large\n"); } else if (status=="580") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 580 Precondition Failure\n"); } else if (status=="600") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 600 Busy Everywhere\n"); } else if (status=="603") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 603 Decline\n"); } else if (status=="604") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 604 Does Not Exist Anywhere\n"); } else if (status=="606") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 606 Not Acceptable\n"); } else { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status not listed\n"); }; xlog("L_NOTICE", "OPENSER_NAME: $mi $mb\n"); return; }

2 1

[Serusers] jabber2foreign IM
by Allan Chao ( 策略性新技術開發處申請 ) 05 Feb '07

05 Feb '07

Dear All: I am a ser user, i want to use sip client that can commuicate with foreign IM client(msn), my flow is sip client -> ser(jabber module) (0.10 version) -> jabber server(has msn gateway) -> MSN network. but i did not see any function that can register to MSN server(just like jabber client browse transport in the jabber server and then register to MSN network) and there is no function or program using msn id and password. i doubt jabber module does not implement this part of program.Maybe there is another mathod can solve it. Does anyone can help me ? thanks. allan DISCLAIMER: Sample Disclaimer added in a VBScript.

2 1

[Users] Exec - environment variables - not found?
by Kjell Arild Frigstad 04 Feb '07

04 Feb '07

Hello! I have just upgraded from 1.0.0 til 1.1.1, and I having problem with exec_msg (the exec_msg working, but environment variabels is "not found".) It was OK in 1.0.0 version. The coding: exec_msg("echo SIP_SRCIP $SIP_SRCIP >>/var/log/messages"); At startup (with debuggging) : 0(0) fixing exec_msg, line 160 0(0) xl_parse_format: parsing [echo SIP_SRCIP $SIP_SRCIP >>/var/log/messages] 0(0) xl_lookup_spec_name: not found [SIP_SRCIP] 0(0) xl_parse_format: format parsed OK: [2] items Error message when running the command (with debugging) : 0(28888) xl_get_spec_value: error - null sp->itf 0(28888) exec:w_exec_msg: executing [echo SIP_SRCIP >>/var/log/messages] Error message when running in "normal" mode Feb 4 21:51:24 test /usr/local/sbin/openser[28947]: xl_get_spec_value: error - null sp->itf SIP_SRCIP As you see the text SIP_SRCIP is beeing displayd, but not the variabel. Any ide what can bee wrong? Best regards K.A.Frigstad

2 1

[Users] auth_radius extra attributes
by Cseke Tamas 03 Feb '07

03 Feb '07

Hello, can anybody tell me how can i send extra attributes to radius in authpacket, in acctpacket i can with radius_extra parameter, is there a way to use it in auth too? Thanks any help, Tamas

1 0

[Serusers] Problem with mysql.so on new server
by sip 03 Feb '07

03 Feb '07

So... I've had ser 0.9.6 running on RH EL3, and FC3 and 4. I'm building out a server on RH EL4 for SER, but I've run into a snag. I'm sure it's something simple, but it has me stumped at the moment. I've recompiled mysql.so on the new server... compiles fine (as do the rest of the modules). However, when I try to start up SER with mysql.so loaded, it gives me this: ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: /usr/local/lib/ser/modules/mysql.so: undefined symbol: log Undefined symbol: log Never seen this before. Anyone know what it means or, more importantly, how to fix it (and no... I'm not going to scrap it all and go with Ottendorf yet, so that's not a solution ;) )? N.

4 12

[Serusers] Which locking mechanism is useful for usrloc in production environment?
by Ramu Yadav 03 Feb '07

03 Feb '07

Hello, At present we are using "sys v semaphores"(USE_SYSV_SEM) for locking. But we see a some delay while acquiring the lock to modify the usrloc's location table. We are in a thought to use "posix semaphores" (USE_POSIX_SEM), will this improve any performance for usrloc location table. Thoughts and Ideas will be appreciated. -- Ramu Yadav

4 5

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users