- sr-users - kamailio.org

[Serusers] SER and Broadsoft
by Arun Kumar 10 Dec '07

10 Dec '07

Hi, I'm trying to configure my ser with broadsoft but to send calls to broadsoft it need to sent via a user no ip based auth. so, is there any why that I can send username/pass from ser to send calls to my broadsoft. thanks arun

2 2

[Serusers] Is SER for me ?
by Charalampos Alexopoulos 09 Dec '07

09 Dec '07

Hi Sorry if this is a stupid question, but before i dive to SIP protocol and SER configuration i would like to know if SER doing what i need. What i want to do is to make voip phone calls with my friends without to use some voip provider like skype. Is the SIP the correct think for this ? Is the SER implementation suitable ? Thank you in advance Charalampos Alexopoulos

3 2

[OpenSER-Users] configure openser with cpl module
by Martin Kardi 08 Dec '07

08 Dec '07

Hi, I installed openser with cpl module and other dependencies but I don't know how to configure it? Openser is functionally without cpl module. I am beginner in this. Could somebody help me how to? thanks Martin -- View this message in context: http://www.nabble.com/configure-openser-with-cpl-module-tf4942775.html#a141… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 2

[OpenSER-Users] loose_route not setting $du ??
by Patrick Baker 07 Dec '07

07 Dec '07

I'm having an issue with messages flowing between asterisk and openser and I believe the issue is related to loose_route and having nulls for $du $dd $ds. Does anyone know how to resolve this with my config? I have also referenced my asterisk config along with some debug information. Thanks in advance for anyone that can help! Asterisk ends up dropping the call after 20 seconds as it appears that openSer isnt responding to asterisk OK message. Also BYE's and others arent working as well.. [Dec 7 04:45:55] WARNING[25642]: chan_sip.c:2334 retrans_pkt: Maximum retries exceeded on transmission 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 for seqno 8992 (Critical Response) [Dec 7 04:45:55] WARNING[25642]: chan_sip.c:2361 retrans_pkt: Hanging up call 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 - no reply to our critical packet. Really destroying SIP dialog '2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115' Method: INVITE ######################################################################## # This configuration is autogenerated by sip:wizard # (http://www.sipwise.com/wizard) on Fri Dec 07 05:52:34 +0100 2007 # for OpenSER 1.2 # # Copyright (C) 2007 Sipwise (support(a)sipwise.com) ######################################################################## ######################################################################## # By obtaining, using, and/or copying this configuration and/or its # associated documentation, you agree that you have read, understood, # and will comply with the Terms of Usage provided at # http://www.sipwise.com/news/?page_id=6 as well as the following # additions: # # Permission to use, copy, modify, and distribute this configuration and # its associated documentation for any purpose and without fee is hereby # granted, provided that the above copyright notice appears in all # copies, and that both that copyright notice and this permission notice # appear in supporting documentation, and that the name of Sipwise or # the author will not be used in advertising or publicity pertaining to # distribution of the configuration without specific, written prior # permission. ######################################################################## ######################################################################## # Before using this configuration, read the following prerequisites in # order to gain the designated functionallity: # # base: # You have to insert all locally served domains (i.e. # "openserctl domain add your.domain.com"). # # nat-rtpproxy: # You have to install RTPProxy # (http://www.openser.org/downloads/snapshots/rtpproxy/) for relaying # RTP traffic. # # offnet-pstn: # You have to add a routing entry for lcr (i.e. "openserctl lcr # addroute '' '' 1 1"). Additionally, you have to add your gateways # (i.e. "openserctl lcr addgw my-test-gw 1.2.3.4 5060 sip udp 1"). # ######################################################################## ######################################################################## # Configuration 'sip:wizard - Fri Dec 07 05:52:34 +0100 2007' ######################################################################## listen = udp:127.0.0.1:5060 listen = udp:10.3.1.31:5060 mpath = "/usr/local/lib/openser/modules" children = 8 debug = 3 fork = yes group = "openser" user = "openser" disable_tcp = no log_facility = LOG_DAEMON log_stderror = no tcp_children = 4 mhomed = no server_signature = yes sock_group = "openser" sock_mode = 0600 sock_user = "openser" unix_sock = "/tmp/openser.sock" unix_sock_children = 1 reply_to_via = no sip_warning = no check_via = no dns = no rev_dns = no disable_core_dump = no dns_try_ipv6 = yes dns_use_search_list = yes loadmodule "usrloc.so" modparam("usrloc", "user_column", "username") modparam("usrloc", "domain_column", "domain") modparam("usrloc", "contact_column", "contact") modparam("usrloc", "expires_column", "expires") modparam("usrloc", "q_column", "q") modparam("usrloc", "callid_column", "callid") modparam("usrloc", "cseq_column", "cseq") modparam("usrloc", "methods_column", "methods") modparam("usrloc", "flags_column", "flags") modparam("usrloc", "user_agent_column", "user_agent") modparam("usrloc", "received_column", "received") modparam("usrloc", "socket_column", "socket") modparam("usrloc", "use_domain", 0) modparam("usrloc", "desc_time_order", 0) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("usrloc", "db_mode", 1) modparam("usrloc", "matching_mode", 0) modparam("usrloc", "cseq_delay", 20) modparam("usrloc", "nat_bflag", 6) loadmodule "textops.so" loadmodule "rr.so" modparam("rr", "enable_full_lr", 0) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 1) modparam("rr", "add_username", 0) loadmodule "tm.so" modparam("tm", "fr_timer", 30) modparam("tm", "fr_inv_timer", 120) modparam("tm", "wt_timer", 5) modparam("tm", "delete_timer", 2) modparam("tm", "noisy_ctimer", 0) modparam("tm", "ruri_matching", 1) modparam("tm", "via1_matching", 1) modparam("tm", "unix_tx_timeout", 2) modparam("tm", "restart_fr_on_each_reply", 1) modparam("tm", "pass_provisional_replies", 0) loadmodule "xlog.so" modparam("xlog", "buf_size", 4096) modparam("xlog", "force_color", 0) loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("mi_fifo", "fifo_mode", 0660) modparam("mi_fifo", "fifo_group", "openser") modparam("mi_fifo", "fifo_user", "openser") modparam("mi_fifo", "reply_dir", "/tmp/") modparam("mi_fifo", "reply_indent", "\t") loadmodule "domain.so" modparam("domain", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 60) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "rtpproxy_disable", 0) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(i:801)") loadmodule "sl.so" modparam("sl", "enable_stats", 1) loadmodule "uri.so" loadmodule "registrar.so" modparam("registrar", "default_expires", 3600) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 0) modparam("registrar", "default_q", 0) modparam("registrar", "append_branches", 1) modparam("registrar", "case_sensitive", 0) modparam("registrar", "received_param", "received") modparam("registrar", "max_contacts", 0) modparam("registrar", "retry_after", 0) modparam("registrar", "method_filtering", 0) modparam("registrar", "path_mode", 2) modparam("registrar", "path_use_received", 0) modparam("registrar", "received_avp", "$avp(i:801)") loadmodule "maxfwd.so" modparam("maxfwd", "max_limit", 256) loadmodule "mysql.so" modparam("mysql", "ping_interval", 300) modparam("mysql", "auto_reconnect", 1) loadmodule "auth.so" modparam("auth", "nonce_expire", 300) modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes") modparam("auth", "rpid_avp", "$avp(s:rpid)") loadmodule "auth_db.so" modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "password_column", "password") modparam("auth_db", "password_column_2", "ha1b") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "use_domain", 0) modparam("auth_db", "load_credentials", "rpid") loadmodule "uri_db.so" modparam("uri_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("uri_db", "uri_table", "uri") modparam("uri_db", "uri_user_column", "username") modparam("uri_db", "uri_domain_column", "domain") modparam("uri_db", "uri_uriuser_column", "uri_user") modparam("uri_db", "subscriber_table", "subscriber") modparam("uri_db", "subscriber_user_column", "username") modparam("uri_db", "subscriber_domain_column", "domain") modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "use_domain", 0) loadmodule "lcr.so" modparam("lcr", "db_url", "mysql://openser:openserrw@localhost/openser") 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", "lcr_table", "lcr") modparam("lcr", "strip_column", "strip") modparam("lcr", "prefix_column", "prefix") modparam("lcr", "from_uri_column", "from_uri") modparam("lcr", "priority_column", "priority") modparam("lcr", "gw_uri_avp", "1400") modparam("lcr", "ruri_user_avp", "1402") modparam("lcr", "contact_avp", "1401") modparam("lcr", "fr_inv_timer_avp", "s:fr_inv_timer_avp") modparam("lcr", "fr_inv_timer", 90) modparam("lcr", "fr_inv_timer_next", 30) modparam("lcr", "rpid_avp", "s:rpid") ######################################################################## # Request route 'main' ######################################################################## route[0] { xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); force_rport(); if(msg:len > max_len) { xlog("L_INFO", "Message too big - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("513", "Message Too Big"); exit; } if (!mf_process_maxfwd_header("10")) { xlog("L_INFO", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("483", "Too Many Hops"); exit; } if(!is_method("REGISTER")) { if(nat_uac_test("19")) { record_route(";nat=yes"); } else { record_route(); } } if(is_method("CANCEL") || is_method("BYE")) { unforce_rtp_proxy(); } if(loose_route()) { if(!has_totag()) { xlog("L_INFO", "Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; } if(nat_uac_test("19") || search("^Route:.*;nat=yes")) { fix_nated_contact(); setbflag(6); } route(3); } if(is_method("REGISTER")) { route(2); } if(is_method("INVITE")) { route(4); } if(is_method("CANCEL") || is_method("ACK")) { route(8); } route(9); } ######################################################################## # Request route 'stop-rtp-proxy' ######################################################################## route[1] { if(isflagset(22)) { unforce_rtp_proxy(); } } ######################################################################## # Request route 'base-route-register' ######################################################################## route[2] { sl_send_reply("100", "Trying"); if(!www_authorize("", "subscriber")) { xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("", "0"); exit; } if(!check_to()) { xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed To-URI Detected"); exit; } consume_credentials(); if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) { fix_nated_register(); setbflag(6); } if(!save("location")) { xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; } xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Request route 'base-outbound' ######################################################################## route[3] { if(isbflagset(6)) { if(!isflagset(22) && !search("^Content-Length:[ ]*0")) { setflag(22); force_rtp_proxy(); } t_on_reply("2"); } else { t_on_reply("1"); } if(!isflagset(21)) { t_on_failure("2"); } if(isflagset(29)) { append_branch(); } if(is_present_hf("Proxy-Authorization")) { consume_credentials(); } xlog("L_INFO", "Request leaving server, D-URI='$du' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # no 100 (we already sent it) and no DNS blacklisting if(!t_relay("0x05")) { sl_reply_error(); if(is_method("INVITE") && isbflagset(6)) { unforce_rtp_proxy(); } } exit; } ######################################################################## # Request route 'base-route-invite' ######################################################################## route[4] { sl_send_reply("100", "Trying"); if(from_gw()) { xlog("L_INFO", "Call from PSTN' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); setflag(23); } else { if(!proxy_authorize("", "subscriber")) { xlog("L_INFO", "Proxy authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); proxy_challenge("", "0"); exit; } if(!check_from()) { xlog("L_INFO", "Spoofed From-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed From-URI Detected"); exit; } } if(nat_uac_test("19")) { fix_nated_contact(); setbflag(6); } route(5); } ######################################################################## # Request route 'invite-find-callee' ######################################################################## route[5] { if(!is_domain_local("$rd")) { setflag(20); route(7); } if(does_uri_exist()) { xlog("L_INFO", "Callee is local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(6); } else { xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); } exit; } ######################################################################## # Request route 'invite-to-internal' ######################################################################## route[6] { if(!lookup("location")) { xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Offline"); } else { xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(3); } exit; } ######################################################################## # Request route 'invite-to-external' ######################################################################## route[7] { if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(3); exit; } if(!isflagset(23)) { # don't allow calls relaying from PSTN to PSTN, if not explicitely forwarded if(uri =~ "^sip:[0-9]+@") { # only route numeric users to PSTN if(!load_gws()) { xlog("L_ERR", "Error loading PSTN gateways - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } if(!next_gw()) { xlog("L_ERR", "No PSTN gateways available - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } setflag(21); t_on_failure("1"); route(3); } } xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Not Found"); exit; } ######################################################################## # Request route 'base-route-local' ######################################################################## route[8] { t_on_reply("1"); if(t_check_trans()) { xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!t_relay()) { sl_reply_error(); } } else { xlog("L_INFO", "Dropping mis-routed request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } exit; } ######################################################################## # Request route 'base-route-generic' ######################################################################## route[9] { xlog("L_INFO", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("501", "Method Not Supported Here"); exit; } ######################################################################## # Request route 'base-filter-failover' ######################################################################## route[10] { if(!t_check_status("408|500|503")) { xlog("L_INFO", "No failover routing needed for this response code - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(1); exit; } } ######################################################################## # Reply route 'base-standard-reply' ######################################################################## onreply_route[1] { xlog("L_INFO", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Reply route 'base-nat-reply' ######################################################################## onreply_route[2] { xlog("L_INFO", "NAT-Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); if(nat_uac_test("1")) { fix_nated_contact(); } if(isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if(!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); } } exit; } ######################################################################## # Failure route 'pstn-failover' ######################################################################## failure_route[1] { xlog("L_INFO", "Failure route for PSTN entered - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(10); if(!next_gw()) { xlog("L_ERR", "Failed to select next PSTN gateway - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(1); exit; } t_on_failure("1"); route(3); } ######################################################################## # Failure route 'base-standard-failure' ######################################################################## failure_route[2] { route(10); route(1); } ##asterisk sip.conf## [general] matchexterniplocally=yes canreinvite=no externip=xxx.206.xxx.136 localnet=10.3.1.0/255.255.255.0 context=default bindport=5061 bindaddr=0.0.0.0 sipdebug=yes nat=yes [openser] type=friend context=default insecure=very externalnotify=yes allow=all ##ser log## Dec 7 04:04:52 phonesys-slave openser[24602]: Request leaving server, D-URI='<null>' - M=INVITE RURI=sip:500@10.3.1.31:5061;transport=udp F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:04:52 phonesys-slave openser[24597]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:04:53 phonesys-slave openser[24602]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:04:54 phonesys-slave openser[24597]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:04:56 phonesys-slave openser[24602]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:05:00 phonesys-slave openser[24597]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:05:04 phonesys-slave openser[24602]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:05:08 phonesys-slave openser[24597]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:05:12 phonesys-slave openser[24589]: Request leaving server, D-URI='<null>' - M=BYE RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:05:13 phonesys-slave openser[24599]: Request leaving server, D-URI='<null>' - M=BYE RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:05:16 phonesys-slave openser[24597]: Request leaving server, D-URI='<null>' - M=BYE RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 ##asterisk sip debug## <--- SIP read from 10.3.1.31:5060 ---> INVITE sip:500@10.3.1.31:5061;transport=udp SIP/2.0 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31> Contact: <sip:pbaker2@10.3.1.115:5060> Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE Max-Forwards: 69 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 252 v=0 o=pbaker2 3001829617 3001829777 IN IP4 10.3.1.115 s=X-Lite c=IN IP4 10.3.1.31 t=0 0 m=audio 35128 RTP/AVP 3 97 101 a=rtpmap:3 gsm/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=nortpproxy:yes <-------------> --- (13 headers 12 lines) --- Sending to 10.3.1.31 : 5060 (no NAT) Using INVITE request as basis request - 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 No user 'pbaker2' in SIP users list Found peer 'openser' for 'pbaker2' from 10.3.1.31:5060 Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 10.3.1.31:35128 Found audio description format gsm for ID 3 Found audio description format speex for ID 97 Found audio description format telephone-event for ID 101 Capabilities: us - 0x27f9fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|t140), peer - audio=0x202 (gsm|speex)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x202 (gsm|speex) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.3.1.31:35128 Looking for 500 in default (domain 10.3.1.31) list_route: hop: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> <--- Transmitting (no NAT) to 10.3.1.31:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31> Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@167.206.216.136> Content-Length: 0 <------------> Audio is at 167.206.216.136 port 37712 Adding codec 0x2 (gsm) to SDP Adding codec 0x200 (speex) to SDP Adding non-codec 0x1 (telephone-event) to SDP phonesys-slave*CLI> <--- Reliably Transmitting (no NAT) to 10.3.1.31:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31>;tag=as70a2356d Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@167.206.216.136> Content-Type: application/sdp Content-Length: 300 v=0 o=root 87430933 87430933 IN IP4 167.206.216.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 167.206.216.136 t=0 0 m=audio 37712 RTP/AVP 3 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> Retransmitting #1 (no NAT) to 10.3.1.31:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31>;tag=as70a2356d Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@167.206.216.136> Content-Type: application/sdp Content-Length: 300 v=0 o=root 87430933 87430933 IN IP4 167.206.216.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 167.206.216.136 t=0 0 m=audio 37712 RTP/AVP 3 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #2 (no NAT) to 10.3.1.31:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31>;tag=as70a2356d Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@167.206.216.136> Content-Type: application/sdp Content-Length: 300 v=0 o=root 87430933 87430933 IN IP4 167.206.216.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 167.206.216.136 t=0 0 m=audio 37712 RTP/AVP 3 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #3 (no NAT) to 10.3.1.31:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31>;tag=as70a2356d Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@167.206.216.136> Content-Type: application/sdp Content-Length: 300 v=0 o=root 87430933 87430933 IN IP4 167.206.216.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 167.206.216.136 t=0 0 m=audio 37712 RTP/AVP 3 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #4 (no NAT) to 10.3.1.31:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31>;tag=as70a2356d Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@167.206.216.136> Content-Type: application/sdp Content-Length: 300 v=0 o=root 87430933 87430933 IN IP4 167.206.216.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 167.206.216.136 t=0 0 m=audio 37712 RTP/AVP 3 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #5 (no NAT) to 10.3.1.31:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31>;tag=as70a2356d Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@167.206.216.136> Content-Type: application/sdp Content-Length: 300 v=0 o=root 87430933 87430933 IN IP4 167.206.216.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 167.206.216.136 t=0 0 m=audio 37712 RTP/AVP 3 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 7 04:45:50] NOTICE[25642]: rtp.c:998 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 10.3.1.31 Retransmitting #6 (no NAT) to 10.3.1.31:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31>;tag=as70a2356d Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@167.206.216.136> Content-Type: application/sdp Content-Length: 300 v=0 o=root 87430933 87430933 IN IP4 167.206.216.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 167.206.216.136 t=0 0 m=audio 37712 RTP/AVP 3 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 7 04:45:55] WARNING[25642]: chan_sip.c:2334 retrans_pkt: Maximum retries exceeded on transmission 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 for seqno 8992 (Critical Response) [Dec 7 04:45:55] WARNING[25642]: chan_sip.c:2361 retrans_pkt: Hanging up call 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 - no reply to our critical packet. Really destroying SIP dialog '2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115' Method: INVITE

2 2

[OpenSER-Users] starting openser trunk version
by Daniel 07 Dec '07

07 Dec '07

Hi all, I'm using the latest trunk version of openser. When I start the proxy I see on debug this messages repeated forever... I'm using a config file that worked well wiht openser release 1.2.2. Dec 6 15:55:14 [28924] DBG:core:qm_free: params(0x817ada0, 0x81caa24), called from nathelper.c: nh_timer(2941) Dec 6 15:55:14 [28924] DBG:core:qm_free: freeing frag. 0x81caa0c alloc'ed from nathelper.c: nh_timer(2841) Dec 6 15:55:15 [28924] DBG:core:qm_malloc: params (0x817ada0, 8), called from nathelper.c: nh_timer(2841) Dec 6 15:55:15 [28924] DBG:core:qm_malloc: params (0x817ada0, 8), returns address 0x81caa24 frag. 0x81caa0c (size=8) on 1 -th hit Can you help me? what does it mean? Regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

4 3

[Serusers] SER From Header
by Arun Kumar 07 Dec '07

07 Dec '07

Hi, I've one exten on my ser box and one exten on Broadsoft switch. From SER I'm not able to dial my broadsoft exten, b'coz broadsoft expect known user in >From filed but I'm calling from SER so the From filed shows my SER user not my broadsoft. So how do I change my From header when I call my broadsoft. Im using ser-0.9.6 version. Please help. Thanks, Arun

1 0

[OpenSER-Users] Accounting issues
by David Villasmil Govea 07 Dec '07

07 Dec '07

Hello all, I finally got accounting on db to work. I added a few extra fields and its all fine. But i have some questions: I added: modparam("acc", "db_extra", "from_uri=$fu ; to_uri=$tu ; from_user=$fU; source_ip=$si") to me config, and all the info gets loaded. BUT, source_ip is not always the ip from which the request was received, either that, or INVITES are not getting logged! well, that's really 2 questions in one... ;) here's my config: # # sample config file to be used with nathelper/rtpproxy # # start RTPProxy with: rtpproxy -l your_public_ip -s udp:localhost:8899 # # ----------- global configuration parameters ------------------------ debug=2 # 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) port=5060 children=20 disable_dns_blacklist=true # # --- module loading mpath="/usr/lib/openser/modules/" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "nathelper.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "mi_fifo.so" loadmodule "dispatcher.so" #loadmodule "domain.so" loadmodule "xlog.so" loadmodule "/usr/lib/openser/modules/acc.so" modparam("acc", "early_media", 1) modparam("acc", "report_cancels", 1) modparam("acc", "db_flag", 2) modparam("acc", "db_url", "mysql://openser:password@localhost/openser") modparam("acc", "report_ack", 0) modparam("acc", "db_missed_flag", 3) modparam("acc", "failed_transaction_flag", 4) modparam("acc", "db_extra", "from_uri=$fu ; to_uri=$tu ; from_user=$fU; source_ip=$si") # # CONSTANTS # # --- setting module parameters #modparam("domain","db_url","mysql://openser:password@localhost/openser") #modparam("domain", "db_mode", 1) #modparam("domain", "domain_table", "domain") #modparam("domain", "domain_col", "domain") modparam("xlog", "buf_size", 8192) # --- setting module parameters modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("usrloc|auth_db","db_url","mysql://openser:password@localhost /openser") # -- usrloc params -- modparam("usrloc", "db_mode", 1) modparam("usrloc", "nat_bflag", 6) #modparam("usrloc", "use_domain", 0) # -- registrar params -- modparam("registrar|nathelper", "received_avp", "$avp(i:42)") # -- auth params -- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") #modparam("auth_db", "use_domain", 0) # -- rr params -- #modparam("rr", "enable_full_lr", 1) # -- nathelper modparam("nathelper", "rtpproxy_sock", "udp:1.2.3.4:8899") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@sip.whatever.com") #modparam("registrar", "max_contacts", 1) # --- main routing logic route{ if (method=="INVITE") { xlog("L_NOTICE", "INVITE MESSAGE RECEIVED - START ACC\n"); setflag(2); setflag(3); setflag(4); }; if (method=="BYE") { xlog ("L_NOTICE", "BYE - STOP ACCOUNTING\n"); setflag(2); setflag(3); setflag(4); }; if (method=="CANCEL") { xlog ("L_NOTICE", "CANCEL - STOP ACCOUNTING\n"); setflag(2); setflag(3); setflag(4); }; if (method=="REGISTER") { setflag(2); setflag(3); setflag(4); }; if (!mf_process_maxfwd_header("100")) { sl_send_reply("483","Too Many Hops"); exit; }; #if (msg:len >= 2048 ) { # sl_send_reply("513", "Message too big"); # exit; #}; # NAT detection route(2); if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # if (!is_uri_host_local() ) { append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { xlog("L_NOTICE", "\nNew Registration\n"); if (!www_authorize("sip.whatever.com", "subscriber")) { xlog("L_NOTICE", "\nRegistration FAILED! CLIENT: $au\n"); www_challenge("sip.whatever.com", "0"); exit; }; if (isflagset(5)) { xlog("L_NOTICE", "\nRegistration OK\n"); setbflag(6); # if you want OPTIONS natpings uncomment next # setbflag(7); }; save("location"); exit; }; }; route(1); } route[1] { xlog("L_NOTICE", "\n**************IN ROUTE 1************\n"); if (subst_uri('/(sip:.*);nat=yes/\1/')){ setbflag(6); }; if (isflagset(5)||isbflagset(6)) { route(3); } if (!t_relay()) { sl_reply_error(); }; exit; } route[2]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; } route[3] { if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); t_on_failure("1"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); } failure_route[1] { if (isbflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } } onreply_route[1] { if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); if (isbflagset(6)) { fix_nated_contact(); } exit; } #======================================= Thanks in advance! Any other comment to make this cfg better, i'm all ears!! David

1 0

[OpenSER-Users] Make uninstall
by Daniel 07 Dec '07

07 Dec '07

Hi all, I have to uninstall openser from my pc. Is there a "make uninstall" ? Regerds, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

3 2

[OpenSER-Users] Problem with Mediaproxy
by Daniel 07 Dec '07

07 Dec '07

Dear all, I'm testing my proxy and I'm using mediaproxy for natted client. Everything works fine, if there is a client behind nat calling a public user, mediaproxy works fine. This is true also if there is a public client calling a natted user. I have some problem when there are 2 client behind NAT. 1. I tryed to test the system with 1 client behind NAT and 1 client behind the SAME NAT (both of client under the same subnet:192.168.10.x). In this case Mediaproxy doesn't work. RTP in direct between clients. 2. I tryed to test the system with 1 client behind NAT and 1 client behind the SAME NAT (with clients under different subnet:192.168.10.x and 192.168.0.x). In this case Mediaproxy doesn't work. RTP in direct between clients. This is Mediaproxy Debug in the last case: ********************************************** Dec 7 10:37:36 SERVER mediaproxy[2116]: request ZDhiMWM2NDhmNzQ1ODNjYThkZjIxZGNiMmEzZmQxMjI. 192.168.0.167:58786:audio NAT_IP PROXY_IP remote NAT_IP remote X-Lite=20release=201011s=20stamp=2041150 info=from:userA@PROXY_IP, to:UserB@PROXY_IP,fromtag:c1310701,totag: Dec 7 10:37:36 SERVER mediaproxy[2116]: session ZDhiMWM2NDhmNzQ1ODNjYThkZjIxZGNiMmEzZmQxMjI.: started. listening on MEDIAPROXY_IP:60014 Dec 7 10:37:36 SERVER mediaproxy[2116]: execution time: 1.56 ms Dec 7 10:37:43 SERVER mediaproxy[2116]: lookup ZDhiMWM2NDhmNzQ1ODNjYThkZjIxZGNiMmEzZmQxMjI. 192.168.10.224:25732:audio NAT_IP PROXY_IP remote unknown unknown X-Lite=20release=201011s=20stamp=2041150 info=from:UserA@PROXY_IP, to:UserB@PROXYIP,fromtag:c1310701,totag:67246350 Dec 7 10:37:43 SERVER mediaproxy[2116]: execution time: 0.51 ms Dec 7 10:38:00 SERVER mediaproxy[2116]: delete ZDhiMWM2NDhmNzQ1ODNjYThkZjIxZGNiMmEzZmQxMjI. info= Dec 7 10:38:00 SERVER mediaproxy[2116]: session ZDhiMWM2NDhmNzQ1ODNjYThkZjIxZGNiMmEzZmQxMjI.: 0/0/0 packets, 0/0/0 bytes (caller/called/relayed) Dec 7 10:38:00 SERVER mediaproxy[2116]: session ZDhiMWM2NDhmNzQ1ODNjYThkZjIxZGNiMmEzZmQxMjI.: ended. Dec 7 10:38:00 SERVER mediaproxy[2116]: execution time: 0.61 ms ******************************* I am sure of mediaproxy change SDP parameter even in these last cases (I can see on openser debug that proxy send out, for example, a 200 OK with SDP part changed. And xlogs DEBUG show me that openser enters in "use_media_proxy" routine) . But when I sniff the packet on client side, I see SDP part with private IP ! Any suggestion? Regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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[OpenSER-Users] T.38 Fax Problem with RTPProxy
by Raymond Chandler 07 Dec '07

07 Dec '07

In the following, I have a T.38 Fax call in which I'm forcing RTPProxy. The signalling follows this draft (http://tools.ietf.org/html/draft-mule-sip-t38callflows-02#section-5.2), but the UDPTL media seems to have problems. The problem is that OpenSER/RTPProxy seems to ignore the address in the SDP of the "200 OK". As I understand it, RTPProxy should, adequately enough, proxy the UDPTL. (not just rtp) The call flow is as follows: UC -> EDGE EDGE -> OpenSER OpenSER -> EU (not shown in this trace) NOTE: IPs have been changed to protect the innocent ;-) 192.168.174.156 represents our UC's (Underlying Carrier's) IP 192.168.48.88 represents our edge proxy 192.168.202.10 represents our OpenSER box Below, I have what I consider to be the "relevent" packets with large portions of repetition SNIP'd.... Below that, I have the two (again, what I consider to be) relevent packets with details of the SDP In both of the SDP details, our edge uses the UC's IP (192.168.174.156) in the c= line of the SDP. I don't see that being overwritten in any of the "media-specific" sections following. OpenSER/RTPProxy handle the RTP appropriately (to the best of my knowledge) but send the t.38 UDPTL media back to our edge which is, of course, rejecting it. "No.", "Time", "Source", "Destination", "Protocol", "Info" "1", "0.000000", "192.168.48.88", "192.168.202.10", "SIP/SDP", "Request: INVITE sip:1NPANXX0119@192.168.202.10:6080, with session description" "2", "0.005027", "192.168.202.10", "192.168.48.88", "SIP", "Status: 100 Trying" "3", "0.286018", "192.168.202.10", "192.168.48.88", "SIP", "Status: 180 Ringing" "4", "9.397591", "192.168.202.10", "192.168.48.88", "SIP/SDP", "Status: 200 OK, with session description" "5", "9.410633", "192.168.202.10", "192.168.48.88", "RTP", "Payload type=ITU-T G.711 PCMU, SSRC=3122798450, Seq=5169, Time=374804094" "6", "9.439195", "192.168.48.88", "192.168.202.10", "SIP", "Request: ACK sip:1NPANXX0119@192.168.202.10:6080" "7", "9.440987", "192.168.202.10", "192.168.48.88", "RTP", "Payload type=ITU-T G.711 PCMU, SSRC=3122798450, Seq=5170, Time=374804334" "8", "9.448560", "192.168.48.88", "192.168.202.10", "ICMP", "Destination unreachable (Port unreachable)" "9", "9.470710", "192.168.202.10", "192.168.48.88", "RTP", "Payload type=ITU-T G.711 PCMU, SSRC=3122798450, Seq=5171, Time=374804574" "10","9.479789","69.25.48.88","67.15.202.10","ICMP","Destination unreachable (Port unreachable)" "11","9.490158","207.218.174.156","67.15.202.10","RTP","PT=ITU-T G.711 PCMU, SSRC=0xCF806328, Seq=0, Time=0, Mark " "12","9.500186","67.15.202.10","207.218.174.156","RTP","PT=ITU-T G.711 PCMU, SSRC=0xBA221F72, Seq=5172, Time=374804814 " "13","9.509146","69.25.48.88","67.15.202.10","ICMP","Destination unreachable (Port unreachable)" "14","9.510270","207.218.174.156","67.15.202.10","RTP","PT=ITU-T G.711 PCMU, SSRC=0xCF806328, Seq=1, Time=160 " "15","9.530265","207.218.174.156","67.15.202.10","RTP","PT=ITU-T G.711 PCMU, SSRC=0xCF806328, Seq=2, Time=320 " "16","9.530283","67.15.202.10","207.218.174.156","RTP","PT=ITU-T G.711 PCMU, SSRC=0xBA221F72, Seq=5173, Time=374805054 " ~~~~ SNIP ~~~~ "174", "11.430039", "192.168.174.156", "192.168.202.10", "RTP", "Payload type=ITU-T G.711 PCMU, SSRC=3481297704, Seq=97, Time=15520" "175", "11.450027", "192.168.174.156", "192.168.202.10", "RTP", "Payload type=ITU-T G.711 PCMU, SSRC=3481297704, Seq=98, Time=15680" "176", "11.457118", "192.168.202.10", "192.168.48.88", "SIP/SDP", "Request: INVITE sip:Restricted@192.168.48.88, with session description" "177", "11.468025", "192.168.202.10", "192.168.174.156", "T.38", "UDP: UDPTLPacket Seq=32768 [UNKNOWN PER: unknown choice extension][Malformed Packet]" "178", "11.470013", "192.168.174.156", "192.168.202.10", "RTP", "Payload type=ITU-T G.711 PCMU, SSRC=3481297704, Seq=99, Time=15840" "179", "11.490125", "192.168.174.156", "192.168.202.10", "RTP", "Payload type=ITU-T G.711 PCMU, SSRC=3481297704, Seq=100, Time=16000" "180", "11.492010", "192.168.202.10", "192.168.174.156", "T.38", "UDP: UDPTLPacket Seq=32768 [UNKNOWN PER: unknown choice extension][Malformed Packet]" "181", "11.499118", "192.168.48.88", "192.168.202.10", "SIP", "Status: 100 Trying" "182", "11.510111", "192.168.174.156", "192.168.202.10", "RTP", "Payload type=ITU-T G.711 PCMU, SSRC=3481297704, Seq=101, Time=16160" "183", "11.515999", "192.168.202.10", "192.168.174.156", "T.38", "UDP: UDPTLPacket Seq=32768 [UNKNOWN PER: unknown choice extension][Malformed Packet]" "184", "11.530100", "192.168.174.156", "192.168.202.10", "RTP", "Payload type=ITU-T G.711 PCMU, SSRC=3481297704, Seq=102, Time=16320" "185", "11.540998", "192.168.202.10", "192.168.174.156", "T.38", "UDP: UDPTLPacket Seq=32768 [UNKNOWN PER: unknown choice extension][Malformed Packet]" "186", "11.550087", "192.168.174.156", "192.168.202.10", "RTP", "Payload type=ITU-T G.711 PCMU, SSRC=3481297704, Seq=103, Time=16480" "187", "11.570073", "192.168.174.156", "192.168.202.10", "RTP", "Payload type=ITU-T G.711 PCMU, SSRC=3481297704, Seq=104, Time=16640" "188", "11.580068", "192.168.174.156", "192.168.202.10", "RTP", "Unknown RTP version 0" "189", "11.581986", "192.168.202.10", "192.168.174.156", "T.38", "UDP: UDPTLPacket Seq=32768 [UNKNOWN PER: unknown choice extension][Malformed Packet]" "190", "11.590061", "192.168.174.156", "192.168.202.10", "RTP", "Unknown RTP version 0" "191", "11.599929", "192.168.174.156", "192.168.202.10", "RTP", "Unknown RTP version 0" "192", "11.605977", "192.168.202.10", "192.168.174.156", "T.38", "UDP: UDPTLPacket Seq=32768 [UNKNOWN PER: unknown choice extension][Malformed Packet]" "193", "11.608549", "192.168.48.88", "192.168.202.10", "SIP/SDP", "Status: 200 OK, with session description" "194", "11.630936", "192.168.202.10", "192.168.48.88", "T.38", "UDP: UDPTLPacket Seq=32768 [UNKNOWN PER: unknown choice extension][Malformed Packet]" "195", "11.660168", "192.168.202.10", "192.168.48.88", "T.38", "UDP: UDPTLPacket Seq=32768 [UNKNOWN PER: unknown choice extension][Malformed Packet]" "196", "11.668760", "192.168.48.88", "192.168.202.10", "ICMP", "Destination unreachable (Port unreachable)" "197", "11.690405", "192.168.202.10", "192.168.48.88", "T.38", "UDP: UDPTLPacket Seq=32768 [UNKNOWN PER: unknown choice extension][Malformed Packet]" "198", "11.697992", "192.168.48.88", "192.168.202.10", "ICMP", "Destination unreachable (Port unreachable)" "199", "11.720762", "192.168.202.10", "192.168.48.88", "T.38", "UDP: UDPTLPacket Seq=32768 [UNKNOWN PER: unknown choice extension][Malformed Packet]" "200", "11.728847", "192.168.48.88", "192.168.202.10", "ICMP", "Destination unreachable (Port unreachable)" "201", "11.750120", "192.168.202.10", "192.168.48.88", "T.38", "UDP: UDPTLPacket Seq=32768 [UNKNOWN PER: unknown choice extension][Malformed Packet]" "202", "11.758827", "192.168.48.88", "192.168.202.10", "ICMP", "Destination unreachable (Port unreachable)" "203", "11.780601", "192.168.202.10", "192.168.48.88", "T.38", "UDP: UDPTLPacket Seq=32768 t30ind: no-signal hdlc-fcs-BAD-sig-end t4-non-ecm-sig-end [Malformed?] hdlc-fcs-BAD-sig-end hdlc-data[Malformed Packet]" "204", "11.788059", "192.168.48.88", "192.168.202.10", "ICMP", "Destination unreachable (Port unreachable)" "205", "11.811073", "192.168.202.10", "192.168.48.88", "T.38", "UDP: UDPTLPacket Seq=32768 t30ind: no-signal hdlc-fcs-BAD-sig-end t4-non-ecm-sig-end [Malformed?] hdlc-fcs-BAD-sig-end hdlc-sig-end[Malformed Packet]" "206", "11.818665", "192.168.48.88", "192.168.202.10", "ICMP", "Destination unreachable (Port unreachable)" "207", "11.833039", "192.168.202.10", "192.168.48.88", "SIP", "Request: ACK sip:Restricted@192.168.48.88:5060" "208", "11.839935", "192.168.202.10", "192.168.48.88", "T.38", "UDP: UDPTLPacket Seq=32768 t30ind: cng hdlc-fcs-BAD-sig-end t4-non-ecm-sig-end [Malformed?] hdlc-fcs-BAD-sig-end hdlc-fcs-OK[Malformed Packet]" "209", "11.849145", "192.168.48.88", "192.168.202.10", "ICMP", "Destination unreachable (Port unreachable)" "210", "11.868409", "192.168.202.10", "192.168.48.88", "T.38", "UDP: UDPTLPacket Seq=00000 t30ind: no-signal" "211", "11.878000", "192.168.48.88", "192.168.202.10", "ICMP", "Destination unreachable (Port unreachable)" "212", "11.905983", "192.168.48.88", "192.168.202.10", "ICMP", "Destination unreachable (Port unreachable)" "213", "14.428661", "192.168.202.10", "192.168.48.88", "T.38", "UDP: UDPTLPacket Seq=00001 t30ind: v21-preamble" ~~~~ SNIP ~~~~ "389", "32.869453", "192.168.202.10", "192.168.48.88", "T.38", "UDP: UDPTLPacket Seq=00114 t30ind: no-signal" "390", "41.696978", "192.168.202.10", "192.168.48.88", "SIP", "Request: BYE sip:Restricted@192.168.48.88:5060" "391", "41.738094", "192.168.48.88", "192.168.202.10", "SIP", "Status: 200 OK" No. Time Source Destination Protocol Info 1 0.000000 192.168.48.88 192.168.202.10 SIP/SDP Request: INVITE sip:1NPANXX0119@192.168.202.10:6080, with session description Frame 1 (966 bytes on wire, 966 bytes captured) Ethernet II, Src: 00:1e:13:ca:4a:3f (00:1e:13:ca:4a:3f), Dst: Dell_ed:4a:1c (00:12:3f:ed:4a:1c) Internet Protocol, Src: 192.168.48.88 (192.168.48.88), Dst: 192.168.202.10 (192.168.202.10) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 6080 (6080) Session Initiation Protocol Request-Line: INVITE sip:1NPANXX0119@192.168.202.10:6080 SIP/2.0 Message Header Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): BRSDP 187392 187392 IN IP4 192.168.174.156 Session Name (s): BRSDP Session Connection Information (c): IN IP4 192.168.174.156 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 53278 RTP/AVP 0 8 18 100 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): rtpmap:100 telephone-event/8000 Media Attribute (a): fmtp:100 0-15 No. Time Source Destination Protocol Info 193 11.608549 192.168.48.88 192.168.202.10 SIP/SDP Status: 200 OK, with session description Frame 193 (1013 bytes on wire, 1013 bytes captured) Ethernet II, Src: 00:1e:13:ca:4a:3f (00:1e:13:ca:4a:3f), Dst: Dell_ed:4a:1c (00:12:3f:ed:4a:1c) Internet Protocol, Src: 192.168.48.88 (192.168.48.88), Dst: 192.168.202.10 (192.168.202.10) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 6080 (6080) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): BRSDP 187392 187393 IN IP4 192.168.174.156 Session Name (s): BRSDP Session Connection Information (c): IN IP4 192.168.174.156 Time Description, active time (t): 0 0 Media Description, name and address (m): image 53278 udptl t38 Media Attribute (a): T38FaxVersion:0 Media Attribute (a): T38MaxBitRate:14400 Media Attribute (a): T38FaxRateManagement:transferredTCF Media Attribute (a): T38FaxMaxBuffer:262 Media Attribute (a): T38FaxMaxDatagram:176 Media Attribute (a): T38FaxUdpEC:t38UDPRedundancy

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[OpenSER-Users] Does MediaProxy have to be loaded if there is no NAT?
by Brian Heath 07 Dec '07

07 Dec '07

So I'm trying to diagnose an intermittent bug we seem to be having, and I'm not sure if Mediaproxy's the cause or not... When a call is made through our system, and then the call is ended normally, the logs show the following mediaproxy entries: Dec 6 12:14:02 server01 mediaproxy[14777]: delete 7dfafbf931195f8b61d7bbea68f9c0f8(a)xxx.xxx.xxx.xx1 info= Dec 6 12:14:02 server01 mediaproxy[14777]: execution time: 0.20 ms It seems to be pretty standard stuff, really. However, if a call is not made from behind a firewall, why is mediaproxy even running? Alas, this is not my main problem... Every once in a while, calls get dropped. It happens more frequently than is acceptable. One common thing I've noticed when this happens, is mediaproxy will log a 'request' and start a session, and then approximately 30 seconds later, the call will be terminated. Take this one for example. A customer originated a call this morning at Dec 6 11:25:18. Half an hour later, mediaproxy terminated the call. This was the log entry: Dec 6 11:55:37 server01 mediaproxy[14777]: request 551afd3c247e7ddc5e8506d753494a3f(a)xxx.xxx.xxx.xx1 xxx.xxx.xxx.xx2:10508:audio xxx.xxx.xxx.xx2 sip.mydomain.com remote xxx.xxx.xxx.xx1 remote Quintum/1.0.0 info=from:1925XXXXXXX@sip.mydomain.com,to:1925XXXXXXX@xxx.xxx.xxx.xx1,fromtag:45364b32-2439a9,totag:as694a235d Dec 6 11:55:37 server01 mediaproxy[14777]: session 551afd3c247e7ddc5e8506d753494a3f(a)xxx.xxx.xxx.xx1: started. listening on xxx.xxx.xxx.xx3:60490 Dec 6 11:55:37 server01 mediaproxy[14777]: execution time: 1.12 ms Dec 6 11:56:08 server01 mediaproxy[14777]: delete 551afd3c247e7ddc5e8506d753494a3f(a)xxx.xxx.xxx.xx1 info= Dec 6 11:56:08 server01 mediaproxy[14777]: session 551afd3c247e7ddc5e8506d753494a3f(a)xxx.xxx.xxx.xx1: 0/0/0 packets, 0/0/0 bytes (caller/called/relayed) Dec 6 11:56:08 server01 mediaproxy[14777]: session 551afd3c247e7ddc5e8506d753494a3f(a)xxx.xxx.xxx.xx1: ended. Dec 6 11:56:08 server01 mediaproxy[14777]: execution time: 0.68 ms Is there any reason why mediaproxy might be arbitrarily loading and starting a new session, when the call has already long been established? I can't seem to make heads or tails of this. I'd appreciate any comments or suggestions. Thanks, Brian This question is also posted at: http://www.voipuser.org/forum_topic_11777.html

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Re: [OpenSER-Users] starting openser trunk version
by Daniel 06 Dec '07

06 Dec '07

On Thursday 06 December 2007, Daniel wrote: > I'm using the latest trunk version of openser. > When I start the proxy I see on debug this messages repeated forever... > I'm using a config file that worked well wiht openser release 1.2.2. > > Dec 6 15:55:14 [28924] DBG:core:qm_free: params(0x817ada0, 0x81caa24), > called from nathelper.c: nh_timer(2941) > Dec 6 15:55:14 [28924] DBG:core:qm_free: freeing frag. 0x81caa0c alloc'ed > from nathelper.c: nh_timer(2841) > Dec 6 15:55:15 [28924] DBG:core:qm_malloc: params (0x817ada0, 8), called > from nathelper.c: nh_timer(2841) > Dec 6 15:55:15 [28924] DBG:core:qm_malloc: params (0x817ada0, 8), returns Hi Daniel, this is the (harmless) memory debugging that was activated to catch some errors in memory management. This will be decativated next week, when 1.3 is released. So you can either deactivate it by yourself (look in Makefile.defs), ignore the messages or wait one week. ;-) Cheers, Henning Hi Henning, but now how can I see the classic debug, without these messages scrolling on the screen? Thanks again, Regards Daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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[OpenSER-Users] OpeSer SVN presence error
by TCB 06 Dec '07

06 Dec '07

I'm trying to setup openser-svn-3261 with presence support and gettign the following error. Dec 5 18:04:27 ss-za /usr/local/sbin/openser[7403]: Handle PUBLISH Dec 5 18:04:27 ss-za /usr/local/sbin/openser[7403]: ERROR:presence:handle_publish: Missing or unsupported event header field value Dec 5 18:04:27 ss-za /usr/local/sbin/openser[7403]: ERROR:presence:handle_publish: event=[presence I'm also having issues starting Oser with presence_xml module, its crashes with no error in my log. Disable it and oser start fine. -- TC

3 3

[OpenSER-Users] FreeRADIUS-CDRTool Prepaid Connector 1.1 Released
by Dan-Cristian Bogos 06 Dec '07

06 Dec '07

Folks, for any of you using (or having interest in using) prepaid authentication based on CDRTool through FreeRADIUS Authorization methods, I announce the release 1.1 available to be downloaded from SourceForge.net. Changelog from version 1.0 to present: !Version 1.1 - 05.12.2007 by Dan-Cristian Bogos [dan(a)itsyscom.com] * Replaced MAXDUR setting with SESSION_MAXDUR to be more descriptive * Introducing experimental support for multiple cdrtool engines by setting MULTIPLE_PREPAID_ENGINES=True * Simplified way of testing memcached connection (simple test against response list). * Fixed BUG with sending non supported methods to CDRTool (eg: REGISTER and MESSAGE). * Automatic mark of authenticated packets by returning third sip-avp "authentic#1". * Return USERLOCK instead of REJECT when the account is locked. * Added authenticate() function to comply with freeradius internals - not really necessary to be used. * Fixed BUG which was failing accounting if Acct-Type was not fix positioned inside parameter tuple coming from FreeRADIUS. * Introduced Acct-Authentic attribute dependency in order to fix double accounting in case of Yate(Cisco CDR with two leg accounting type). OpenSER should send Acct-Authentic attribute as radius_extra param with the value 1 for RADIUS or 3 for Remote. * Fixed BUG which was emptying destination-uri in case of empty Canonical-URI. * Forcing timeout for the keys in memcache based on SESSION_MAXDUR, for the case when no more accounting packets will arive after authorizaiton ones. * Introduced support for Acct-Session-Time attribute which will produce more aqurate session time then automatic generated based on timestamps ones. * Introduced support for combined accounting to fix the problems OpenSER has when no BYE packet arrives (eg: authorization generated by openser and accounting by Yate). * Introduced support for newer cdrtool response 'Not Prepaid' derived from old 'NotPrepaid' * Proper detection in case of Stop packets without Start being recorded inside memcached. * Updated README-txt file to reflect the changes necessary in the configuration of OpenSER plus simplified way of installing freeradius with "./configure --with-experimental-modules" I am at your disposal if any question might pop-up. Project link: https://sourceforge.net/projects/frad-cdrtool DanB

3 3

[Serusers] mailbox
by Szasz Szabolcs 06 Dec '07

06 Dec '07

Hello! I'm new in SEMS, I have a problem using mailbox application. I have set up mailbox to use the same imap mailbox to store the messages and mailbbox_query to get from there the messages, it works in this way. 1. We have two numbers: 200 and 201, respectivelly we have imap subfolders inbox/200/ and inbox/201/. 200 calls 201, 201 isn't online, sems records a message and I want to put in 201's imap subfulder this message. 2. when 201 is calling the mailbox_query I would like to be able to access direct the inbox/201 folder and get from there the messages. How can I configure to store in different subfolders for different destination numbers? How to get a number the mailbox messages from his subfolder when he calls in the mailbox_query? Thank you Szasz Szabolcs

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Re: [Serusers] myself value
by franz.edler＠inode.at 06 Dec '07

06 Dec '07

> So why do I receive icmp response on sip register request. If ser built > properly the myself value it should know that "new" > myselfxxx.xxx.xxx.xxxx:5066 and to take care or requests he receives on > port 5066 on sip level Two questions: 1. Are the ICMP responses generated by requests destined to port 5066? 2. Did you set the port directive accordingly in ser.cfg? Franz

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[Serusers] SIP Express Bundle ready-to-go idea, request for input
by Greger V. Teigre 06 Dec '07

06 Dec '07

Hi guys, I have been playing with the following idea: Create a ready-to-run OS image with everything that is needed for iptel.org apps pre-installed + a complete installation of: * SER 2.0 (release) * rtpproxy * SEMS * SERweb * maybe sipsak, some monitoring tools, etc The idea is to create a small script 'config_iptelorg' for configuring the installation to your needs. You should then be able to download the ready image, boot it, go through the script and have an up and running iptel.org proxy and app server just like the iptel.org free SIP service in maybe 10-15 minutes. This way you could host a SIP service for your own domain with close to no setup at all. Some questions to you: * Is there any interest for this at all? * I was thinking about using Ubuntu 7.10 server as the OS. Any thoughts/preferences? * Should the image be an Amazon EC3 image (you could use http://www.rightscale.com and get it running in no time with 10 run-hours free) or should it be a VMware appliance to be run with free VMware Player? * Other suggestions/comments? I could need some help with this, anyone interested in lending me a hand? (could be anything, documentation, testing, installation, etc) g-)

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[OpenSER-Users] force_tcp_alias
by Johnny Karlsen 06 Dec '07

06 Dec '07

There are numerous discussion on using force_tcp_alias() in order to route all SIP requests over an already established TCP between the client and OpenSER, however, I cannot make it work. What I would like to accomplish, any client registering using TCP should have all SIP requests sent towards the client using the already established TCP and not routed based on the Contact header used during REGISTER. Reading posts it would seem that putting force_tcp_alias(); within the route{ } should do the trick. However, this doesn't seem to do the trick for me, SIP requests towards the client are still routed using the Contact information and for example a SIP MESSAGE is sent over UDP from OpenSER towards the client using the information in the Conatct header. Any hints what to look for? Many thanks

2 2

[Serusers] myself value
by Evgeny Miloslavsky 06 Dec '07

06 Dec '07

Hi I'm kind of newbie at all the SER issue so sorry for bothering you with my questions. I defined at ser.cfg to listen on port 5066 instead of default 5060 port, saved changes and restarted the SER. At Wireshark sniffer I see that there is an ICMP--> Destination Port Unreachable response on all sip REGISTER requests arrived from various endpoints at my setup. As far as I understand this means that configured 5066 port is not bound with SER application. So my question is: where can I see, how is "myself" built or concatenation of which parameters creates "myself" value? Thanks

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[OpenSER-Users] update trunk version
by Daniel 06 Dec '07

06 Dec '07

Hi, I'm trying to compile the latest trunk version of cpl-c module. I used the following commands: make make modules=modules/cpl-c modules but I had these errors: Compiling cpl.c gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=athlon64 -DNAME='"openser"' -DVERSION='"1.2.2-notls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.1.1"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -I/usr/include/libxml2 -I/usr/local/include/libxml2 -I/usr/local/include -c cpl.c -o cpl.o cpl.c:52:24: error: ../../pvar.h: No such file or directory cpl.c:118: error: expected '=', ',', ';', 'asm' or '__attribute__' before 'cpl_procs' cpl.c:129: warning: excess elements in struct initializer cpl.c:129: warning: (near initialization for 'cmds[0]') cpl.c:131: warning: excess elements in struct initializer cpl.c:131: warning: (near initialization for 'cmds[1]') cpl.c:133: warning: excess elements in struct initializer cpl.c:133: warning: (near initialization for 'cmds[2]') cpl.c:134: warning: excess elements in struct initializer cpl.c:134: warning: (near initialization for 'cmds[3]') cpl.c:183: error: 'cpl_procs' undeclared here (not in a function) cpl.c:184: warning: initialization from incompatible pointer type cpl.c:185: warning: initialization from incompatible pointer type cpl.c:186: warning: initialization from incompatible pointer type cpl.c:188: warning: excess elements in struct initializer cpl.c:188: warning: (near initialization for 'exports') cpl.c: In function 'fixup_cpl_run_script': cpl.c:202: warning: implicit declaration of function 'LM_ERR' cpl.c: In function 'cpl_init': cpl.c:233: error: 'pv_spec_t' undeclared (first use in this function) cpl.c:233: error: (Each undeclared identifier is reported only once cpl.c:233: error: for each function it appears in.) cpl.c:233: error: expected ';' before 'avp_spec' cpl.c:237: warning: implicit declaration of function 'LM_INFO' cpl.c:241: warning: implicit declaration of function 'LM_CRIT' cpl.c:260: warning: implicit declaration of function 'pv_parse_spec' cpl.c:260: error: 'avp_spec' undeclared (first use in this function) cpl.c:261: error: 'PVT_AVP' undeclared (first use in this function) cpl.c:266: warning: implicit declaration of function 'pv_get_avp_name' cpl.c:353: warning: implicit declaration of function 'LM_NOTICE' make[1]: *** [cpl.o] Error 1 make[1]: Leaving directory `/usr/src/openser-1.2.2-notls/modules/cpl-c' make: *** [modules] Error 2 what's the problem? regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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[OpenSER-Users] CPL Errors
by Daniel 05 Dec '07

05 Dec '07

Hi all, What does this error means ? It occurs only when I call a CPL script af a natted User. Refer to OUTGOING CPL Script: 4(9914) ERROR:cpl_c: overflow detected ip=0xb608fa30 offset=506 in func. cpl_run.c, line 1000 4(9914) ERROR: t_reply: cannot send a t_reply to a message for which no T-state has been established Refer to INCOMING CPL Script: 4(9914) ERROR:cpl_c:cpl_proxy_to_loc_set: do_action failed 4(9914) ERROR:cpl_c:cpl_run_script: runtime error Regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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[OpenSER-Users] Openser is giving compile errors for avp_write and avp_pushto
by Sandeep Ammaladinna 05 Dec '07

05 Dec '07

Hi Henning, I commented out the "fifo" part and still the config file errs. When i tried with openser -c option it says, config file OK. But when I actually run the openser it cribs. Attached is the log file. Cheers, -Sandeep -----Original Message----- From: users-bounces(a)lists.openser.org [mailto:users-bounces@lists.openser.org] On Behalf Of users-request(a)lists.openser.org Sent: Wednesday, December 05, 2007 5:30 PM To: users(a)lists.openser.org Subject: Users Digest, Vol 31, Issue 11 Send Users mailing list submissions to users(a)lists.openser.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.openser.org/cgi-bin/mailman/listinfo/users or, via email, send a message with subject or body 'help' to users-request(a)lists.openser.org You can reach the person managing the list at users-owner(a)lists.openser.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Users digest..." Today's Topics: 1. configure openser with cpl module (Martin Kardi) 2. Re: Proxy with OpenSER (Andreas M.) 3. Openser is giving compile errors for avp_write and avp_pushto (Sandeep Ammaladinna) 4. Re: Openser is giving compile errors for avp_write and avp_pushto (Henning Westerholt) 5. Re: uac_replace_from issues (Henning Westerholt) 6. Could someone help me with WeSIP ? (Daniel Grotti) ---------------------------------------------------------------------- Message: 1 Date: Tue, 4 Dec 2007 04:15:18 -0800 (PST) From: Martin Kardi <martin.kardos(a)pego.sk> Subject: [OpenSER-Users] configure openser with cpl module To: users(a)lists.openser.org Message-ID: <14149601.post(a)talk.nabble.com> Content-Type: text/plain; charset=us-ascii Hi, I installed openser with cpl module and other dependencies but I don't know how to configure it? Openser is functionally without cpl module. I am beginner in this. Could somebody help me how to? thanks Martin -- View this message in context: http://www.nabble.com/configure-openser-with-cpl-module-tf4942775.html#a 14149601 Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com. ------------------------------ Message: 2 Date: Wed, 05 Dec 2007 08:57:33 +0100 From: "Andreas M." <openser(a)breitenlee.net> Subject: Re: [OpenSER-Users] Proxy with OpenSER To: openser list user <users(a)lists.openser.org> Message-ID: <475659ED.9060809(a)breitenlee.net> Content-Type: text/plain; charset=ISO-8859-1 Hello, an IMS P-CSCF is not just a Proxy, there are a lot of routines and functions inside, that could not be done with an IETF SIP Proxy. Take a look at www.openimscore.org, there are also config examples, based on SER. r, Andreas M. hammou moujane schrieb: > Hello everybody, > > I am new in this mailing list and new user of OpenSER. I would like to > know where can I find the doc which can help me to make OpenSER as a > proxy server like P-CSCF of IMS core. what to be modified in > openser.conf configuration file. > > > > Bests Regards. > -- > Hammou MOUJANE > Tel: +212 62 46 33 89 > INPT > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users -- g, Andreas M. ------------------------------ Message: 3 Date: Wed, 5 Dec 2007 14:13:11 +0530 From: "Sandeep Ammaladinna" <sammaladinna(a)velankani.com> Subject: [OpenSER-Users] Openser is giving compile errors for avp_write and avp_pushto To: <users(a)lists.openser.org>, <devel(a)lists.openser.org> Message-ID: <001b01c8371a$e14dd170$1010000a(a)blr.velankani.com> Content-Type: text/plain; charset="us-ascii" Hi, When I am compiling my openser.cfg, Openser is throwing errors for avp_write and avp_pushto. The syntax I am using is as follows: avp_write("$ruri", "i:10"); avp_pushto("$ruri", "i:10"); What could be the issue; I am following the syntax from the cook book. Attached is my .cfg Thanks in advance. Cheers -Sandeep

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[OpenSER-Users] Openser is giving compile errors for avp_write and avp_pushto
by Sandeep Ammaladinna 05 Dec '07

05 Dec '07

Hi, When I am compiling my openser.cfg, Openser is throwing errors for avp_write and avp_pushto. The syntax I am using is as follows: avp_write("$ruri", "i:10"); avp_pushto("$ruri", "i:10"); What could be the issue; I am following the syntax from the cook book. Attached is my .cfg Thanks in advance. Cheers -Sandeep

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[OpenSER-Users] uac_replace_from issues
by Alex Hermann 05 Dec '07

05 Dec '07

Hello, I have 2 issues with uac_replace_from. 1) Can someone add a big fat warning to the documentation that "append_fromtag" from the rr module MUST be enabled for automatic restore to function? I got random memory locations copied into the To: or From: header when it was disabled. 2) I get the message "ERROR:uac:restore_from: new URI shorter than old URI". Is there a limitation to what I can replace or what it can be replaced with? Can I safely remove this limitation in modules/uac/from.c ? diff -ru openser-1.2-orig/modules/uac/from.c openser-1.2/modules/uac/from.c --- openser-1.2-orig/modules/uac/from.c 2007-10-22 11:48:48.000000000 +0200 +++ openser-1.2/modules/uac/from.c 2007-12-04 17:20:44.000000000 +0100 @@ -472,10 +472,6 @@ } /* get new uri */ - if ( new_uri.len<old_uri.len ) { - LOG(L_ERR,"ERROR:uac:restore_from: new URI shorter than old URI\n"); - goto failed; - } for( i=0 ; i<old_uri.len ; i++ ) new_uri.s[i] ^= old_uri.s[i]; if (new_uri.len==old_uri.len) { if ( new_uri.len<old_uri.len ) { LOG(L_ERR,"ERROR:uac:restore_from: new URI shorter than old URI\n"); goto failed; } -- Greetings, Alex Hermann

2 1

[OpenSER-Users] Proxy with OpenSER
by hammou moujane 05 Dec '07

05 Dec '07

Hello everybody, I am new in this mailing list and new user of OpenSER. I would like to know where can I find the doc which can help me to make OpenSER as a proxy server like P-CSCF of IMS core. what to be modified in openser.conf configuration file. Bests Regards. -- Hammou MOUJANE Tel: +212 62 46 33 89 INPT

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[OpenSER-Users] Parallel forking to loopback problem - for ringing multiple numbers at same time, simulring
by Dave Singer 05 Dec '07

05 Dec '07

by the way, I'm trying this using openser 1.3.0-pre1-notls from latest svn of trunk as of 12-4-07 I'm trying to do findme logic.. Call comes in, lookup settings for where it is supposed to go. >From the settings I find that the user wants it to ring a friends home phone, a cell phone, and the home ata associated with account. To keep voicemail handled by the home account and not by one of the places that they are forwarding to they want voicemail to be triggered after 15 seconds. The friends home phone is actually one of our accounts and has two ata's associated with it that register as the same sip user and they have it also ring a cell phone and the account sends the call to voicemail after 18 seconds. Psudo Call flow/Logic of what I have so far. Call comes in to openser for 555/123-4567 pull from usr_preferences settings for the call destination number. so I get a stack of avps $avp(s:brahch_uri) from usr_preferences like this: sip:5551239876@127.0.0.1 sip:localuser3@127.0.0.1 sip:5553216548@127.0.0.1 I do diversion header then branch for them with: avp_pushto("$br", "$avp(s:branch_uri)" I drop the local branch. The branched calls come in as expected and verify that it is a valid intentional loopback by testing: if ( ruri =~ "127.0.0.1" && dst_ip == 127.0.0.1 ) { < mark as valid loopback > } Like the beginning of the original call I lookup where they should go and if the user in the diversion header is allowed to send them there (PSTN) In this case one call would be sent out through the PSTN gateways with full failover logic, another would call the device associated with the account by doing a lookup("location") and the third, (the one to the friend that our customer) would add another diversion header and start another parallel fork with loopback for the cell phone and two ata's When I test with a simple forward to a single PSTN number, in the loopbacked session, I rewrite the ruri accordingly with real dest user and real ip address. the ruri shows what I want in the logs just before t_relay and exit and in branch_route[1] after last log entry from branch_route, I have in the log: ERROR:core:udp_send: sendto(sock,0xb619d954,1376,0,0xb619ba6c,16): Invalid argument(22) CRITICAL:core:udp_send: invalid sendtoparameters one possible reason is the server is bound to localhost and attempts to send to the net ERROR:tm:msg_send: udp_send failed ERROR:tm:t_forward_nonack: sending request failed Then I have log entrys from onreply_route[1]: Reply - S=100 M=INVITE RURI=<null> src_ip = 127.0.0.1 Reply - S=477 M=INVITE RURI=<null> src_ip = 127.0.0.1 not the real world IP address of the other server I set in the ruri and there is nothing in the logs on the intended server or the local server indicating routing logic for an invite failed I'm woundering if the TM module, while processing the invite in the loopback, is matching up the transaction with the session that originated the loopback and using the settings and status from it. Any Ideas, Thoughts, HELP!!? Or is OpenSER just not ready for this yet? Thanks Dave

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[OpenSER-Users] uac_restore_from problem
by Zoltán Kovács 05 Dec '07

05 Dec '07

Dear Colleagues, first of all, thank you for the great software which is a very nice and convenient way to build up a stable VoIP telephone system! We already have a working system based on OpenSER and Asterisk. After many experiments I finally raise this question to you because I cannot figure out the solution. The problem described on http://www.openser.org/pipermail/users/2006-January/002747.html seems to be very similar to mine. I am using openser-1.1.0-9etch1. I tried to replace the FROM variable as follows: loadmodule "uac.so" modparam("uac","from_restore_mode","auto") ... route { ... avp_write("$fU", "$avp(s:display)"); avp_write("$fU", "$avp(s:dispuri)"); avp_subst("$avp(s:display)", "/^36(.*)/06\1/"); avp_subst("$avp(s:dispuri)", "/^36(.*)/sip:06\1@datanet.hu/"); uac_replace_from("$avp(s:display)", "$avp(s:dispuri)"); ... } Unfortunately nothing seems to happen: no change is made on several VoIP telephones, including X-Lite as well. loose_route() and record_route() were also used in my configuration. Could you please help me what I am missing here? If needed, I can also send a full config file. Thank you in advance, Zoltan Kovacs sysadmin

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[Serusers] tm module problem
by Szasz Szabolcs 05 Dec '07

05 Dec '07

Hi I have a problem with tm module, I can't set the module parameter pass_provisional_replies . The module tm is loaded succesfully, I insert in to the configuration this line: modparam("tm", "pass_provisional_replies", 1) When I start ser I got this error message : 0(9618) set_mod_param_regex: parameter <pass_provisional_replies> not found in module <tm> 0(9618) parse error (33,19-20): Can't set module parameter ser version used 0.9.6 , I couldn't found any solution on the google. I have to recompile the tm module with some modifications ? or I have other mistake in the configuration? Please help! Thank you Szasz Szabolcs

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[OpenSER-Users] Problem with setting the Acct-Authentic Radius Attribute
by Dan-Cristian Bogos 04 Dec '07

04 Dec '07

Folks, I am trying to make use of Acct-Authentic to identify authenticated sessions inside accounting packets. All ok, no problem with dictionaries, but when it comes to receiving the value in freeradius server, I get all sort of strange ones (integers). I am already using Acct-Authentic properly working with other software on the same radius server. Here is my config: modparam("acc", "radius_extra", " Called-Station-Id=$tu; Calling-Station-Id=$fu; Canonical-URI=$tu; User-Name=$au; Sip-User-Realm=$ar; Source-IP=$si; Source-Port=$sp; Acct-Authentic=$avp(s:acct_authentic); From-Header=$hdr(from); User-Agent=$hdr(user-agent); Contact=$hdr(contact); Event=$hdr(event)") I have tried different values for the $avp(s:acct_authentic): 1, "1", RADIUS, "RADIUS", etc. and each time I got a corresponding encoded. EG: If I will set the AVP through SIP-AVP or directly inside the routing script, for acct_authentic=1 I will receive in radius Acct-Authentic = 49. Are there any developers around who have implemented radius support in ser/openser and could help me troubleshooting this issue? My openser version: openser -V version: openser 1.2.2-tls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: 2:3003M @(#) $Id: main.c 2304 2007-05-25 16:36:07Z bogdan_iancu $ Ta, DanB

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Re: [OpenSER-Users] OpenSER Sending INVITE twice
by Ovidiu Sas 04 Dec '07

04 Dec '07

There is no per upstream server config. Best thing to do is upgrade to openser 1.2. Starting with this release the timer granularity has been increased. If you have a high RTT, you can increase the T1 timer: http://www.openser.org/docs/modules/1.2.x/tm#AEN168 Hope this helps, Ovidiu Sas On Dec 4, 2007 12:44 PM, Douglas Garstang <dougmig33(a)yahoo.com> wrote: > > Oh, and RTT as OpenSER determines it to where? Is it one RTT for all > outgoing traffic or is it a different RTT for every upstream server? > > > > ----- Original Message ---- > From: Ovidiu Sas <osas(a)voipembedded.com> > To: Douglas Garstang <dougmig33(a)yahoo.com> > Cc: Users(a)lists.openser.org > Sent: Monday, November 26, 2007 5:47:23 PM > Subject: Re: [OpenSER-Users] OpenSER Sending INVITE twice > > This is the way that is supposed to work. If nothing is received from > upstream, the request is resent. > From the rfc3261: > > 17.1.1.1 Overview of INVITE Transaction > > The INVITE transaction consists of a three-way handshake. The client > transaction sends an INVITE, the server transaction sends responses, > and the client transaction sends an ACK. For unreliable transports > (such as UDP), the client transaction retransmits requests at an > interval that starts at T1 seconds and doubles after every > retransmission. T1 is an estimate of the round-trip time (RTT), and > it defaults to 500 ms. > > > Regards, > Ovidiu Sas > > On Nov 26, 2007 3:29 PM, Douglas Garstang <dougmig33(a)yahoo.com> wrote: > > > > I have an OpenSER 1.1 install here, and for some reason, OpenSER is > sending > > the INVITE message twice to the upstream host. > > > > The time difference between them is about 1/5 of a second. Nothing is > > received between the first and second INVITE's. > > > > Why is OpenSER doing this? > > > > I have xlog() statements everywhere in openser.cfg, and OpenSER is only > > logging ONE outgoing INVITE message, eventhough it's sending two. It is > > logging the multiple TRYING messages that come back however. > > > > OpenSER is not calling failure_route because nothing is logged in there. > > What could be going wrong? Why is it doing this? > > > > Doug. > > > > > > > > ________________________________ > > Get easy, one-click access to your favorites. Make Yahoo! your homepage. > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > > ________________________________ > Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it > now.

1 0

[OpenSER-Users] invalid cseq for aor
by Gerson Matiolli 04 Dec '07

04 Dec '07

I used the SER for 2 years and not had this problem. I was using OPENSER for 20 days without problems. How can I fix this in Openser ? Thanks [OpenSER-Users] invalid cseq for aor Bogdan-Andrei Iancu bogdan at voice-system.ro Tue Dec 4 09:09:53 UTC 2007 * Previous message: [OpenSER-Users] invalid cseq for aor * Next message: [OpenSER-Users] perl * Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] Hi Gerson, The error is generated when a re-REGISTER (same contact and callid) is received but with invalid CSEQ number (equal or less than the previous). Regards, Bogdan Gerson Matiolli wrote: > Hi, All > > I am with the OPENSER running some time without problems. > > Yesterday, he went not to complete any call, and the system had the following message log: > > > ERROR:registry:update_contacts: invalid cseq for aor > ERROR:registry:update_contacts: invalid cseq for aor > ERROR:registry:update_contacts: invalid cseq for aor > > for all numbers > > Thanks >

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[OpenSER-Users] Remove header from Cancel
by Jarkko Nevala 04 Dec '07

04 Dec '07

Hello. Is it possible to remove Route: -header from Cancel requests. I have an user agent that inserts Route: -header in Cancel requests. I've already tried to get rid of it with remove_hf("Route") function, but it did not have any effect. From what I understand is, that this header is quite unnecessary in a stateful proxy, because the Cancel messages are handled hop-by-hop basis. It's just that it causes problems at the next hop proxy. I didn't even know about this problem with the UA before I started using the development version of OpenSER, the 1.2.2 version removed the Route: -header automatically before relaying Cancel forward. Thank you JN

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[OpenSER-Users] how does fix_nated_register() work?
by Daniel Grotti 04 Dec '07

04 Dec '07

Hi all, I have a user behind NAT with address : sip:userA@192.168.10.210:42276. When user send REGISTER, the proxy checks if user is behind NAT. If it is, proxy makes: fix_nated_register(); force_rport(); setbflag(6). When I check what kind of address proxy has saved on LOCATION database, I see at Contact column: sip:userA@IP_NAT:42276. So proxy mantains the same Port of the user and doesn't save the NAT's IP:PORT pair. Why? In this way, in fact, proxy is not able to keep connection alive. Regards, daniel

2 1

[OpenSER-Users] Module Programming
by Daniel Corbe 04 Dec '07

04 Dec '07

Hello, Is there a better module programming guide than the one found on the docuwiki at http://www.openser.org/dokuwiki/doku.php/development:write-module ? It really doesn't explain much; for example the fact that functions are (or used to be) called with a paramater struct sip_msg *. The original SER 0.8.x and 0.9.x core had an extremely comprehensive guide to writing modules; however, I'm reluctant to use those texts as a guide for fear that the material is dated. Regards, Daniel

2 1

[OpenSER-Users] Regarding CRBT
by Amit Vijayvargiya 04 Dec '07

04 Dec '07

Hi all, can anybody tell where we have to put the routing code in openser.cfg for CRBT(caller ring backtone ) application. A------------------------------------------make call -------------->B same time A------------------------------------------------------------------------->C in "C" we are playing prompts. Regards Amit

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[OpenSER-Users] Could Any body suggest me an SIP programming interface please?
by Samira Salam 04 Dec '07

04 Dec '07

Hi , As a part of my MS project I proposed a 3-party authentication protocol which can be used in SIP too. for SIP. The proxy server utilizes this protocol to authenticate the UC's Invite,using the Registrar server. Now, I have to estimate the totoal overhead of the scheme on a normal SIP call. I searched dozens of SIP, UMTS, GSM related papers to find a simulator or emulator which could be used to to achive the goal but I couldn't get any solutions as I need to consider execution of the RSA,AES and SHA on the agent's processor, I also need the solution to answer me with not too much programming due to my time constraints. I browsed the Resirocate project docs but I don't think then can answer me in time. I am also not sure whether SIPp, MS LIve communication server and SERcan help me? Could you please suggest me any rapid solutions? Any simulation techniques?any piece of software or programmining interface? By the way I am a little familiar with opnet as I did a tiny project with it. Doest it help me? Thanks in advance. Samira. --------------------------------- Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.

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[OpenSER-Users] openser: general question
by Helmut Kuper 04 Dec '07

04 Dec '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I'm quite unsure about openser's hangling of message processing. So can anybody answere following 2 questions please ? 1. When multiple messages come in, lets say 3 INVITEs. Will openser fork one child for each message, run the route script and terminate the child after that? Or will openser throw all 3 INVITEs at the same time into the route script? 2. If the 3 INVITEs share the same route script, are they handled sequentially? What happens to the AVPs and $var variables? Are they gonna mixed? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHVRWA4tZeNddg3dwRAkQZAJ9Ruy3kr2RzODDlm55uZ4Y6yZTcPACgrl+v qmOW524gE6vgsWNgOalpfW4= =oR4D -----END PGP SIGNATURE-----

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[OpenSER-Users] Napting interval
by Daniel 04 Dec '07

04 Dec '07

Hi all, what is the different between napting interval done by nathelper-module and the other ones done by mediaproxy-module ? Can I set both of them at the same time ? regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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[OpenSER-Users] Calls Between Registered Accounts
by VoIP Forums www.Go4Calls.com 04 Dec '07

04 Dec '07

Good day friends, I was trying to install Openser and Asterisk on same server. I need some configuration or information if anyone can point me. I need to forward only International call to the Asterisk but the same time if registered accounts of openser want to make call between them the can call each other without routing to asterisk. Could you please give us the right parameters of openser.cfg to route calls between openser accounts? Regards, www.Go4Calls.Com VoIP Forums _________________________________________________________________ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE

2 1

[OpenSER-Users] invalid cseq for aor
by Gerson Matiolli 04 Dec '07

04 Dec '07

Hi, All I am with the OPENSER running some time without problems. Yesterday, he went not to complete any call, and the system had the following message log: ERROR:registry:update_contacts: invalid cseq for aor ERROR:registry:update_contacts: invalid cseq for aor ERROR:registry:update_contacts: invalid cseq for aor for all numbers Thanks

2 1

[OpenSER-Users] Help with Setup
by Arun Kumar 04 Dec '07

04 Dec '07

Hi, I need to configure my ser / openser in such a way that these sites should be able to call each other : Site A -> 01113, 01114 Site B -> 02113, 02114 Site C -> 03113, 03114 User in Site A (Dial 02113 or 03113) --------> SER / OpenSER --------> User in Site B or Site C and they should be able to call within site : User in Site A (Call 114) ----------> SER / OpenSER ---------> User in Site A here 01, 02, 03 are my site code so site code is only needed when users want to call between sites. If they are calling within same site no need to dial site id. please advise and give some links. Thanks, arun

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[Serusers] SER data model discussion - developers please read
by X Z 04 Dec '07

04 Dec '07

Hi All, This is specifically for the SER/OpenSER developers, but I'm not a serdev list member so I'm posting here. I've been using SER since version 0.8.X and I'm still running 0.8.14production for my company PBX to this day. I was very excited as version 2 became a release candidate and I downloaded it for testing. I was pretty disappointed with one aspect of the new data model and I'm requesting that the developers consider a further revision on the data model. Basically, taking all fields out of the subscriber table like Last_name, first_name, email, timezone, rpid/asserted identity, etc, etc is not the greatest idea. It's a better idea from a database architecture and performance perspective to keep adding columns into that table for data that has a 1 to 1 relationship with a user, and that is common in > 90% of SER's use cases (i.e. corporate, carrier/VSP.) I would suggest adding voicemail_password, and maybe every other field that is being added into the default attributes script that I saw in CVS recently. If you already know what attributes a user has (and they have a 1 to 1 relationship), then its far better from a db performance perspective to keep these attributes in the user table. I know that the code becomes more complicated, but I think it may be a tradeoff worth discussing. See this discussion ( http://asktom.oracle.com/pls/asktom/f?p=100:11:0::::P11_QUESTION_ID:1067808…) between Oracle users and Tom, (an Oracle engineer/architect.) The full text of this discussion is very informative and I highly recommend people read it through. Tom's conclusion is that the type of data model being discussed, and now being used in SER fails for all but the most trivial of applications. Maybe SER *by itself* qualifies as "trivial" from a database architect's perspective, but think about things like Asterisk integration, which is quite common. You quickly run into some very nasty queries . . . Please note that I am not a software developer nor a database engineer, just a user who reads a lot, so I'm open to being the ignorant one here, but I thought that this should be discussed among users and developers. Thanks for considering, Mahatma http://asktom.oracle.com/pls/asktom/f?p=100:11:0::::P11_QUESTION_ID:1067808… The following is an excerpt from the above link: Here is a excerpt from my forthcoming book where I talk about this (and show you how ugly, hard and inefficient queries against your very flexible model will be) (2)Do not use Generic Data Models Frequently I see applications built on a generic data model for "maximum flexibility" or applications built in ways that prohibit performance. Many times - these are one in the same thing! For example, it is well known you can represent any object in a database using just four tables: Create table objects ( oid int primary key, name varchar2(255) ); Create table attributes ( attrId int primary key, attrName varchar2(255), datatype varchar2(25) ); Create table object_Attributes ( oid int, attrId int, value varchar2(4000), primary key(oid,attrId) ); Create table Links ( oid1 int, oid2 int, primary key (oid1, oid2) ); That's it - no more CREATE TABLE for me! I can fill the attributes table up with rows like this: insert into attributes values ( 1, 'DATE_OF_BIRTH', 'DATE' ); insert into attributes values ( 2, 'FIRST_NAME', 'STRING' ); insert into attributes values ( 3, 'LAST_NAME', 'STRING' ); commit; And now I'm ready to create a PERSON record: insert into objects values ( 1, 'PERSON' ); insert into object_Attributes values( 1, 1, '15-mar-1965' ); insert into object_Attributes values( 1, 2, 'Thomas' ); insert into object_Attributes values( 1, 3, 'Kyte' ); commit; insert into objects values ( 2, 'PERSON' ); insert into object_Attributes values( 2, 1, '21-oct-1968' ); insert into object_Attributes values( 2, 2, 'John' ); insert into object_Attributes values( 2, 3, 'Smith' ); commit; And since I'm good at SQL, I can even query this record up to get the FIRST_NAME and LAST_NAME of all PERSON records: ops$tkyte@ORA920> select max( decode(attrName, 'FIRST_NAME', value, null )) first_name, 2 max( decode( attrName, 'LAST_NAME', value, null ) ) last_name 3 from objects, object_attributes, attributes 4 where attributes.attrName in ( 'FIRST_NAME', 'LAST_NAME' ) 5 and object_attributes.attrId = attributes.attrId 6 and object_attributes.oid = objects.oid 7 and objects.name = 'PERSON' 8 group by objects.oid 9 / FIRST_NAME LAST_NAME -------------------- -------------------- Thomas Kyte John Smith Looks great, right? I mean, the developers don't have to create tables anymore, we can add columns at the drop of a hat (just requires an insert into the ATTRIBUTES table). The developers can do whatever they want and the DBA can't stop them. This is ultimate "flexibility". I've seen people try to build entire systems on this model. But, how does it perform? Miserably, terribly, horribly. A simple "select first_name, last_name from person" query is transformed into a 3-table join with aggregates and all. Further, if the attributes are "NULLABLE" - that is, there might not be a row in OBJECT_ATTRIBUTES for some attributes, you may have to outer join instead of just joining which in some cases can remove more optimal query plans from consideration. Writing queries might look pretty straightforward, but it's impossible to do in a performant fashion. For example, if we wanted to get everyone that was born in MARCH or has a LAST_NAME = 'SMITH', we could simply take the query from above and just wrap an inline view around that: ops$tkyte@ORA920> select * 2 from ( 3 select max(decode(attrName, 'FIRST_NAME', value, null)) first_name, 4 max(decode(attrName, 'LAST_NAME', value, null)) last_name, 5 max(decode(attrName, 'DATE_OF_BIRTH', value, null)) date_of_birth 6 from objects, object_attributes, attributes 7 where attributes.attrName in ( 'FIRST_NAME', 'LAST_NAME', 'DATE_OF_BIRTH' ) 8 and object_attributes.attrId = attributes.attrId 9 and object_attributes.oid = objects.oid 10 and objects.name = 'PERSON' 11 group by objects.oid 12 ) 13 where last_name = 'Smith' 14 or date_of_birth like '%-mar-%' 15 / FIRST_NAME LAST_NAME DATE_OF_BIRTH -------------------- -------------------- -------------------- Thomas Kyte 15-mar-1965 John Smith 21-oct-1968 So, it looks "easy" to query, but think about the performance! If you had a couple thousand OBJECT records, and a couple tens of thousands of OBJECT_ATTRIBUTES - Oracle would have to process the entire inner group by query first and then apply the WHERE clause. This is not a made up data model, one that I crafted just to make a point. This is an actual data model that I've seen people try to use. Their goal is ultimate flexibility. They don't know what OBJECTS they need, they don't know what ATTRIBUTES they will have. Well - that is what the database was written for in the first place: Oracle implemented this thing called SQL to define OBJECTS and ATTRIBUTES and lets you use SQL to query them. You are trying to put a generic layer on top of a generic layer - and it fails each and every time except for the most trivial of applications.

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Re: [OpenSER-Users] OpenSER Sending INVITE Twice!?
by Ovidiu Sas 03 Dec '07

03 Dec '07

Hi Douglas, You will need to upgrade to 1.2.2. This issue was fixed starting with 1.2.0. Regards, Ovidiu Sas On Dec 3, 2007 3:12 PM, Douglas Garstang <dougmig33(a)yahoo.com> wrote: > > Ok, I can accept that it might be a timeout since it doesn't happen every > single time. > 1/5s seems awefully short though. How can I increase this? I can't find > anything in the docs about increasing this timeout... > > Doug. > > > > ----- Original Message ---- > From: Ovidiu Sas <osas(a)voipembedded.com> > To: Tavis P. <tavis.lists(a)galaxytelecom.net> > Cc: users(a)openser.org > Sent: Monday, December 3, 2007 12:01:53 PM > Subject: Re: [OpenSER-Users] OpenSER Sending INVITE Twice!? > > Right, 1/5 is not 1/2 :) > > -ovi > > On Dec 3, 2007 10:10 AM, Tavis P. <tavis.lists(a)galaxytelecom.net> wrote: > > > > It may also be related to the timer accuracy of OpenSER 1.1 > > > > If the T1 interval is 500ms but OpenSER 1.1 doesn't support sub-second > > timer accuracy it is probable to see it retransmitting in situations where > > T1 hasn't been reached yet > > It shouldn't cause any problems though, just a few erroneous retransmits > > from time to time > > > > tavis > > > > Ovidiu Sas wrote: > > Hi Douglas, > > > > You posted this questionawhileago and by checkingthe archives > > Irealised that my answer was truncated somehow ... > > Here's the full answer: > > > > This is the way that is supposed to work. If nothing is received from > > upstream, the request is resent. > > >From the rfc3261: > > > > 17.1.1.1 Overview of INVITE Transaction > > > > The INVITE transaction consists of a three-way handshake. The client > > transaction sends an INVITE, the server transaction sends responses, > > and the client transaction sends an ACK. For unreliable transports > > (such as UDP), the client transaction retransmits requests at an > > interval that starts at T1 seconds and doubles after every > > retransmission. T1 is an estimate of the round-trip time (RTT), and > > it defaults to 500 ms. > > > > > > It seems that your upstream UAS is slowin sending out the 100 Trying > > and therefor openSER is resending out the INVITE. > > > > > > Regards, > > Ovidiu Sas > > > > On Dec 3, 2007 12:35 PM, Douglas Garstang <dougmig33(a)yahoo.com> wrote: > > > > > > I have an OpenSER 1.1 install here, and for some reason, OpenSER is > sending > > the INVITE message twice to the upstream host. > > > > Asterisk OpenSER Provider > > ----------------------------------------------- > > | ---INVITE---> | | > > | | ---INVITE---> | > > | | ---INVITE---> | > > | | <---TRYING--- | > > | | <---TRYING--- | > > > > The time difference between them is about 1/5 of a second. Nothing is > > received between the first and second INVITE's. > > > > Why is OpenSER doing this? > > > > I have xlog() statements everywhere in openser.cfg, and OpenSER is only > > logging ONE outgoing INVITE message, eventhough it's sending two. It is > > logging the multiple TRYING messages that come back however. > > > > OpenSER is not calling failure_route because nothing is logged in there. > > What could be going wrong? Why is it doing this? > > > > Doug. > > > > ________________________________ > > Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it > > now. > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > ________________________________ > Get easy, one-click access to your favorites. Make Yahoo! your homepage.

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[OpenSER-Users] OpenSER Sending INVITE Twice!?
by Douglas Garstang 03 Dec '07

03 Dec '07

I have an OpenSER 1.1 install here, and for some reason, OpenSER is sending the INVITE message twice to the upstream host. Asterisk OpenSER Provider ----------------------------------------------- | ---INVITE---> | | | | ---INVITE---> | | | ---INVITE---> | | | <---TRYING--- | | | <---TRYING--- | The time difference between them is about 1/5 of a second. Nothing is received between the first and second INVITE's. Why is OpenSER doing this? I have xlog() statements everywhere in openser.cfg, and OpenSER is only logging ONE outgoing INVITE message, eventhough it's sending two. It is logging the multiple TRYING messages that come back however. OpenSER is not calling failure_route because nothing is logged in there. What could be going wrong? Why is it doing this? Doug. ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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[OpenSER-Users] Multidomain Support
by Arun Kumar 03 Dec '07

03 Dec '07

Hi, I'm going to setup multi domain ser or openser , if some one can have any sample configuration of some good guide to configure. Please help. thanks Arun

2 1

[OpenSER-Users] why mediaproxy doesn't works?
by Daniel Grotti 03 Dec '07

03 Dec '07

Hi all, I'm usign 2 users behind the same Nat and I'm using x-lite 3.0. User A calls user B. I want to use mediaproxy between 2 users and I have configured my openser file in that way. Mediaproxy doesn't works, and I've noticed diferences between the "200 OK" message received by user A and that sent by proxy. user A --- INVITE ----> proxy --- INVITE-----> user B user A <----200 OK ---- proxy < ----- 200 OK --- user B this is the 200 OK message sent by proxy to USER A. This has been already manipulated by proxy (chenge IP and PORT in SDP part) for using mediaproxy : SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.245:14620;received=85.18.14.25;branch=z9hG4bK-d87543-37310b5f2633e814-1--d87543-;rport=18267 Record-Route: <sip:137.204.107.138:5060;lr;nat=yes;ftag=910e1011> Contact: <sip:daniel@85.18.14.25:20881;rinstance=8a9df9cdfc3da728> To: "daniel"<sip:daniel@137.204.107.138>;tag=6e33c440 From: "daniel.grotti"<sip:daniel.grotti@137.204.107.138>;tag=910e1011 Call-ID: Y2M3ZDNjOGExN2Q0YTBlYjViMTg1NDAwOTAxZjliMDY. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1011s stamp 41150 Content-Length: *375* v=0 o=- 2 2 IN IP4 192.168.10.224 s=CounterPath X-Lite 3.0 *c=IN IP4 137.204.107.138 IP of media relay* t=0 0 *m=audio 60004 RTP/AVP 107 119 100 106 0 105 98 8 101 # 60004 port of mediarelay* a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv and this is the message sniffed by client A: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.245:14620;received=85.18.14.25;branch=z9hG4bK-d87543-37310b5f2633e814-1--d87543-;rport=18267 Record-Route: <sip:137.204.107.138:5060;lr;nat=yes;ftag=910e1011> Contact: <sip:daniel@85.18.14.25:20881;rinstance=8a9df9cdfc3da728> To: "daniel"<sip:daniel@137.204.107.138>;tag=6e33c440 From: "daniel.grotti"<sip:daniel.grotti@137.204.107.138>;tag=910e1011 Call-ID: Y2M3ZDNjOGExN2Q0YTBlYjViMTg1NDAwOTAxZjliMDY. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1011s stamp 41150 Content-Length: *374* v=0 o=- 2 2 IN IP4 192.168.10.224 s=CounterPath X-Lite 3.0 *c=IN IP4 192.168.10.224* t=0 0 *m=audio 36124 RTP/AVP 107 119 100 106 0 105 98 8 101* a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv It's the same message but with c and m parameter changed. So, mediaproxy doesn't work and 2 users are directly connected. any suggestion? regards, daniel

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Re: [OpenSER-Users] [Serusers] Multidomain Support
by Jiri Kuthan 03 Dec '07

03 Dec '07

Default SER configuration does that, you just need command-line tools to add the domain. -jiri At 16:41 03/12/2007, Arun Kumar wrote: >Hi, > >I'm going to setup multi domain ser or openser , if some one can have any sample configuration of some good guide to configure. Please help. > > >thanks > >Arun >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

1 0

Re: [OpenSER-Users] perl
by David Villasmil Govea 03 Dec '07

03 Dec '07

> > > Message: 9 > Date: Mon, 3 Dec 2007 15:59:06 +0100 > From: Bastian Friedrich <bastian.friedrich(a)collax.com> > Subject: Re: [OpenSER-Users] perl > To: users(a)lists.openser.org > Message-ID: <200712031559.06986.bastian.friedrich(a)collax.com> > Content-Type: text/plain; charset="iso-8859-15" > > On Monday 03 December 2007, David Villasmil Govea wrote: > [...] > > 0(0) perl error: Can't locate OpenSER/Utils/PhoneNumber.pm in @INC > [...] > > > > Although modules are there in: > > > > No, there is no "PhoneNumber.pm" in that path. > > [...] > > me@server:/usr/local/lib/openser/perl# ls -l > > /usr/local/lib/openser/perl/OpenSER/Utils/ > > total 8 > > -rw-r--r-- 1 root root 6096 2007-12-03 14:00 PhoneNumbers.pm > > It's "PhoneNumbers.pm". Correct typing is _not_ optional while programming > :) thanks for the heads-up.. but i didn't type a thing, its all in the tar file... so there's a typo there. so THIS should be the correct /usr/local/etc/openser/openser.pl file, which comes with the tar ball: ************************************************************************************************************** push @INC, '/usr/local/lib/openser/perl/'; use OpenSER::Utils::PhoneNumbers; OpenSER->import; my $phonenumber = new OpenSER::Utils::PhoneNumbers( publicAccessPrefix => "0", internationalPrefix => "+", longDistancePrefix => "0", areaCode => "761", pbxCode => "456842", countryCode => "49" ); $canonical = canonicalForm("07612034567"); $number = dialNumber("+497612034567"); exit 0; ************************************************************************************************************** thanks > > > > > > - Is there a tutorial for using MySQL as a backend? How about a > > > > cluster? > > > > > > MySQL backend for what? > > > > I have it working as a MySQL backend for users. What i need is the > logging > > of cdrs... i understand that there are several records for each call. > But, > > how do i configure this? is it possible with MySQL? thanks a lot. > > You should start a different thread with a different subject. > > What you are looking for seems to be accounting, provided by the OpenSER > module "acc". Module documentation is available on > http://www.openser.org/docs/modules/1.3.x/acc.html > The "acc" module works fine with MySQL. > > Bastian > > -- > Collax GmbH . Burkheimer Stra?e 3 . 79111 Freiburg . Germany > p: +49 (0) 761-45684-24 > f: +49 (0) 761-45684-10 www.collax.com > > Gesch?ftsf?hrer: William K. Hite / Boris Nalbach > AG M?nchen HRB 158898 . Ust.-IdNr: DE 814464942 > \ There are 2 ways to handle women and I know neither. > > > > ------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > End of Users Digest, Vol 31, Issue 5 > ************************************ > -- DVG -- Imagination is more important than knowledge Albert Einstein

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[OpenSER-Users] openser trunk: presence bug?
by Helmut Kuper 03 Dec '07

03 Dec '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I use openser 1.3 latest trunk and try to use presence module. If presence server receives PUBLISH message the following error occurs on console: Dec 3 13:24:03 [616] ERROR:presence:handle_publish: Missing or unsupported event header field value Dec 3 13:24:03 [616] ERROR:presence:handle_publish: event=[presence] My openser config is this: xlog("Handle $rm\n"); # presence handling if(is_method("PUBLISH")) { handle_publish(); xlog("DONE!\n"); sl_send_reply("200","PUBLISH ok "); t_release(); } Is this a bug or do I have to define the events somewhere? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHU/cP4tZeNddg3dwRAujxAKCEG2zdB7nhqychuMcgf6oEurky4gCdHH/v NGJILGEwcG7Xc3mwfQytKFY= =Wwcp -----END PGP SIGNATURE-----

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[OpenSER-Users] perl
by David Villasmil Govea 03 Dec '07

03 Dec '07

> > > OpenSER modules can be included while compiling by using the > "include_modules" > statement. Thus, the perl module (not "mod_perl", as Daniel already > pointed > out) can be compiled along with OpenSER by using > make include_modules="perl" > > You will need your distribution's perl development packages installed > while > compiling. Yes, I have it compiled with no problems, but when i try to load a perl script, i get this: 0(0) perl: Setting lib path: '/usr/local/lib/openser/perl/' 0(0) perl error: Can't locate OpenSER/Utils/PhoneNumber.pm in @INC (@INC contains: /usr/local/lib/openser/perl/ /etc/perl /usr/local/lib/perl/5.8.8 /usr/local/share/perl/5.8.8 /usr/lib/perl5 /usr/share/perl5 /usr/lib/perl/5.8 /usr/share/perl/5.8 /usr/local/lib/site_perl .) at /usr/local/etc/openser/openser.pl line 4. 0(0) perl error: Can't locate OpenSER/Utils/PhoneNumber.pm in @INC (@INC contains: /usr/local/lib/openser/perl/ /etc/perl /usr/local/lib/perl/5.8.8 /usr/local/share/perl/5.8.8 /usr/lib/perl5 /usr/share/perl5 /usr/lib/perl/5.8 /usr/share/perl/5.8 /usr/local/lib/site_perl .) at /usr/local/etc/openser/openser.pl line 4. BEGIN failed--compilation aborted at /usr/local/etc/openser/openser.pl line 4. Can't locate OpenSER/Utils/PhoneNumber.pm in @INC (@INC contains: /usr/local/lib/openser/perl/ /etc/perl /usr/local/lib/perl/5.8.8 /usr/local/share/perl/5.8.8 /usr/lib/perl5 /usr/share/perl5 /usr/lib/perl/5.8 /usr/share/perl/5.8 /usr/local/lib/site_perl .) at /usr/local/etc/openser/openser.pl line 4. BEGIN failed--compilation aborted at /usr/local/etc/openser/openser.pl line 4. Although modules are there in: me@server:/usr/local/lib/openser/perl# ls -l /usr/local/lib/openser/perl/ total 4 drwxrwx--- 4 root root 168 2007-12-03 14:00 OpenSER -rwxrwx--- 1 root root 1354 2007-12-03 14:00 OpenSER.pm me@server:/usr/local/lib/openser/perl# ls -l /usr/local/lib/openser/perl/OpenSER total 12 -rw-r--r-- 1 root root 4844 2007-12-03 14:00 Constants.pm drwxr-xr-x 2 root root 120 2007-12-03 14:00 LDAPUtils -rw-r--r-- 1 root root 1602 2007-12-03 14:00 Message.pm drwxr-xr-x 2 root root 80 2007-12-03 14:00 Utils me@server:/usr/local/lib/openser/perl# ls -l /usr/local/lib/openser/perl/OpenSER/Utils/ total 8 -rw-r--r-- 1 root root 6096 2007-12-03 14:00 PhoneNumbers.pm > > > Feel free to ask if you have any concrete questions. > > > - Is there a tutorial for using MySQL as a backend? How about a > > cluster? > > MySQL backend for what? I have it working as a MySQL backend for users. What i need is the logging of cdrs... i understand that there are several records for each call. But, how do i configure this? is it possible with MySQL? thanks a lot. > > > Bastian > thanks much for your help! David -- DVG -- Imagination is more important than knowledge Albert Einstein

2 1

[Serusers] Problems with syslog and mysql on version 0.96
by Michael Roeper 03 Dec '07

03 Dec '07

Hi, we have SER version 0.96 running on a debian system using mysql for user authentication. Everything works fine as long as we use the standard logging to /var/log/syslog. When we change the logging to use another syslog facility (e.g. log_facility=LOG_LOCAL0 in ser.cfg) all messages are written to the file specified in syslog.conf, but the users are not able to register to ser. It looks as if the db-connection to the mysql database is not available. Has anybody observed this behaviour and can provide a hint ? We appreciate any kind of help. Kind regards, Michael -- Dr.Michael Roeper Email: m.roeper(a)wenzel-elektronik.de Wenzel Elektronik GmbH Tel.: 04121/4872-69 Max-Planck-Str. 2-4 Fax : 04121/4872-18 25335 Elmshorn http://www.wenzel-elektronik.de Sitz der Gesellschaft: Elmshorn Registergericht: Pinneberg, HRB 1974 EL USt-IdNr.: DE 812946778 Geschäftsführung: Dipl. Ing Jörg Simon, Dr. Volker Henkel --

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[OpenSER-Users] OpenSER: t-check_trans question
by Helmut Kuper 03 Dec '07

03 Dec '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, today I found, that t_check_trans function of TM module seems to has a problem with at least PUBLISH and SUBSCRIBE Messages. If openser checks theses messages with t_check_trans() route script execution will stop. Is this intended or a bug? Module docc says, that t_check_trans() cares about ACKs and CANCELs .. regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHU9mU4tZeNddg3dwRAlk/AJ9k/B/YBum72U45csWBqyi7RMLxXQCfWLLx rxRkJmYXFSOFvq6FlYKR3ng= =0bIH -----END PGP SIGNATURE-----

2 1

[Serusers] function get_from_did(&did, msg) error
by Tomasz Zieleniewski 03 Dec '07

03 Dec '07

Hi, I was trying to set up media proxy module and after some time of debuging I discovered that the problems were in the get_from_did(&did, msg) function. in mediaproxy.c file in line 469 there is such piece of code: if (get_from_did(&did, msg) == 0) { *type = "local"; return did; } when this code is invoked the result is wrong. this is some nowhere pointer with the length of the field unpredictable. after this invocation there is another piece of code wich tries to allocate some memory baseing on the length of the did string.(line 1449) cmdlen = callId.len + strlen(clientIP) + fromDomain.len + toDomain.len + userAgent.len*3 + infolen + 128; command = pkg_malloc(cmdlen); if (!command) { LOG(L_ERR, "error: use_media_proxy() - (!command): out of memory\n"); return -1; } this causes out of memory error when value of fromDomain.len gives random big number(below example of one my shots) Dec 2 18:17:17 rd ser[5884]: route[MEDIA]: use_media_proxy() Dec 2 18:17:17 rd ser[5884]: error: get_from_did - true Dec 2 18:17:17 rd ser[5884]: error: callId.len: 44 Dec 2 18:17:17 rd ser[5884]: error: strlen(clientIP): 14 Dec 2 18:17:17 rd ser[5884]: error: fromDomain.len: 1235653920 Dec 2 18:17:17 rd ser[5884]: error: toDomain.len: 14 Dec 2 18:17:17 rd ser[5884]: error: userAgent.len*3: 96 Dec 2 18:17:17 rd ser[5884]: error: infolen: 107 Dec 2 18:17:17 rd ser[5884]: error: cmdlen 1235654349 Dec 2 18:17:17 rd ser[5884]: error: use_media_proxy() - (!command): out of memory please, can someone check this and fix best regards tomasz

2 2

[OpenSER-Users] mod_perl
by David Villasmil Govea 03 Dec '07

03 Dec '07

Hello all, Its my first time writing to the list, i hope someone can help me out. - Can anyone point me in the right direction for compiling with mod_perl? I tried compiling but it wouldn't work. That was a few months ago, though. Is there a good step-by-step tutorial? - Is there a tutorial for using MySQL as a backend? How about a cluster? thanks to all. David

3 2

[Serusers] error in mediaproxy - module generates wrong request format
by Tomasz Zieleniewski 02 Dec '07

02 Dec '07

Hi, I am using mediaproxy for rtp traffic. I am using ser rel_2_0_0_rc5 and media proxy 1.9.0. When there is an invite message and i invoke use_media_proxy media proxy module in ser send wrong request to proxydispatcher or mediaproxy (depending on the type of configuration) The lookup request format according to the modules/request.py file is 'cmd', 'sessionId', 'streams', 'signalingIP', 'callerDomain', 'callerType','calledDomain', 'calledType', 'userAgent', 'info' but mediaproxy module sends such one: lookup MjI3MjcwMzA3NGQxNTc3YjUxZDA5NmQ1NzE4ZmNiOTQ. 192.168.129.74:13122:audio 192.168.129.74 local voip.touk.pl remote Asterisk=20PBX info= from:tzl@voip.touk.pl,to:poczta@voip.touk.pl ,fromtag:cb239428,totag:as707ae074 there is no callerDomain attribute. Please tell me did I miss something or this is realy bug in mediaproxy module. Cheers -tomasz

1 0

Re: [OpenSER-Users] Users Digest, Vol 31, Issue 2
by David Villasmil Govea 02 Dec '07

02 Dec '07

Oops! sorry, wrong list!

1 0

[OpenSER-Users] compiling woomera for FS
by David Villasmil Govea 02 Dec '07

02 Dec '07

Hello all again, I jus compiled beta2 of FS perfectly. I want to use woomera H.323 but it is simply impossible to do. I tried to follow the instructions on the wiki but it simply doesn't work. pwlib and openh323 compiled fine but woomera comes back with: g++ -D_REENTRANT -Wall -I/root/openh323/../pwlib/include -DPTRACING -I/root/openh323/include -Os -felide-constructors -Wreorder -c main.cxx -o obj_linux_x86_r/main.o In file included from main.cxx:129: openh323.h:212:22: warning: no newline at end of file /root/openh323/../pwlib/include/ptclib/pwavfile.h:166: warning: â?~packedâ?T attribute ignored for field of type â?~char [4]â?T /root/openh323/../pwlib/include/ptclib/pwavfile.h:172: warning: â?~packedâ?T attribute ignored for field of type â?~PWAV::ChunkHeaderâ?T /root/openh323/../pwlib/include/ptclib/pwavfile.h:173: warning: â?~packedâ?T attribute ignored for field of type â?~char [4]â?T /root/openh323/../pwlib/include/ptclib/pwavfile.h:178: warning: â?~packedâ?T attribute ignored for field of type â?~PWAV::ChunkHeaderâ?T lib/woomera.h:819: warning: â?~class Woomera::ExternalCommandâ?T has virtual functions but non-virtual destructor /root/openh323/include/h4601.h: In member function â?~H460_FeatureContent::operator H460_FeatureTable*()â?T: /root/openh323/include/h4601.h:292: warning: type-punning to incomplete type might break strict-aliasing rules after repeating the lines above several times: g++ -D_REENTRANT -Wall -I/root/openh323/../pwlib/include -DPTRACING -I/root/openh323/include -Os -felide-constructors -Wr eorder -c wivr.cxx -o obj_linux_x86_r/wivr.o /root/openh323/../pwlib/include/ptclib/pwavfile.h:166: warning: â?~packedâ?T attribute ignored for field of type â?~char [4]â ?T /root/openh323/../pwlib/include/ptclib/pwavfile.h:172: warning: â?~packedâ?T attribute ignored for field of type â?~PWAV::Chu nkHeaderâ?T /root/openh323/../pwlib/include/ptclib/pwavfile.h:173: warning: â?~packedâ?T attribute ignored for field of type â?~char [4]â ?T /root/openh323/../pwlib/include/ptclib/pwavfile.h:178: warning: â?~packedâ?T attribute ignored for field of type â?~PWAV::Chu nkHeaderâ?T ./lib/woomera.h:819: warning: â?~class Woomera::ExternalCommandâ?T has virtual functions but non-virtual destructor g++ -o obj_linux_x86_r/woomera -L/root/openh323/../pwlib/lib -L/root/openh323/lib ./obj_linux_x86_r/main.o ./obj_linux_ x86_r/openh323.o ./obj_linux_x86_r/woomera.o ./obj_linux_x86_r/server.o ./obj_linux_x86_r/connection.o ./obj_linux_x86_r/ client.o ./obj_linux_x86_r/wivr.o -lh323_linux_x86_r -lpt_linux_x86_r -lpthread -lssl -lcrypto -lresolv -ldl ./obj_linux_x86_r/openh323.o: In function `OpenH323WoomeraCall::OnH323OpenAudioChannel(H323AudioCodec&, int)': openh323.cxx:(.text+0x2b49): undefined reference to `PVXMLSession::PVXMLSession(PTextToSpeech*, int)' openh323.cxx:(.text+0x2ba2): undefined reference to `PVXMLSession::SetTextToSpeech(PString const&)' ./obj_linux_x86_r/woomera.o: In function `WoomeraPlayableToneDevice::Open(PVXMLChannel&, PString const&, int, int, int)': woomera.cxx:(.text+0x1239): undefined reference to `PVXMLPlayableData::SetData(PBYTEArray const&)' woomera.cxx:(.text+0x1253): undefined reference to `PVXMLPlayableData::Open(PVXMLChannel&, PString const&, int, int, int)' ./obj_linux_x86_r/woomera.o: In function `PVXMLPlayable::~PVXMLPlayable()': woomera.cxx:(.text._ZN13PVXMLPlayableD2Ev[PVXMLPlayable::~PVXMLPlayable()]+0x10): undefined reference to `vtable for PVXMLPla yable' ./obj_linux_x86_r/woomera.o: In function `PVXMLPlayableData::~PVXMLPlayableData()': woomera.cxx:(.text._ZN17PVXMLPlayableDataD2Ev[PVXMLPlayableData::~PVXMLPlayableData()]+0x10): undefined reference to `vtable for PVXMLPlayableData' ./obj_linux_x86_r/woomera.o: In function `PVXMLPlayableFilename::~PVXMLPlayableFilename()': woomera.cxx:(.text._ZN21PVXMLPlayableFilenameD2Ev[PVXMLPlayableFilename::~PVXMLPlayableFilename()]+0x10): undefined reference to `vtable for PVXMLPlayableFilename' ./obj_linux_x86_r/woomera.o: In function `PVXMLPlayableURL::~PVXMLPlayableURL()': woomera.cxx:(.text._ZN16PVXMLPlayableURLD2Ev[PVXMLPlayableURL::~PVXMLPlayableURL()]+0x10): undefined reference to `vtable for PVXMLPlayableURL' ./obj_linux_x86_r/woomera.o: In function `PVXMLPlayable::PVXMLPlayable()': woomera.cxx:(.text._ZN13PVXMLPlayableC2Ev[PVXMLPlayable::PVXMLPlayable()]+0xe): undefined reference to `vtable for PVXMLPlaya ble' ./obj_linux_x86_r/woomera.o: In function `PVXMLPlayableData::PVXMLPlayableData()': woomera.cxx:(.text._ZN17PVXMLPlayableDataC2Ev[PVXMLPlayableData::PVXMLPlayableData()]+0x13): undefined reference to `vtable f or PVXMLPlayableData' ./obj_linux_x86_r/woomera.o: In function `PVXMLPlayableFilename::PVXMLPlayableFilename()': woomera.cxx:(.text._ZN21PVXMLPlayableFilenameC2Ev[PVXMLPlayableFilename::PVXMLPlayableFilename()]+0x16): undefined reference to `vtable for PVXMLPlayableFilename' ./obj_linux_x86_r/woomera.o: In function `PVXMLPlayableURL::PVXMLPlayableURL()': woomera.cxx:(.text._ZN16PVXMLPlayableURLC2Ev[PVXMLPlayableURL::PVXMLPlayableURL()]+0x16): undefined reference to `vtable for PVXMLPlayableURL' ./obj_linux_x86_r/woomera.o: In function `WoomeraPlayableFilename::OnStop()': woomera.cxx:(.text._ZN23WoomeraPlayableFilename6OnStopEv[WoomeraPlayableFilename::OnStop()]+0xf): undefined reference to `PVX MLPlayableFilename::OnStop()' ./obj_linux_x86_r/woomera.o:(.rodata._ZTV25WoomeraPlayableToneDevice[vtable for WoomeraPlayableToneDevice]+0x38): undefined r eference to `PVXMLPlayableData::Play(PVXMLChannel&)' ./obj_linux_x86_r/woomera.o:(.rodata._ZTV25WoomeraPlayableToneDevice[vtable for WoomeraPlayableToneDevice]+0x54): undefined r eference to `PVXMLPlayable::ReadFrame(PVXMLChannel&, void*, int)' ./obj_linux_x86_r/woomera.o:(.rodata._ZTV25WoomeraPlayableToneDevice[vtable for WoomeraPlayableToneDevice]+0x58): undefined r eference to `PVXMLPlayableData::Rewind(PChannel*)' ./obj_linux_x86_r/woomera.o:(.rodata._ZTV26WoomeraPlayableSoundDevice[vtable for WoomeraPlayableSoundDevice]+0x54): undefined reference to `PVXMLPlayable::ReadFrame(PVXMLChannel&, void*, int)' ./obj_linux_x86_r/woomera.o:(.rodata._ZTI25WoomeraPlayableToneDevice[typeinfo for WoomeraPlayableToneDevice]+0x10): undefined reference to `typeinfo for PVXMLPlayableData' ./obj_linux_x86_r/woomera.o:(.rodata._ZTI26WoomeraPlayableSoundDevice[typeinfo for WoomeraPlayableSoundDevice]+0x10): undefin ed reference to `typeinfo for PVXMLPlayable' ./obj_linux_x86_r/woomera.o:(.rodata._ZTV23WoomeraPlayableFilename[vtable for WoomeraPlayableFilename]+0x34): undefined refer ence to `PVXMLPlayableFilename::Open(PVXMLChannel&, PString const&, int, int, int)' ./obj_linux_x86_r/woomera.o:(.rodata._ZTV23WoomeraPlayableFilename[vtable for WoomeraPlayableFilename]+0x38): undefined refer ence to `PVXMLPlayableFilename::Play(PVXMLChannel&)' ./obj_linux_x86_r/woomera.o:(.rodata._ZTV23WoomeraPlayableFilename[vtable for WoomeraPlayableFilename]+0x54): undefined refer ence to `PVXMLPlayable::ReadFrame(PVXMLChannel&, void*, int)' ./obj_linux_x86_r/woomera.o:(.rodata._ZTV23WoomeraPlayableFilename[vtable for WoomeraPlayableFilename]+0x58): undefined refer ence to `PVXMLPlayableFilename::Rewind(PChannel*)' ./obj_linux_x86_r/woomera.o:(.rodata._ZTI23WoomeraPlayableFilename[typeinfo for WoomeraPlayableFilename]+0x10): undefined ref erence to `typeinfo for PVXMLPlayableFilename' ./obj_linux_x86_r/woomera.o:(.rodata._ZTV18WoomeraPlayableURL[vtable for WoomeraPlayableURL]+0x34): undefined reference to `P VXMLPlayableURL::Open(PVXMLChannel&, PString const&, int, int, int)' ./obj_linux_x86_r/woomera.o:(.rodata._ZTV18WoomeraPlayableURL[vtable for WoomeraPlayableURL]+0x38): undefined reference to `P VXMLPlayableURL::Play(PVXMLChannel&)' ./obj_linux_x86_r/woomera.o:(.rodata._ZTV18WoomeraPlayableURL[vtable for WoomeraPlayableURL]+0x54): undefined reference to `P VXMLPlayable::ReadFrame(PVXMLChannel&, void*, int)' ./obj_linux_x86_r/woomera.o:(.rodata._ZTI18WoomeraPlayableURL[typeinfo for WoomeraPlayableURL]+0x10): undefined reference to `typeinfo for PVXMLPlayableURL' ./obj_linux_x86_r/server.o: In function `Woomera::Call::SetVar(PString const&, PString const&)': server.cxx:(.text+0x59c): undefined reference to `PVXMLSession::SetTextToSpeech(PString const&)' ./obj_linux_x86_r/server.o: In function `Woomera::Call::OnCmdRecord(PStringArray const&, PMIMEInfo const&, PString const&)': server.cxx:(.text+0x70bc): undefined reference to `PVXMLChannel::EndRecording()' ./obj_linux_x86_r/server.o: In function `Woomera::Call::OnCmdSpeak(PStringArray const&, PMIMEInfo const&, PString const&)': server.cxx:(.text+0x7e79): undefined reference to `PVXMLSession::ConvertTextToFilenameList(PString const&, PTextToSpeech::Tex tType, PStringArray&, int)' server.cxx:(.text+0x8053): undefined reference to `PVXMLPlayableFilenameList::Open(PVXMLChannel&, PStringArray const&, int, i nt, int)' ./obj_linux_x86_r/server.o: In function `PVXMLPlayableFilenameList::~PVXMLPlayableFilenameList()': server.cxx:(.text._ZN25PVXMLPlayableFilenameListD2Ev[PVXMLPlayableFilenameList::~PVXMLPlayableFilenameList()]+0x10): undefine d reference to `vtable for PVXMLPlayableFilenameList' ./obj_linux_x86_r/server.o: In function `PVXMLRecordableFilename::~PVXMLRecordableFilename()': server.cxx:(.text._ZN23PVXMLRecordableFilenameD2Ev[PVXMLRecordableFilename::~PVXMLRecordableFilename()]+0x10): undefined refe rence to `vtable for PVXMLRecordableFilename' ./obj_linux_x86_r/server.o: In function `PVXMLPlayableFilenameList::PVXMLPlayableFilenameList()': server.cxx:(.text._ZN25PVXMLPlayableFilenameListC2Ev[PVXMLPlayableFilenameList::PVXMLPlayableFilenameList()]+0x13): undefined reference to `vtable for PVXMLPlayableFilenameList' ./obj_linux_x86_r/server.o: In function `WoomeraPlayableFilenameList::OnStop()': server.cxx:(.text._ZN27WoomeraPlayableFilenameList6OnStopEv[WoomeraPlayableFilenameList::OnStop()]+0xf): undefined reference to `PVXMLPlayableFilenameList::OnStop()' ./obj_linux_x86_r/server.o: In function `PVXMLRecordableFilename::PVXMLRecordableFilename()': server.cxx:(.text._ZN23PVXMLRecordableFilenameC2Ev[PVXMLRecordableFilename::PVXMLRecordableFilename()]+0x16): undefined refer ence to `vtable for PVXMLRecordableFilename' ./obj_linux_x86_r/server.o:(.rodata._ZTV27WoomeraPlayableFilenameList[vtable for WoomeraPlayableFilenameList]+0x3c): undefine d reference to `PVXMLPlayableFilenameList::OnRepeat(PVXMLChannel&)' ./obj_linux_x86_r/server.o:(.rodata._ZTV27WoomeraPlayableFilenameList[vtable for WoomeraPlayableFilenameList]+0x54): undefine d reference to `PVXMLPlayable::ReadFrame(PVXMLChannel&, void*, int)' ./obj_linux_x86_r/server.o:(.rodata._ZTI27WoomeraPlayableFilenameList[typeinfo for WoomeraPlayableFilenameList]+0x10): undefi ned reference to `typeinfo for PVXMLPlayableFilenameList' ./obj_linux_x86_r/server.o:(.rodata._ZTV25WoomeraRecordableFilename[vtable for WoomeraRecordableFilename]+0x30): undefined re ference to `PVXMLRecordableFilename::Open(PString const&)' . . . . . wivr.cxx:(.text+0x5464): undefined reference to `PXMLElement::GetElement(int) const' wivr.cxx:(.text+0x5529): undefined reference to `PXML::~PXML()' wivr.cxx:(.text+0x553a): undefined reference to `PXML::~PXML()' collect2: ld returned 1 exit status make: *** [obj_linux_x86_r/woomera] Error 1 root@icetech-server:~/woomera# What's the problem? Any help? Thanks a lot David

1 0

[OpenSER-Users] Replace Openser dns based sip location service by a DHT
by Victor Pascual Ávila 01 Dec '07

01 Dec '07

Hi all, we are working on a P2PSIP prototype and we are considering to couple a DHT Chord-based node with Openser. Current dns based sip location service would be replaced by a DHT, allowing several proxies/registrars to read/write overlays resource records. Any hint or guidelines would be appreciated. Thanks in advance and sorry for cross posting, I'm not sure this issue is related to devel or users. -- Victor Pascual Ávila

1 0

[OpenSER-Users] Changing From: and To: URIs...
by Phil D'Amore 30 Nov '07

30 Nov '07

Hi Everyone... I'm looking for some advice. Let's say I have a To: header in an initial INVITE: To: Bob <sip:8005551212@sip.example.com> Due to carrier requirements, it really needs to be: To: Bob <sip:+18005551212@sip.example.com> I'm not asking how to do it. I know I can use some textops functions to make this happen. I also know that doing this might break something. The uac module already nicely handles the From: header in a way that doesn't break things. I thought about extending this to handle To:, but I worried that the Record-Route header might get too long for some UAs if I wind up having to change both headers. The To tag must also be dealt with somehow, which I think might make it harder to do than From. Using a 302 response is not an option for me, and I'm really trying to avoid getting a B2BUA involved with this right now. Moving from openser 1.1.1 to 1.3 is enough for me without changing my architecture ;). I'm wondering what would be the best way given what openser can do right now (1.3) to re-write that header that doesn't break anything. I'm not adverse to writing some code to make it happen if that is what is needed. Any ideas? Thanks, Phil

3 5

[OpenSER-Users] OpenSER 1.3 - release name
by Bogdan-Andrei Iancu 30 Nov '07

30 Nov '07

Hi, I would like to make a survey to see if people will consider a nice policy to give to the further openser releases not only version numbers, but also names. If so, I would like to be the first to suggest a name for the upcoming openser 1.3 release -"blackcat" (mainly because of the 13 version :D ). Any other suggestions or comments on the topic are welcomed. Thanks and regards, Bogdan

7 11

[OpenSER-Users] OpenSER 1.3 - release date
by Bogdan-Andrei Iancu 30 Nov '07

30 Nov '07

Hi, There are still some bugs that need to be fixed before the release of 1.3. I think 2 weeks should from now on should be enough, so I suggest to try to set the release date to 13th of December. Please let me know your thoughts. Regards and a nice weekend, Bogdan

1 0

[OpenSER-Users] does CPL change SDP parameter on reply message?
by Daniel 30 Nov '07

30 Nov '07

Hi all, if I use <PROXY> in a CPL script, then, how can I change SDP parameter (I want to use media proxy ) in the reply message (200 OK) ? regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

3 5

[OpenSER-Users] CPL and on reply route
by Daniel 30 Nov '07

30 Nov '07

Hi all, I use CPL script with "Proxy" element, like this: <cpl> <incoming> <lookup source="registration"> <success> <proxy ordering="sequential" /> </success> </lookup> </incoming> </cpl> and I use onreply_route(), where I make the changes on 180/200 reply messages in case of NAT (change callee's SDP parameter and change Callee's Contact hf). When an INVITE is forwarded, through the <PROXY> element, to the callee, then reply messagges (180 Ringing and 200OK) hasn't processed by Onreply_route. So I can't make my NAT control. Why? Regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

2 1

[OpenSER-Users] openser 1.3 sometimes fails to start
by Robert Dyck 30 Nov '07

30 Nov '07

I am testing openser 1.3.0-pre1-notls svnversion 2:3201. I believe it is failing while starting the child processes for the modules. If it starts successfully it seems to perform normally. I believe the following excerpt from the console might hold the clue. Thanks Rob Nov 29 14:40:30 [10492] DBG:xlog:child_init: init_child [0] pid [10492] Nov 29 14:40:30 [10492] DBG:core:init_mod_child: type=PROC_MAIN, rank=0, module=avpops Nov 29 14:40:30 [10498] DBG:core:io_watch_add: io_watch_add(0x8160900, 6, 4, 0xb5f824a4), fd_no=0 Nov 29 14:40:30 [10498] DBG:core:io_watch_add: io_watch_add(0x8160900, 7, 4, 0xb5f82530), fd_no=1 Nov 29 14:40:30 [10498] DBG:core:io_watch_add: io_watch_add(0x8160900, 8, 4, 0xb5f825bc), fd_no=2 Nov 29 14:40:30 [10498] DBG:core:io_watch_add: io_watch_add(0x8160900, 9, 4, 0xb5f82648), fd_no=3 Nov 29 14:40:30 [10498] DBG:core:io_watch_add: io_watch_add(0x8160900, 12, 4, 0xb5f826d4), fd_no=4 Nov 29 14:40:30 [10498] DBG:core:io_watch_add: io_watch_add(0x8160900, 11, 4, 0xb5f82760), fd_no=5 Nov 29 14:40:30 [10498] ERROR:core:io_watch_add: epoll_ctl failed: Invalid argument [22] Nov 29 14:40:30 [10498] CRITICAL:core:tcp_main_loop: failed to add process 6 (TCP main) unix socket to the fd list Nov 29 14:40:30 [10498] CRITICAL:core:tcp_main_loop: exiting...Nov 29 14:40:30 [10497] INFO:nathelper:rtpp_test: rtp proxy <unix:/var/run/rt pproxy.sock> found, support for it enabled Nov 29 14:40:30 [10497] DBG:core:init_mod_child: type=CHILD, rank=3, module=xlog Nov 29 14:40:30 [10497] DBG:xlog:child_init: init_child [3] pid [10497] Nov 29 14:40:30 [10497] DBG:core:init_mod_child: type=CHILD, rank=3, module=avpops Nov 29 14:40:30 [10497] DBG:core:io_watch_add: io_watch_add(0x8160a60, 11, 1, (nil)), fd_no=0 Nov 29 14:40:30 [10492] INFO:core:handle_sigs: child process 10498 exited normally, status=255 Nov 29 14:40:30 [10492] INFO:core:handle_sigs: terminating due to SIGCHLD Nov 29 14:40:30 [10493] INFO:core:sig_usr: signal 15 received Memory status (pkg): qm_status (0x8181760): heap size= 1048576 used= 44448, used+overhead=71504, free=977072 max used (+overhead)= 71504 dumping all alloc'ed. fragments: 0. N address=0x8186908 frag=0x8186900 size=32 used=1 1. N address=0x8186938 frag=0x8186930 size=16 used=1 2. N address=0x8186958 frag=0x8186950 size=32 used=1 3. N address=0x8186988 frag=0x8186980 size=32 used=1 4. N address=0x81869b8 frag=0x81869b0 size=16 used=1 5. N address=0x81869d8 frag=0x81869d0 size=16 used=1 6. N address=0x81869f8 frag=0x81869f0 size=32 used=1 7. N address=0x8186a28 frag=0x8186a20 size=32 used=1 8. N address=0x8186a58 frag=0x8186a50 size=64 used=1 9. N address=0x8186aa8 frag=0x8186aa0 size=64 used=1 10. N address=0x8186af8 frag=0x8186af0 size=64 used=1

3 3

Re: [OpenSER-Users] auth_db module - get_ha1(): Error while querying database
by Daniel-Constantin Mierla 30 Nov '07

30 Nov '07

Hello, from the logs seems that the content of subscriber table is corrupted. Please attach /usr/local/etc/openser/dbtext/subscriber file and send it to me. Cheers, Daniel On 11/30/07 11:30, Mihail Gluhowchenko wrote: > Hello, > In file log on level 8 when try 9 openser down. > <list of dir(/usr/local/etc/openser/dbtext)> > acc > active_watchers > address > aliases > dbaliases > domaingrp > gw > gw_grp > lcr > location > missed_calls > pdt > presentity > pua > re_grp > silo > speed_dial > subscriber > trusted > uri > usr_preferences > version > watchers > xcap_xml > <end_list_of_dir(/usr/local/etc/openser/dbtext)> > In README file for dbtext error - > break; replace exit; > modparam("auth_db", "calculate_ha1", 1) > replace > modparam("auth_db", "calculate_ha1", yes) > > Regards, > Mihail Gluhowchenko. > > 2007/11/29, Daniel-Constantin Mierla <daniel(a)voice-system.ro > <mailto:daniel@voice-system.ro>>: > > Hello, > > can you list the content of directory: /usr/local/etc/openser/dbtext ? > > Also, if you can set the debug=9 in cofig, run again, get the > output and > send it to me, would help a lot. > > Thanks, > Daniel > > On 11/28/07 13:46, Mihail Gluhowchenko wrote: > > Hello All; > > I`m trying to tune openser-1.2.2 with dbtext data base > > But have some problems. > > I make db use - textdb.sh > > Then add users use - sc.dbtext > > and add to openser.cfg: > > #set module path > > mpath="/usr/local/lib/openser/modules/" > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > >

1 0

[Serusers] fixup_hostport2proxy
by Tuan Nguyen 30 Nov '07

30 Nov '07

Hi everyone, i have a problem when i try with two version of SER: SER 0.9.6 and SER 2.1to connect with SEMS (the 1. SER proxy ). With SER 0.9.6 i don't have problem when start it, but with SER 2.1 it's so: 0(13446) ERROR: mk_proxy: could not resolve hostname: "localhost" 0(13446) ERROR: fixup_hostport2proxy: bad host name in URI <localhost> I guess, problem locate in this code: log("Forwarding to 5070"); rewritehostport("localhost:5070"); t_relay_to_udp("localhost","5070"); Anyone know that where can i fix this problem with new version of SER? Thanks for responce. Tuan.

2 2

[OpenSER-Users] FYI: [Fwd: [Sip] SIPit 21 summary]
by Klaus Darilion 30 Nov '07

30 Nov '07

-------- Original-Nachricht -------- Betreff: [Sip] SIPit 21 summary Datum: Mon, 19 Nov 2007 14:55:53 -0600 Von: Robert Sparks <rjsparks(a)nostrum.com> An: sip List <sip(a)ietf.org> SIPit 21 was hosted by the BII Group and the Beijing University of Posts and Telecommunications, November 5-9 in Beijing, China. The event was a little smaller than the most recent SIPits: There were 94 attendees from 38 companies visiting from 15 countries. There were 44 teams and around 70 distinct implementations. The majority of the testing I saw was focusing on advanced scenarios rather than basic registration and call setup. As with SIPit 20, the most common area of interoperability problems centered around offer/answer, particularly when attempting to negotiate alternate versions of streams or to explicitly state parameters to be used with a given stream. Forty of the attending teams completed the survey - from those answers: The roles represented (some implementations act in more than one role): 21 endpoints 10 proxy/registrars 3 standalone registrars 5 event servers 4 gateways 6 automaton UAs (voicemail, conference, appserver, etc) 8 b2bua/sbcs 4 UAs with signaling, but without media 1 test/monitoring tool Implementations using each transport for SIP messages: UDP 100% TCP 82% TLS 49% (server auth only) TLS 6% (mutual auth) SCTP 3% DTLS 0% 36% of the implementations supported SIP over IPv6 (up from 25% at SIPit20) 18% supported SIP over IPSec For DNS we had support for: Full RFC3263 : 61% SRV only : 21% A records only : 9% no DNS support : 9% Support for various items: 61% rport 15% sigcomp 22% enum 21% sending multiplexing STUN and SIP 28% receiving multiplexed STUN and SIP 22% RFC4320 fixes 12% identity 70% session-timer There was one implementation of parts of the session-policy framework. There was one implementation of the sip-consent framework. There were two implementations of parts of the sip-config framework (I do not know yet if they worked together). There were three implementations of outbound and four of GRUU. There were two implementations of MSRP present. The endpoints implemented these methods: 100% INVITE, CANCEL, ACK, BYE 87% REGISTER 87% OPTIONS 87% NOTIFY <- still a lot of unsolicted NOTIFYs 87% REFER 77% SUBSCRIBE 77% INFO 70% UPDATE 67% PRACK 47% MESSAGE 33% PUBLISH Support for various extensions in the endpoints: 43% RFC3323 privacy 41% RFC3327 path 13% RFC3840 pref 13% RFC3841 caller-prefs 59% RFC3891 replaces 20% RFC4488 norefersub 0% RFC4538 target-dialog <- There were several folks starting to talk about supporting this Support for various things in the endpoints: 10% ICE (but there was no interoperability - see below) 0% ICE-TCP 13% STUNbis 17% TURN (again, there was no interoperability) 75% symmetric RTP 25% SRTP 0% RTP over DTLS This is how the endpoints answered how they supported multipart/mime: 7% I break if someone sends me multipart/mime 30% I pretend multipart mime doesn't exist if someone sends it to me 20% I ignore multipart/mime, but will proxy it or hand it to my application if it shows up 17% I try to do something useful with multipart/mime I receive, but I never send it 3% I ignore multipart/mime I receive, but I try to send it to do something useful 20% I try to do something useful with multipart/mime I send and receive There were 4 endpoints that would send media over TCP - none of them supported RFC4145 comedia. One of those supported media over TLS. Implementation is all over the map for fork-loop-fix. However, only 3 of the proxy/b2buas present were still vulnerable to the simple form of the attack described in the draft. There were no implementations present with support for location- conveyance or the geopriv common-policy framework. There was one implementation present with support for the RFC4967 dial-string parameter, and one with support for the ecrit service-urn. Of the SIP-Events implementations, the following packages were supported: 62% refer 48% message-summary 38% presence 29% dialog 19% reg 19% conf There was one implementation of each of the following packages: ua-profile certificate/credentials vx-rtcpxp (sipping-rtcp-summary) There was only one implementation of event-lists present. Only 20% of the SIP-Events implementations supported winfo. We had a multiparty sesssion for a full morning focusing on NAT traversal. Basic use of STUN with SIP is hightly interoperable. No two implementations of TURN could even start trying to talk to each other (each having implemented to different points in the recent torrent of changes). I don't think we'll get much implementation feedback until the spec stops making big changes so frequently. No two implementations of ICE worked together. The closest was between two implementations that have worked in the past, but failed during this session when the connectivity checks arrived before the SDP. _______________________________________________ Sip mailing list https://www1.ietf.org/mailman/listinfo/sip This list is for NEW development of the core SIP Protocol Use sip-implementors(a)cs.columbia.edu for questions on current sip Use sipping(a)ietf.org for new developments on the application of sip

2 1

[Serusers] Segmentation Fault (SER Not Starting)
by Alexandr Volkov 29 Nov '07

29 Nov '07

Hi All, I am having trouble when trying to start SER. I am using Ser 0.9.6 on centos4. When I am trying to start it gives: Segmentation Fault; but only if in 'location' table there is a row with callId starting by number: <field name="callid">49c72d7ed685500e7d844d1cf1357dbd</field> if I change this field in database (simple add any character), ser will start. <field name="callid">a49c72d7ed685500e7d844d1cf1357dbd</field> seems like a bug

1 0

[Serusers] ERROR: forward_reply: no 2nd via found in reply - via parsing error
by Tomasz Zieleniewski 29 Nov '07

29 Nov '07

Hi, I have a chcek via parameter set in the ser.cfg file When ser receives reply with such via header it returns ERROR: forward_reply: no 2nd via found in reply This Via seems to be ok? Via: SIP/2.0/UDP 192.168.0.165:7160;branch=z9hG4bK1589.3bc310b3.1,SIP/2.0/UDP 192.168.0.165:7060;branch=z9hG4bK1589.e1306596.0,SIP/2.0/UDP 192.168.129.74 ;rport=5060;branch=z9hG4bK1589.606a3e61.0,SIP/2.0/UDP 192.168.129.74:5160 ;branch=z9hG4bK1589.a5bd9fb4.0,SIP/2.0/UDP 192.168.129.74;branch= z9hG4bK1589.506a3e61.0,SIP/2.0/UDP 192.168.0.102:7060 ;rport=7060;branch=z9hG4bK-d87543-a228576fa32c0b22-1--d87543- What is wrong? Cheers tomasz

1 0

[Serusers] Unexpected disconnect - lost SIP packets?
by Jeremy A 29 Nov '07

29 Nov '07

Hello, I'm need some help with SIP protocol / SER configuration to resolve a problem. I have a system with a SER instance and a SIP client application running on the same box. SER is 10.50.32.10:5060 APP is 10.50.32.10:5070 Remote SIP server 10.1.39.10 Remote phone is 10.1.39.202 The user agent is registered with the SER instance as extension 1009. It has aliases 1001-1008 and is actually called as 231x which maps onto 100x which then connects to extension 1009. The remote phone makes a call to the user agent application via the remote SIP server and the SER instance. It dials 2311 which gets automagically converted to extension 1009 The call is established but the user application times out and detects a disconnect despite the calling phone not hanging up. In my wireshark trace it seems that SER is ignoring some form of keep-alive from the remote SIP server and this is causing the timeout. The lines of interest include 39, 42, 45, 50, 53, 56, 59 Lines 28 and others similar are on the loopback interface and are packets between port 5060 and 5070 Can someone please advise if the SIP protocol has problems and/or if SER configuration has problems? 22 0.218826 10.50.32.10 -> 127.0.0.1 SIP Request: REGISTER sip:localhost.localdomain 23 0.219570 127.0.0.1 -> 10.50.32.10 SIP Status: 200 OK (1 bindings) 24 19.472555 10.1.39.10 -> 10.50.32.10 SIP/SDP Request: INVITE sip:1001@butler.fesa.sto;transport=udp, with session description 25 19.473394 10.50.32.10 -> 10.1.39.10 SIP Status: 100 trying -- your call is important to us 26 19.473596 10.50.32.10 -> 10.50.32.10 SIP/SDP Request: INVITE sip:1009@10.50.32.10:5070;LINEID=1d5a67e38809, with session description 27 19.484244 10.50.32.10 -> 10.50.32.10 SIP Status: 100 Trying 28 19.725476 10.50.32.10 -> 10.50.32.10 SIP Status: 180 Ringing 29 19.725828 10.50.32.10 -> 10.1.39.10 SIP Status: 180 Ringing 32 21.770798 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 33 21.782437 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 38 21.789396 10.50.32.10 -> 10.1.39.202 RTCP Source port: 8001 Destination port: 16391 39 21.800069 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 40 22.278096 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 41 22.278713 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 42 22.298101 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 43 23.285536 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 44 23.288988 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 45 23.307839 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 48 25.298214 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 49 25.299105 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 50 25.317892 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 51 29.306496 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 52 29.306758 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 53 29.328420 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 54 33.314736 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 55 33.318736 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 56 33.338302 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 57 37.323175 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 58 37.324791 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 59 37.347936 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 --> client detects disconnect here Thanks jeremy

3 8

[Serusers] Presence in SIP
by Carlos Angeles 29 Nov '07

29 Nov '07

Hello I have installed Ser-2.0 with the PA module.... I am testing the server with x-lite... When a user is NOTIFYING some event change the server is answering with a Status: 420 Bad Extension.... Is thst SER doesn´t support this kind of presence events? Thanks Carlos _________________________________________________________________ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE

1 0

[OpenSER-Users] Handling re-INVITES --hellp required
by srinivas Antarvedi 29 Nov '07

29 Nov '07

Hello all, i have users one is on global ip and another behind NAT am using asterisk as media server leg 1: caller : Global ip UAC. callee: asterisk leg2: caller :asterisk callee: NATed UAC. sdp of NATed client is handled at openser reply route at first stage when asterisk re-invites the NATed UAC to bridge the two call-Leg's the sdp from NATed UAC is not changed ,, even if i call t_on_reply in the loose route section of the script.. it is still showing privat ip so finally after 2 or 3 sec's there was an end to the dialog can anybody have any idea to handle re-invite's 200 ok SDP mangling? please help me out.. Thanks in advance regards srinivas -- Srinivas Antarvedi

3 3

[OpenSER-Users] auth_db module - get_ha1(): Error while querying database
by Mihail Gluhowchenko 29 Nov '07

29 Nov '07

Hello All; I`m trying to tune openser-1.2.2 with dbtext data base But have some problems. I make db use - textdb.sh Then add users use - sc.dbtext and add to openser.cfg: #set module path mpath="/usr/local/lib/openser/modules/" # use dbtext database loadmodule "dbtext.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" # Uncomment this if you want digest authentication loadmodule "auth.so" loadmodule "auth_db.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- 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 -- # use dbtext database for persistent storage modparam("usrloc", "db_mode", 2) modparam("usrloc|auth_db", "db_url", "dbtext:///usr/local/etc/openser/dbtext") # 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", "user_column", "username") modparam("auth_db", "password_column", "password") modparam("auth_db", "domain_column", "domain") # -- 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"); log(1, "Too Many Hops\n"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); log(1, "Message too big\n"); 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==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("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # 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") { log(1, "this is a request REGISTER users\n"); if (!www_authorize("my.domain", "subscriber")) { log(1, "this is a www_challenge\n"); www_challenge("my.domain", "0"); exit; }; save("location"); exit; }; but when trying REGISTER user use(sjphone) in log i see: Nov 28 17:30:03 devel s-eltex[12797]: this is a request for my.domain users Nov 28 17:30:03 devel s-eltex[12797]: this is a request REGISTER users Nov 28 17:30:03 devel s-eltex[12797]: this is a www_challenge Nov 28 17:30:03 devel s-eltex[12794]: this is a request for my.domain users Nov 28 17:30:03 devel s-eltex[12794]: this is a request REGISTER users Nov 28 17:30:03 devel s-eltex[12794]: get_ha1(): Error while querying database In sjphone LOG i see: Service unavailable server error; What do I adjust not correctly?

2 1

[OpenSER-Users] t_relay can not handle PRACK retransmission
by Klaus Darilion 29 Nov '07

29 Nov '07

Hi! I have encountered this problem at an openser 1.1.1 proxy: Client Proxy Client ---PRACK------> ------PRACK-----> retansmission -----PRACK-------> here at this point (when openser receives the PRACK retransmission) t_relay fails: ERROR: t_newtran: transaction already in process 0xb66e7880 Is this a known problem? Were there some fixes to 1.1 or 1.2 branch? regards klaus

2 4

[OpenSER-Users] Branch limitations
by Daniel 29 Nov '07

29 Nov '07

Hi all, I've read this from iptel website. It is true for SER 0.11.0. What about Openser 1.2 ? " * Script processing of multiple branches on forking Warning ser's request processing language allows to make request decisions based on current URI. When a request if forked to multiple destinations, only the first branch's URI is used as input for script processing. This might lead to unexpected results. Whenever a URI resolves to multiple different next-hop URIs, only the first is processed which may result in handling not appropriate for the other branch. For example, a URI might resolve to an IP phone SIP address and PSTN gateway SIP address. If the IP phone address is the first, then script execution ignores the second branch. If a script includes checking gateway address in request URI, the checks never match. That might result in ignoring of gateway admission control rules or applying them unnecessarily to non-gateway destinations. List of known problems is publicly available at the ser webpage at [70]http://www.iptel.org/ser/ . See the "ISSUES" link. -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

2 1

[OpenSER-Users] OpenSER: Support of advanced RegExpr?
by Helmut Kuper 28 Nov '07

28 Nov '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, is openser supporting advanced regular expressions as described here: http://www.regular-expressions.info/refadv.html regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHTVg64tZeNddg3dwRAl+JAKCst6W4QdNjJ6ZT6fWRaE9otPcX+wCdEKbw iQy1J1Md3CKE7pUgx4lld3k= =sRw7 -----END PGP SIGNATURE-----

2 1

[OpenSER-Users] OpenXCAP: 'presence-rules' PUT sent: '409 Conflict' received
by ddonate 28 Nov '07

28 Nov '07

Hi, I am using OpenSER and openXCAP to handle 'resource list' and 'presence rules' of some buddies. The 'resource-list' works fine: I can GET my file (404 the 1st time, and I PUT one). But I have a problem with the 'presence-rules' PUT, when I receive a 'winfo' NOTIFY (when the user 'john' subscribes to my presence): I send an HTTP request but I receive a '409 Conflict': - My request: PUT /pres-rules/users/diego(a)openser/presence_rules.xml HTTP/1.1 User-Agent: IMS_DialcomClient Accept: text/* Host: 192.168.1.208:8000 Content-Length: 284 ... <?xml version='1.0' encoding='UTF-8'?><ruleset xmlns='urn:ietf:params:xml:ns:presence-rules'><rule id='sip:john@openser_rule'><conditions><identity><one id='sip:john@openser'/></identity></conditions><actions><sub-handling>allow</sub-handling></actions></rule></ruleset> - The traces in the openXCAP are: -------------------- Failed to validate document against XML schema: <string>:1:ERROR:SCHEMASV:SCHEMAV_CVC_ELT_1: Element '{urn:ietf:params:xml:ns:pres-rules}ruleset': No matching global declaration available for the validation root. -------------------- I know it is related to the schema, but I have not managed to fix it... Thanks in advance Diego -- View this message in context: http://www.nabble.com/OpenXCAP%3A-%27presence-rules%27-PUT-sent%3A-%27409-C… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

1 0

[OpenSER-Users] routing table number
by Patrick Miccio 28 Nov '07

28 Nov '07

Hi everyone, I came across another problem when using a newly compiled OpenSER binary (from 1.1.1-notls source with no modifications). When starting OpenSER with -c -f configfile I would get the following errors: ----- 0(18837) parse error (354,1-2): invalid routing table number ----- Looking around with google I found out that this is because in the original config.h the following is set: ------ define RT_NO 40 /* routing tables number */ ------ In my openser.cfg there is already 44 routing tables ;) So my question is: can you somehow set RT_NO in openser.cfg, or is that only possible in config.h + recompiling? digital greetings, Patrick. This e-mail is confidential and may well also be legally privileged. If you have received it in error, you are on notice of its status. Please notify us immediately by reply e-mail and then delete this message from your system. Please do not copy it or use it for any purposes, or disclose its contents to any other person: to do so could be a breach of confidence. Thank you for your cooperation. Information pursuant to paragraph 14 Austrian Companies Code: UPC Austria GmbH; Registered Office: Wolfganggasse 58-60, 1120 Vienna Company Register Number: FN 189858d at the Commercial Court of Vienna

3 3

[Serusers] ser presence question
by Yufei Tao 28 Nov '07

28 Nov '07

Hi I want to be able to use PUBLISH/SUBSCRIBE/NOTIFY for presence information of users. So users can update their presence information by PUBLISH and watchers be informed by NOTIFY when there are changes in the presence status. I am using pa.so, mysql. Not using XCAP. My problems are 1. The OK from SER to PUBLISH has no SIP_ETag. Also SER prints out debug messages saying: publish_presentity: no tuple for 01(a)sertest.com I wonder if there's something wrong with my PUBLISH and PIDF so SER is not processing it? 2. The watcher subscribes to the presentity 01(a)sertest.com. But the NOTIFY message the watcher gets always has basic status being closed, despite that the presentity has PUBLISHed its status as open. I guess this may be due to problem 1 - SER never understood the PUBLISH? Currently my UA sends PUBLISH message: PUBLISH sip:01@sertest.com:5060 SIP/2.0 via: SIP/2.0/UDP 192.168.168.149:40070;branch=z9hG4bK314278337 max-forwards: 70 to: <sip:01@sertest.com:5060> from: <sip:01@sertest.com:5060>;tag=314279337 call-id: cd674de3-0-1196253409 cseq: 46046 PUBLISH contact: <sip:01@192.168.168.149:40070> expires: 3600 user-agent: Red v1.00 0.6.3.42 - von event: presence content-type: application/pidf+xml content-length: 274 <?xml version="1.0" encoding="UTF-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" entity="sip:01@sertest.com"> <tuple id="status"> <status> <basic>open</basic> </status> <contact priority="0.500000">sip:01@sertest.com</contact> </tuple> </presence> and SER replies with the following msg where there is no SIP-ETag field: SIP/2.0 200 OK via: SIP/2.0/UDP 192.168.168.149:40070;branch=z9hG4bK314278338 to: <sip:01@sertest.com:5060>;tag=482760500770a188d9ef565befb876b4-7979 from: <sip:01@sertest.com:5060>;tag=314279338 call-id: cd674de3-0-1196253409 cseq: 46047 PUBLISH Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 192.168.168.241:5060 "Noisy feedback tells: pid=14095 req_src_ip=192.168.168.149 req_src_port=40070 in_uri=sip:01@sertest.com:5060 out_uri=sip:01@sertest.com:5060 via_cnt==1" The parts in my ser.cfg that are relevant to SUBSCRIBE and PUBLISH: modparam("usrloc", "db_mode", 2) modparam("usrloc", "timer_interval", 10) modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) # for SUBSCRIBE/NOTIFY modparam("pa", "default_expires", 3600) modparam("pa", "use_db", 1) modparam("pa", "db_url", "mysql://ser:heslo@localhost/ser") modparam("pa", "pa_domain", "sertest.com")# -- same as or different from registrar ... ... ... ... if (method=="SUBSCRIBE") { if (!t_newtran()) { sl_reply_error(); break; }; handle_subscription("registrar"); break; } if (method=="PUBLISH") { if (!t_newtran()) { log(1, "newtran error\n"); sl_reply_error(); }; handle_publish("registrar"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; };

2 3

[OpenSER-Users] OpenSer: Group Module Parameter "re_exp_column"
by Helmut Kuper 28 Nov '07

28 Nov '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, the example in OpenSER 1.3 and devel module doc for module group is wrong. It gives this line as example: modparam("group", "reg_exp_column", "re") but it should be: modparam("group", "reg_exp_column", "reg_exp") minor thing, but anoying, when u just copy and paste modparams into your config without reading the chapter titles .... Maybe there should be a block with all modparams and their default values in each module's doc... regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHTDhm4tZeNddg3dwRAh/5AKC0r9jbisrrR+YZC0BKtk6hzGz8oQCdEtcv 2oHfwzkoe43hGB4YO4dv6mM= =ysim -----END PGP SIGNATURE-----

4 5

[OpenSER-Users] multiple contact
by Daniel 28 Nov '07

28 Nov '07

Hi all, I have a user A registering with 2 differrent contact, one of each behind NAT. If I set "modparam("registrar", "append_branches", 1)", and if q-values are setted as 1, the proxy, aftel lookup (location) will call both of contacts in parallel mode. But how can I manage the Natted contact and set mediaproxy ? regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

2 1

[OpenSER-Users] editing the via header
by Othmar Pasteka 28 Nov '07

28 Nov '07

hello, I would like to know when the via header of the openser proxy is inserted (the new one). and/or if there is a practical way to manipulate it by me in a different module, like that some additional information is added after the branch tag. I browsed through the documentation and also googled but couldn't find valuable information how the via headers are treated in this respect. I found the reference in parser/msg_parser.h:struct sip_msg, that they reference via1/via2 but I assume via1 is not the current one which will be added by the local openser proxy. any additional information is greatly appreciated. thanks. regards, Othmar

2 1

[OpenSER-Users] [CPL] Why aren't loaded bflags when <lookup source="registration"> ?
by Iñaki Baz Castillo 28 Nov '07

28 Nov '07

Hi, I set: modparam("cpl-c","proxy_route", 10) and in route[10] I set RtpProxy if flag(5) (NAT in the caller) or bflag(6) (NAT in the called) is present. But unfortunatelly the bflag(6) is never set when usingCPL. For example the following and very simple CPL: <cpl> <incoming> <lookup source="registration"> <success> <proxy /> </success> </lookup> </incoming> </cpl> I understand that the operation of CPL is "encapsulated" and so, but since we can spec some parameters as: modparam("cpl-c","lookup_domain","location") modparam("cpl-c","lookup_append_branches", 1) why then can't bflags be loaded? If it is not possible then I can't use <lookup source="registration"> and need to do, neccesarialy, a spiral with: <location url="sip:SAME_USER;cpl=no"> <proxy /> </location> and don't apply CPL in the spiral incoming message if URI has "cpl=no". Any explanation of this limitation could be really great. Thanks a lot. -- Iñaki Baz Castillo

2 1

Re: [Serusers] nathelper doesn't ping registered users
by Jiri Kuthan 27 Nov '07

27 Nov '07

I'm kind of finding this conversation a bit confusing -- what is cl in this context actually good for? I mean the "boundary proxy" (what is it actually good for?) can have shadow usrloc (it is presumably in the path anyhow?) if natping shall be sent from there. sip-based natping can be run from the "inside proxy" as well, very much as any other SIP request. That causes heavier traffic but also increases the success chance (for example behind my firewall and with my SIP phone, non-SIP pings does not help because client doesn't send enough outbound traffic) Just wild guesses -- I'm not really understanding the specifics of this proxy chain. -jiri At 15:13 27/11/2007, Tomasz Zieleniewski wrote: >unfortunatelly it is not available >any chance to have it for ser 2.0?? > >regards >tomasz > >On Nov 27, 2007 2:51 PM, Greger V. Teigre <greger(a)teigre.com> wrote: >> >> usrloc-cl is a cacheless version of usrloc module, i.e. nothing is stored >> in memory. I believe Martin Hoffmann took over maintenance of it, but I'm >> not sure about the status on 2.0.0. If your edge proxy uses usrloc-cl >> instead of usrloc, it will pull locations to ping from the DB instead of >> from memory. >> >> >> g-) >> >> Tomasz Zieleniewski wrote: >> what is usrloc-cl?? >> What do you mean by that and how can I do that? >> >> -tomasz >> >> On Nov 27, 2007 8:58 AM, Greger V. Teigre <greger(a)teigre.com> wrote: >> >> >> You use usrloc-cl? >> If not, unless you replicate, the non-REGISTER handling server will not >> have the usrloc in memory. >> g-) >> >> >> >> Tomasz Zieleniewski wrote: >> Hi Jiri, >> >> On Nov 25, 2007 10:08 PM, Jiri Kuthan <jiri(a)iptel.org> wrote: >> >> >> At 10:56 25/11/2007, Tomasz Zieleniewski wrote: >> >> >> Hi, >> >> I don't have those lines because I have a registrar on seperate SIP Proxy. >> This proxy here is the boundary one, but they both use the same database >> and have access to location table common. >> >> Well, I'm not sure I'm getting the context of this configuration so I don't >> feel certain with providing some hints (in particular two proxy serveer >> in a series both using usrloc seems unusual to me). A hand-waving estimate >> is you need to configure the "boundary proxy" to change all natted clients >> to form "as seen by the network" and the "actual proxy" to send pings to >> all contacts. >> >> >> Boundary proxy uses usrloc only for the purpose of pinging the natted >> users. >> Nathelper module requires usrloc. Because all reuqest travels 'boundary >> proxy' >> before entering network the 'boundary proxy' must ping users to keep >> my nat bindings >> right. they are both on seperate machines. >> >> -tomek >> >> >> -jiri >> >> >> >> >> >> -tomasz >> >> On Nov 25, 2007 10:40 AM, Jiri Kuthan <jiri(a)iptel.org> wrote: >> >> >> Hi Tomasz, >> >> well it could be that reusing an existing config and commiting than your >> specific >> changes to it would save you quite some laborous exercises rather than >> writing >> it from scratch.... >> http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/etc/ser-oob.cfg?re… >> >> The lines in question are (out of which I think you are missing the >> registrar params) >> >> >> FLAG_NAT : 3, # the UAC is behind a NAT >> >> ... >> # identify natted contacts using a flag >> modparam("registrar", "load_nat_flag", "FLAG_NAT") >> modparam("registrar", "save_nat_flag", "FLAG_NAT") >> ... >> modparam("rr", "cookie_filter", "(account|uac_nat)") >> ... >> modparam("nathelper", "ping_nated_only", 1 ) >> # if this option is not set, simple 4-bytes ping is sent >> modparam("nathelper", "natping_method", "OPTIONS" ) >> #temporary statefull natping test (only in future versions) >> #modparam("nathelper", "natping_stateful", 1) >> ... >> >> setflag(FLAG_NAT); >> >> >> -jiri >> >> >> At 22:20 24/11/2007, Tomasz Zieleniewski wrote: >> >> >> I'm sorry for the CC >> here is my config: >> >> debug=3 # debug level (cmd line: -dddddddddd) >> memdbg=5 >> memlog=5 >> log_facility=LOG_LOCAL0 >> fork=yes >> log_stderror=no >> check_via=yes # (cmd. line: -v) >> dns=no # (cmd. line: -r) >> rev_dns=no # (cmd. line: -R) >> port=5060 >> tls_port_no=5061 >> children=4 >> user=ser >> group=ser >> mhomed=yes >> disable_tcp=no >> tcp_accept_aliases=yes >> enable_tls=no >> sip_warning=0 >> reply_to_via=no >> alias="voip.touk.pl" >> >> dns_retr_time=1 >> dns_retr_no=1 >> dns_servers_no=1 >> dns_use_search_list=no >> dns_try_ipv6=no >> # dns cache & failover >> use_dns_cache=on >> use_dns_failover=on >> # dns_cache_flags=0 >> dns_cache_negative_ttl=300 >> dns_cache_min_ttl=60 >> dns_cache_max_ttl=86400 # 1 day >> dns_cache_mem=2048 # 2 MB >> dns_cache_gc_interval=60 >> >> tcp_connection_lifetime=3600 >> tcp_max_connections=2048 >> tcp_connect_timeout=1 >> >> loadmodule ... >> .... >> >> # specify the path to you database here >> modparam("acc_db|auth_db|avp_db|domain|gflags|usrloc|uri_db|speeddial|msilo|lcr", >> "db_url", "mysql://ser:ser@localhost/ser") >> >> # -- usrloc params -- >> modparam("usrloc", "db_mode", 1) >> >> # -- registrar params -- >> modparam("registrar", "default_expires", 300) >> modparam("registrar", "min_expires", 60) >> modparam("registrar", "max_expires", 300) >> modparam("registrar", "append_branches", 1) >> modparam("registrar", "received_param", "rcv") >> modparam("registrar", "max_contacts", 2) >> >> # -- auth params -- >> modparam("auth_db", "calculate_ha1", yes) >> modparam("auth_db", "password_column", "password") >> >> # -- rr params -- >> modparam("rr", "enable_full_lr", 1) >> modparam("rr", "cookie_filter", "uac_nat") >> modparam("rr", "cookie_secret", "secret") >> modparam("rr", "user_part_avp", "route_user") >> modparam("rr", "next_route_avp", "route_next") >> >> # -- gflags params -- >> modparam("gflags", "load_global_attrs", 1) >> >> # -- domain params -- >> modparam("domain", "load_domain_attrs", 1) >> >> # -- ctl params -- >> modparam("ctl", "binrpc", "unixs:/tmp/ser-bsp_ctl") >> modparam("ctl", "fifo", "fifo:/tmp/ser-bsp_fifo") >> # modparam("ctl", "binrpc", "tcp:localhost:2046") >> modparam("ctl", "mode", 0777) >> >> # -- tm params -- >> modparam("tm", "restart_fr_on_each_reply", 1) >> >> # -- auth_radius params -- >> modparam("auth_radius", "radius_config", >> "/usr/local/etc/radiusclient-ng/radiusclient.conf") >> modparam("auth_radius", "service_type", 15) >> >> # -- options params -- >> modparam("options", "accept", "*/*") >> modparam("options", "accept_encoding", "") >> modparam("options", "accept_language", "en") >> >> # -- ratelimit params -- >> modparam("ratelimit","timer_interval", 10) >> modparam("ratelimit","invite_limit", 100) >> modparam("ratelimit","register_limit", 100) >> modparam("ratelimit","subscribe_limit", 100) >> >> # -- xmlrpc params -- >> modparam("xmlrpc", "route","RPC") >> >> # -- nathelper params -- >> modparam("nathelper", "natping_interval", 20) >> modparam("nathelper", "ping_nated_only", 0) >> modparam("nathelper", "rtpproxy_disable", 1) >> modparam("nathelper", "tcpping_crlf", 1) >> >> # -- tls params -- >> modparam("tls", "config", "/home/ser/ser/ca/tls-bsp.cfg") >> modparam("tls", "tls_force_run", 0) >> modparam("tls", "tls_log",3) >> modparam("tls", "handshake_timeout", 10) >> modparam("tls", "send_timeout", 10) >> modparam("tls", "tls_disable_compression", 0) >> #modparam("tls", "private_key", "/etc/certs/key.pem") >> #modparam("tls", "certificate", "/etc/certs/cert.pem") >> #modparam("tls", "ca_list", "/etc/certs/ca_list.pem") >> #modparam("tls", "require_certificate", 0) >> #modparam("tls", "verify_certificate", 1) >> #modparam("tls", "tls_method", "TLSv1") >> >> route{ >> ... >> } >> >> On Nov 24, 2007 9:35 PM, Atle Samuelsen <clona(a)cyberhouse.no> wrote: >> >> >> Hi, >> >> Always CC the list. >> >> then I dont know.. post your full config so people can see if they see >> something >> -A >> >> * Tomasz Zieleniewski <tzieleniewski(a)gmail.com> [071124 20:38]: >> >> >> >> Hi, >> >> I had that parameter set to 0, but still no udp packets... >> >> On Nov 24, 2007 6:43 PM, Atle Samuelsen <clona(a)cyberhouse.no> wrote: >> >> >> * Tomasz Zieleniewski <tzieleniewski(a)gmail.com> [071124 16:50]: >> >> >> >> HI, >> >> I have the following nathelper settings: >> modparam("nathelper", "natping_interval", 20) >> modparam("nathelper", "ping_nated_only", 1) >> modparam("nathelper", "rtpproxy_disable", 1) >> modparam("nathelper", "tcpping_crlf", 1) >> >> But i cant see any udp packets being sent to registered users. >> Please point me what did I miss. >> >> Hi, >> >> without seeing any more of your config, I'm guessing that you do not set >> the natflag, so if you change ping_nated_only to 0, I think you will see >> some udp packets. >> >> - Atle >> (shot in the whild world of sip) >> >> >> >> >> Best >> Tomasz >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >> >> -- >> Jiri Kuthan http://iptel.org/~jiri/ >> >> >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >> >> -- >> Jiri Kuthan http://iptel.org/~jiri/ >> >> >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> -- Jiri Kuthan http://iptel.org/~jiri/

1 0

Re: [Serusers] nathelper doesn't ping registered users
by Jiri Kuthan 27 Nov '07

27 Nov '07

At 10:56 25/11/2007, Tomasz Zieleniewski wrote: >Hi, > >I don't have those lines because I have a registrar on seperate SIP Proxy. >This proxy here is the boundary one, but they both use the same database >and have access to location table common. Well, I'm not sure I'm getting the context of this configuration so I don't feel certain with providing some hints (in particular two proxy serveer in a series both using usrloc seems unusual to me). A hand-waving estimate is you need to configure the "boundary proxy" to change all natted clients to form "as seen by the network" and the "actual proxy" to send pings to all contacts. -jiri >-tomasz > >On Nov 25, 2007 10:40 AM, Jiri Kuthan <jiri(a)iptel.org> wrote: >> Hi Tomasz, >> >> well it could be that reusing an existing config and commiting than your specific >> changes to it would save you quite some laborous exercises rather than writing >> it from scratch.... >> http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/etc/ser-oob.cfg?re… >> >> The lines in question are (out of which I think you are missing the registrar params) >> >> >> FLAG_NAT : 3, # the UAC is behind a NAT >> >> ... >> # identify natted contacts using a flag >> modparam("registrar", "load_nat_flag", "FLAG_NAT") >> modparam("registrar", "save_nat_flag", "FLAG_NAT") >> ... >> modparam("rr", "cookie_filter", "(account|uac_nat)") >> ... >> modparam("nathelper", "ping_nated_only", 1 ) >> # if this option is not set, simple 4-bytes ping is sent >> modparam("nathelper", "natping_method", "OPTIONS" ) >> #temporary statefull natping test (only in future versions) >> #modparam("nathelper", "natping_stateful", 1) >> ... >> >> setflag(FLAG_NAT); >> >> >> -jiri >> >> >> At 22:20 24/11/2007, Tomasz Zieleniewski wrote: >> >I'm sorry for the CC >> >here is my config: >> > >> >debug=3 # debug level (cmd line: -dddddddddd) >> >memdbg=5 >> >memlog=5 >> >log_facility=LOG_LOCAL0 >> >fork=yes >> >log_stderror=no >> >check_via=yes # (cmd. line: -v) >> >dns=no # (cmd. line: -r) >> >rev_dns=no # (cmd. line: -R) >> >port=5060 >> >tls_port_no=5061 >> >children=4 >> >user=ser >> >group=ser >> >mhomed=yes >> >disable_tcp=no >> >tcp_accept_aliases=yes >> >enable_tls=no >> >sip_warning=0 >> >reply_to_via=no >> >alias="voip.touk.pl" >> > >> >dns_retr_time=1 >> >dns_retr_no=1 >> >dns_servers_no=1 >> >dns_use_search_list=no >> >dns_try_ipv6=no >> ># dns cache & failover >> >use_dns_cache=on >> >use_dns_failover=on >> ># dns_cache_flags=0 >> >dns_cache_negative_ttl=300 >> >dns_cache_min_ttl=60 >> >dns_cache_max_ttl=86400 # 1 day >> >dns_cache_mem=2048 # 2 MB >> >dns_cache_gc_interval=60 >> > >> >tcp_connection_lifetime=3600 >> >tcp_max_connections=2048 >> >tcp_connect_timeout=1 >> > >> >loadmodule ... >> >.... >> > >> ># specify the path to you database here >> >modparam("acc_db|auth_db|avp_db|domain|gflags|usrloc|uri_db|speeddial|msilo|lcr", >> >"db_url", "mysql://ser:ser@localhost/ser") >> > >> ># -- usrloc params -- >> >modparam("usrloc", "db_mode", 1) >> > >> ># -- registrar params -- >> >modparam("registrar", "default_expires", 300) >> >modparam("registrar", "min_expires", 60) >> >modparam("registrar", "max_expires", 300) >> >modparam("registrar", "append_branches", 1) >> >modparam("registrar", "received_param", "rcv") >> >modparam("registrar", "max_contacts", 2) >> > >> ># -- auth params -- >> >modparam("auth_db", "calculate_ha1", yes) >> >modparam("auth_db", "password_column", "password") >> > >> ># -- rr params -- >> >modparam("rr", "enable_full_lr", 1) >> >modparam("rr", "cookie_filter", "uac_nat") >> >modparam("rr", "cookie_secret", "secret") >> >modparam("rr", "user_part_avp", "route_user") >> >modparam("rr", "next_route_avp", "route_next") >> > >> ># -- gflags params -- >> >modparam("gflags", "load_global_attrs", 1) >> > >> ># -- domain params -- >> >modparam("domain", "load_domain_attrs", 1) >> > >> ># -- ctl params -- >> >modparam("ctl", "binrpc", "unixs:/tmp/ser-bsp_ctl") >> >modparam("ctl", "fifo", "fifo:/tmp/ser-bsp_fifo") >> ># modparam("ctl", "binrpc", "tcp:localhost:2046") >> >modparam("ctl", "mode", 0777) >> > >> ># -- tm params -- >> >modparam("tm", "restart_fr_on_each_reply", 1) >> > >> ># -- auth_radius params -- >> >modparam("auth_radius", "radius_config", >> >"/usr/local/etc/radiusclient-ng/radiusclient.conf") >> >modparam("auth_radius", "service_type", 15) >> > >> ># -- options params -- >> >modparam("options", "accept", "*/*") >> >modparam("options", "accept_encoding", "") >> >modparam("options", "accept_language", "en") >> > >> ># -- ratelimit params -- >> >modparam("ratelimit","timer_interval", 10) >> >modparam("ratelimit","invite_limit", 100) >> >modparam("ratelimit","register_limit", 100) >> >modparam("ratelimit","subscribe_limit", 100) >> > >> ># -- xmlrpc params -- >> >modparam("xmlrpc", "route","RPC") >> > >> ># -- nathelper params -- >> >modparam("nathelper", "natping_interval", 20) >> >modparam("nathelper", "ping_nated_only", 0) >> >modparam("nathelper", "rtpproxy_disable", 1) >> >modparam("nathelper", "tcpping_crlf", 1) >> > >> ># -- tls params -- >> >modparam("tls", "config", "/home/ser/ser/ca/tls-bsp.cfg") >> >modparam("tls", "tls_force_run", 0) >> >modparam("tls", "tls_log",3) >> >modparam("tls", "handshake_timeout", 10) >> >modparam("tls", "send_timeout", 10) >> >modparam("tls", "tls_disable_compression", 0) >> >#modparam("tls", "private_key", "/etc/certs/key.pem") >> >#modparam("tls", "certificate", "/etc/certs/cert.pem") >> >#modparam("tls", "ca_list", "/etc/certs/ca_list.pem") >> >#modparam("tls", "require_certificate", 0) >> >#modparam("tls", "verify_certificate", 1) >> >#modparam("tls", "tls_method", "TLSv1") >> > >> >route{ >> >... >> >} >> > >> >On Nov 24, 2007 9:35 PM, Atle Samuelsen <clona(a)cyberhouse.no> wrote: >> >> Hi, >> >> >> >> Always CC the list. >> >> >> >> then I dont know.. post your full config so people can see if they see >> >> something >> >> -A >> >> >> >> * Tomasz Zieleniewski <tzieleniewski(a)gmail.com> [071124 20:38]: >> >> >> >> > Hi, >> >> > >> >> > I had that parameter set to 0, but still no udp packets... >> >> > >> >> > On Nov 24, 2007 6:43 PM, Atle Samuelsen <clona(a)cyberhouse.no> wrote: >> >> > > * Tomasz Zieleniewski <tzieleniewski(a)gmail.com> [071124 16:50]: >> >> > > >> >> > > > HI, >> >> > > > >> >> > > > I have the following nathelper settings: >> >> > > > modparam("nathelper", "natping_interval", 20) >> >> > > > modparam("nathelper", "ping_nated_only", 1) >> >> > > > modparam("nathelper", "rtpproxy_disable", 1) >> >> > > > modparam("nathelper", "tcpping_crlf", 1) >> >> > > > >> >> > > > But i cant see any udp packets being sent to registered users. >> >> > > > Please point me what did I miss. >> >> > > >> >> > > Hi, >> >> > > >> >> > > without seeing any more of your config, I'm guessing that you do not set >> >> > > the natflag, so if you change ping_nated_only to 0, I think you will see >> >> > > some udp packets. >> >> > > >> >> > > - Atle >> >> > > (shot in the whild world of sip) >> >> > > >> >> > > >> >> > > > >> >> > > > Best >> >> > > > Tomasz >> >> > > > _______________________________________________ >> >> > > > Serusers mailing list >> >> > > > Serusers(a)lists.iptel.org >> >> > > > http://lists.iptel.org/mailman/listinfo/serusers >> >> > > >> >> >> >_______________________________________________ >> >Serusers mailing list >> >Serusers(a)lists.iptel.org >> >http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> -- >> Jiri Kuthan http://iptel.org/~jiri/ >> >> >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

4 6

[OpenSER-Users] OpenSER Core CookBook link on openser.org
by Ovidiu Sas 27 Nov '07

27 Nov '07

It would be really handy to have a link to the CoreBook page from the main doc page: http://www.openser.org/docs/ This page starts with a shortcut for the wiki (the first link), but after that it would be nice to have links to the CoreBook/version page just like we have for modules, PV and transformations. Regards, Ovidiu Sas

2 1

[OpenSER-Users] OpensSer Release Name
by Walter Rodrigues Filho 27 Nov '07

27 Nov '07

Dear Bogdan - Since OpenSer has deep roots in Rumania, and considering the suggestions like: * blackcat (because of the 13 thing...) * Santa "Curz"(instead of Cruz it looks like a CURSE...) * deadly SINS (like luxuria, gula...) They are all drive you into either BAD or superstitious concepts.... heheh IN THIS LINE, I would suggest then people names related somehow to Rumanian history (eventough they may have not been Rumanians but after all OpenSer is OPEN to other nationalities to improve and contribute on it...) * DRACULA * IVAN * ATTILA * CORVINUS All of them have had something to do with Rumania in certain time like all members of this opensource community. IN CONCLUSION: All of them is good but "blackcat" is much to much similar to american missiles (tomcat and things..). We really should have here a LATIN BASED NAME or Rumanian history related name (eventhough the names could be of temporary CONQUERORS of Rumania...hehe). END OF LINE> Walter Rodrigues Filho Brazil

4 3

[OpenSER-Users] wesip startup error
by Daniel 27 Nov '07

27 Nov '07

Hi all, I'm using wesip with openser 1.2.2. I've configured both of the configuration files (openser.cfg with SEAS module and server.xml) but when I run the wesip startup script I get this Exception: Starting WESIP Application Server VozTelecom... [root@SERVER conf]# Exception during startup processing java.lang.reflect.InvocationTargetException at java.lang.reflect.Method.invoke (libgcj.so.7) at org.apache.catalina.startup.Bootstrap.main (Bootstrap.java:156) Caused by: java.lang.NoSuchMethodError: method java.lang.Class.getSimpleNamewith signature ()Ljava.lang.String; was not found. at org.apache.catalina.connector.http.HttpConnector.start ( HttpConnector.java:1194) at org.apache.catalina.core.StandardService.start (StandardService.java :452) at org.apache.catalina.core.StandardServer.start (StandardServer.java :2142) at org.apache.catalina.startup.Catalina.start (Catalina.java:474) at org.apache.catalina.startup.Catalina.execute (Catalina.java:358) at org.apache.catalina.startup.Catalina.process (Catalina.java:129) at java.lang.reflect.Method.invoke (libgcj.so.7) ...1 more Can somebody help me? Regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

2 1

Re: [OpenSER-Users] OpenSER 1.3 - release name
by lu 27 Nov '07

27 Nov '07

Hi Guys, I like the whole idea of naming and my self i propose this great OpenSource Tool to be Code named "UHURU". It is a Swahili word which means Freedom, it can be freedom of anything, be it colonialization, of expression, way of doing things etc.etc. ....am not John the Baptist though..teh! teh!..teh!.. WBR, LU.

1 0

[Serusers] 483 Too many Hops Error with SER Presence Agent
by Carlos Angeles 27 Nov '07

27 Nov '07

Hi... I have installed SER-0.10.99 with presence capabilities following the presence hadnbook, the server is running OK, but I have problems when I tried to register a client (x-lite). I am getting 483: Too many hops... What can be the problem????? Thank you for your help, Carlos Config file: 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)) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=2 alias="test-domain.com" #user=ser #group=ser #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) #tcp_poll_method="sigio_rt" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/base/ser/directory/lib/ser/modules//sl.so" loadmodule "/base/ser/directory/lib/ser/modules/avp.so" loadmodule "/base/ser/directory/lib/ser/modules//avpops.so" loadmodule "/base/ser/directory/lib/ser/modules//tm.so" loadmodule "/base/ser/directory/lib/ser/modules//rr.so" loadmodule "/base/ser/directory/lib/ser/modules//maxfwd.so" loadmodule "/base/ser/directory/lib/ser/modules//usrloc.so" loadmodule "/base/ser/directory/lib/ser/modules//registrar.so" loadmodule "/base/ser/directory/lib/ser/modules//textops.so" loadmodule "/base/ser/directory/lib/ser/modules//mysql.so" loadmodule "/base/ser/directory/lib/ser/modules//dialog.so" loadmodule "/base/ser/directory/lib/ser/modules//rls.so" loadmodule "/base/ser/directory/lib/ser/modules/pa.so" loadmodule "/base/ser/directory/lib/ser/modules//presence_b2b.so" loadmodule "/base/ser/directory/lib/ser/modules//uri.so" loadmodule "/base/ser/directory/lib/ser/modules//uri_db.so" loadmodule "/base/ser/directory/lib/ser/modules//domain.so" loadmodule "/base/ser/directory/lib/ser/modules//fifo.so" loadmodule "/base/ser/directory/lib/ser/modules//xmlrpc.so" loadmodule "/base/ser/directory/lib/ser/modules//xlog.so" #loadmodule "/base/ser/directory/lib/ser/modules//unixsock.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/base/ser/directory/lib/ser/modules//auth.so" loadmodule "/base/ser/directory/lib/ser/modules//auth_db.so" loadmodule "/base/ser/directory/lib/ser/modules//msilo.so" # ----------------- setting module-specific parameters --------------- # modparam("msilo","registrar","sip:registrar@test-domain.com") modparam("msilo","use_contact",0) modparam("msilo","expire_time",7200) # -- usrloc params -- # -- 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) modparam("rls", "min_expiration", 200) modparam("rls", "max_expiration", 300) modparam("rls", "default_expiration", 300) modparam("rls", "auth", "none") modparam("rls", "xcap_root", "http://localhost/xcap") modparam("rls", "reduce_xcap_needs", 1) modparam("rls", "db_mode", 0) modparam("rls", "db_url", "mysql://ser:heslo@localhost:3306/ser") modparam("pa", "use_db", 0) # allow storing authorization requests for offline users into database modparam("pa", "use_offline_winfo", 1) # how often try to remove old stored authorization requests modparam("pa", "offline_winfo_timer", 600) # how long stored authorization requests live modparam("pa", "offline_winfo_expiration", 600) modparam("pa", "db_url", "mysql://ser:heslo@localhost:3306/ser") # mode of PA authorization: none, implicit or xcap modparam("pa", "auth", "xcap") modparam("pa", "auth_xcap_root", "http://localhost/xcap") # do not authorize watcherinfo subscriptions modparam("pa", "winfo_auth", "none") # use only published information if set to 0 modparam("pa", "use_callbacks", 1) # don't accept internal subscriptions from RLS, ... modparam("pa", "accept_internal_subscriptions", 0) # maximum value of Expires for subscriptions modparam("pa", "max_subscription_expiration", 600) # maximum value of Expires for publications modparam("pa", "max_publish_expiration", 120) # how often test if something changes and send NOTIFY modparam("pa", "timer_interval", 10) # route for generated SUBSCRIBE requests for presence modparam("presence_b2b", "presence_route", "") # waiting time from error to new attepmt about SUBSCRIBE modparam("presence_b2b", "on_error_retry_time", 60) # how long wait for NOTIFY with Subscription-Status=terminated after unsubscribe modparam("presence_b2b", "wait_for_term_notify", 33) # how long before expiration send renewal SUBSCRIBE request modparam("presence_b2b", "resubscribe_delta", 30) # minimal time to send renewal SUBSCRIBE request from receiving previous response modparam("presence_b2b", "min_resubscribe_time", 60) # default expiration timeout modparam("presence_b2b", "default_expiration", 3600) # process internal subscriptions to presence events modparam("presence_b2b", "handle_presence_subscriptions", 1) modparam("usrloc", "db_mode", 0) modparam("domain", "db_mode", 1) modparam("domain|uri_db|acc|auth_db|usrloc|msilo", "db_url", "mysql://ser:heslo@localhost:3306/ser") modparam("fifo", "fifo_file", "/tmp/ser_fifo") # ------------------------- request routing logic ------------------- # main routing logic route{ # XML RPC if (method == "POST" || method == "GET") { create_via(); dispatch_rpc(); break; } # 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(); # 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); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (!lookup_domain("To")) { xlog("L_ERR", "Unknown domain to: %tu from: %fu\n"); route(1); break; } if (method=="SUBSCRIBE") { if (!t_newtran()) { sl_reply_error(); break; }; if (@to.tag=="") { # only for new subscriptions (with empty to tag) if (lookup_user("To")) { # existing user -> it is subscription to PA if (handle_subscription("registrar")) { if ((@msg.event=~"presence\.winfo")) { # new watcher info subscription # sends one watcher info NOTIFY message with all saved authorization requests xlog("L_ERR", "dumping stored winfo to %fu\n"); dump_stored_winfo("registrar", "presence"); } else { # new presence subscription if ((@msg.event=~"presence") && (%subscription_status=="pending")) { # if offline user and new pending subscription if (!target_online("registrar")) { #%subscription_status="waiting"; # store it as waiting subscription xlog("L_ERR", "storing 'pending' winfo to: %tu, from: %fu\n"); store_winfo("registrar"); } } } } break; } if ((@msg.supported=~"eventlist")) { # such user doesn't exist and Supported header field # -> probably RLS subscription if (lookup_domain("From")) { if (lookup_user("From")) { if (is_simple_rls_target("$uid-list")) { # log(1, "it is simple subscription!\n"); # takes From UID and makes XCAP query for user's # list named "default" if (!query_resource_list("default")) { t_reply("404", "No such user list"); break; } } } } if (!have_flat_list()) { # query_resource_list failed or was not called # do standard RLS query acording to To/AOR if (!query_rls_services()) { log(1, "XCAP query failed\n"); t_reply("404", "No such list URI"); break; } } # uncomment this if you want to authenticate first SUBSCRIBE request to resource list # if (!proxy_authenticate("test-domain.com", "credentials")) { # proxy_challenge( "test-domain.com", "0"); # break; # }; handle_rls_subscription("1"); } else { # not resource list subscription -> invalid user xlog("L_ERR", "subscription to invalid user %tu\n"); t_reply("404", "User not found"); } break; } else { # renewal subscriptions - try to handle it as RLS and if failed, handle it as PA subscription # FIXME: better will be test like existing_rls_subscription() # and existing_subscription("registrar") if (!handle_rls_subscription("0")) { handle_subscription("registrar"); } break; } }; # get user (common for all other messages than SUBSCRIBE) if (!lookup_user("To")) { # log(1, "Unknown user - message should be forwarded?"); # # break; append_hf("P-hint: unknown user\r\n"); route(1); break; } if (method=="PUBLISH") { if (!t_newtran()) { # log(1, "newtran error\n"); sl_reply_error(); break; }; handle_publish("registrar"); # deliver messages to online user # TODO: only if user goes from offline to online? if (target_online("registrar")) { # log(1, "Dumping stored messages\n"); # dump stored messages - route it through myself (otherwise routed via DNS!) if (m_dump("sip:127.0.0.1")) { xlog("L_ERR", "MSILO: offline messages for %fu dumped\n"); } } break; }; if (method=="NOTIFY") { if (!t_newtran()) { log(1, "newtran error\n"); sl_reply_error(); break; }; # handle notification sent in internal subscriptions (presence_b2b) if (!handle_notify()) { t_reply("481", "Unable to handle notification"); } break; }; if (method=="MESSAGE") { if (authorize_message("http://localhost/xcap")) { # use usrloc for delivery if (lookup("location")) { log(1, "Delivering MESSAGE using usrloc\n"); t_on_failure("1"); if (!t_relay()) { sl_reply_error(); } break; } else { # store messages for offline user xlog("L_ERR", "MSILO: storing MESSAGE for %tu\n"); if (!t_newtran()) { log(1, "newtran error\n"); sl_reply_error(); break; }; # store only text messages NOT isComposing... ! if (search("^(Content-Type|c):.*application/im-iscomposing\+xml.*")) { log(1, "it is only isComposing message - ignored\n"); t_reply("202", "Ignored"); break; } if (m_store("0", "sip:127.0.0.1")) { # log(1, "MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; } else { log(1, "MSILO: error storing offline message\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; break; } break; } else { # log(1, "unauthorized message\n"); sl_reply("403", "Forbidden"); } break; } if (method=="REGISTER") { # uncomment this if you want to authenticate REGISTER request # if (!www_authenticate("", "credentials")) { # www_challenge("", "0"); # break; # }; save("location"); # dump stored messages - route it through myself (otherwise routed via DNS!) if (m_dump("sip:127.0.0.1")) { xlog("L_ERR", "MSILO: offline messages for %fu dumped\n"); } break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # 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(); }; } failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { break; }; log(1, "MSILO: MESSAGE forward failed - storing it\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("0", "")) { t_reply("202", "Accepted"); } else { log(1, "MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; } _________________________________________________________________ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+world&mkt=en-US&form=QBRE

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[Serusers] SER and rtpproxy stability
by Steven Wang 27 Nov '07

27 Nov '07

Hello We use SER and rtpproxy for a couple of years over public network. It worked perfectly before. The users from different LNA can talk to each other over SIP call (xlite or Linksys adapters). We could fax over SIP from other countries. However, fax stopped working shortly (about 6-month). But SIP call worked fine till the beginning of this year. Suddenly, sip-phone only hears ring. There is no sound on both sides. Do anyone have the similar problem or encounter stability issues in this setup (SER + Rtpproxy)? Best Steven

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Re: [Serusers] [Sems] How to setup conference?
by roayu 27 Nov '07

27 Nov '07

Hi, >From the documentation (mini-Howto on how to configure SER and SEMS to work together), the SER that it's mentioned, does it means the SIP-stack SER or the Proxy SER? I'm having 2 SERs in one machine and I'm a bit confuse... Hope someone can give me a hint =) Thanks. Roa Yu -----Original Message----- From: sems-bounces(a)lists.iptel.org [mailto:sems-bounces@lists.iptel.org] On Behalf Of roayu Sent: Monday, November 26, 2007 10:40 PM To: 'sems(a)lists.iptel.org' Subject: [Sems] How to setup conference? Hi, I'm new to SEMS and i managed to setup 2 SERs and 1 SEMS to work together. However, i'm not sure how to configure the Conference. Can anyone guide me on the conference setup? Thanks. Roa Yu _______________________________________________ Sems mailing list Sems(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/sems

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Re: [Serusers] Can SER support MySQL 5.0.22
by roayu 27 Nov '07

27 Nov '07

Hi, I managed to create the mysql.so after re-intall the MySQL, especially the mysql-dev package. Thanks for your help! Roa Yu -----Original Message----- From: SIP [mailto:sip@arcdiv.com] Sent: Monday, November 26, 2007 8:36 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Can SER support MySQL 5.0.22 First off, do you have the mysql-dev package installed? As for whether or not it will run with mysql 5.0.22, it can, but it's tricky. The makefile for the mysql module doesn't look everywhere for the libmysqlclient libs when it's compiling. It expects the libs to be in a particular place, and if they're not there, it ignores the libpath, compiles ANYway, and they just fails to work once you start up SER. It gives NO errors on compile, which makes it utterly difficult to track down. The latest RPMs for RHEL/CentOS from the MySQL community site place the libmysqlclient.so.X libs into /usr/lib and NOT /usr/lib/mysql as SER expects. SER will simply NOT find it in that location. What we've ended up having to do is symlinking the /usr/lib/libmysqlclient* into /usr/lib/mysql, recompile the mysql module, and install it in the correct place. Then, SER runs happily and readily without weird errors. Also note that there is a patch that must be applied to use MySQL 5.0.X for SER 0.9.6 see http://www.iptel.org/faq/why_does_my_mysql_5_0_x_connection_time_out_when_us ing_ser_0_9_x N. roayu wrote: > > Hi there! > > > > Would like to ask, can SER 0.9.6 support MySQL 5.0.22 ? I got the > following errors when I tried start SER and to connect to MySQL: > > > > [root@localhost ~]# /usr/local/sbin/ser > > 0(3291) ERROR: load_module: could not open module > </usr/local/lib/ser/modules/mysql.so>: libmysqlclient.so.14: cannot > open shared object file: No such file or directory > > 0(3291) parse error (31,13-14): failed to load module > > ERROR: bad config file (1 errors) > > > > > > Thanks. > > > > Roa Yu > > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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[OpenSER-Users] OpenSER Sending INVITE twice
by Douglas Garstang 27 Nov '07

27 Nov '07

I have an OpenSER 1.1 install here, and for some reason, OpenSER is sending the INVITE message twice to the upstream host. The time difference between them is about 1/5 of a second. Nothing is received between the first and second INVITE's. Why is OpenSER doing this? I have xlog() statements everywhere in openser.cfg, and OpenSER is only logging ONE outgoing INVITE message, eventhough it's sending two. It is logging the multiple TRYING messages that come back however. OpenSER is not calling failure_route because nothing is logged in there. What could be going wrong? Why is it doing this? Doug. ____________________________________________________________________________________ Get easy, one-click access to your favorites. Make Yahoo! your homepage. http://www.yahoo.com/r/hs

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[Serusers] Invoking force_rtp_proxy() after fix_nated_sdp() error
by Tomasz Zieleniewski 26 Nov '07

26 Nov '07

Hi, Invoking force_rtp_proxy() after fix_nated_sdp() causes error that in the connection filed one ip address apears after second (previous one is not removed) v=0. o=- 3 2 IN IP4 192.168.1.3. s=CounterPath X-Lite 3.0. c=IN IP4 82.210.160.59192.168.129.74. t=0 0. m=audio 35036 RTP/AVP 107 119 100 106 0 105 98 8 101. a=fmtp:101 0-15. a=rtpmap:107 BV32/16000. a=rtpmap:119 BV32-FEC/16000. a=rtpmap:100 SPEEX/16000. a=rtpmap:106 SPEEX-FEC/16000. a=rtpmap:105 SPEEX-FEC/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. a=direction:active. a=oldmediaip:192.168.1.3. a=nortpproxy:yes. in the example above fix_nated_sdp put the 82.210.160.59 address and force_rtp_proxy instead of replacing the 82.210.160.59 appended it at the end of the line Cheers TOmasz

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[OpenSER-Users] Opensedbctl Error
by TCB 26 Nov '07

26 Nov '07

Hello all, Just checkout the lates svn version of OSer. trying to run openserdbctl create with DB=PGSQL i get the following errors. ERROR: relation "subscriber" does not exist ERROR: relation "dbaliases" does not exist ERROR: Grant privileges to database failed! Any pointers welcome. OSer details: Repository UUID: 689a6050-402a-0410-94f2-e92a70836424 Revision: 3203 -- TC

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[OpenSER-Users] Openser-tls - Registration Error: 503
by Marcio Ribeiro 26 Nov '07

26 Nov '07

Hi, i'm using openser-tls 1.0.1, and i intend to estabilish a security call between 2 users agents. When using eyeBeam 1.5.6 in the automatic mode i can register and make calls, no problem. But when i choose TLS in the softphone i always get Registration Error: 503 - Service Unavailable. the eyeBeam log shows: [07-11-24]13:01:01.386 | Info (debug) RESIP:TRANSPORT | "Adding message to tx buffer to: [ V4 xxx.xx.xx.xxx:5061 TLS target domain=xxx.xx.xx.xxxreceived on: Transport: [ V4 0.0.0.0:35085 TLS target domain=unspecified connectionId=0 ] connectionId=0 ]" | [07-11-24]13:01:01.487 | Info (debug) RESIP:TRANSPORT | "Processing write for [ V4 xxx.xx.xx.xxx:5061 TLS target domain=xxx.xx.xx.xxx received on: Transport: [ V4 0.0.0.0:35085 TLS target domain=unspecified connectionId=0 ] connectionId=0 ]" | [07-11-24]13:01:01.487 | Info (debug) RESIP:TRANSPORT | "Creating fd=1660 V4/TCP" | [07-11-24]13:01:01.487 | Info (debug) RESIP:TRANSPORT | "Opening new connection to [ V4 xxx.xx.xx.xxx:5061 TLS target domain=xxx.xx.xx.xxxreceived on: Transport: [ V4 0.0.0.0:35085 TLS target domain=unspecified connectionId=0 ] connectionId=0 ]" | [07-11-24]13:01:01.488 | Info (debug) RESIP:TRANSPORT | "ConnectionBase::ConnectionBase, who: [ V4 xxx.xx.xx.xxx:5061 TLS target domain=xxx.xx.xx.xxx received on: Transport: [ V4 0.0.0.0:35085 TLS target domain=unspecified connectionId=0 ] connectionId=0 ] 024AA840" | [07-11-24]13:01:01.488 | Info (more) RESIP:TRANSPORT | "Creating TLS connection for domain [ V4 xxx.xx.xx.xxx:5061 TLS target domain= xxx.xx.xx.xxx received on: Transport: [ V4 0.0.0.0:35085 TLS target domain=unspecified connectionId=0 ] connectionId=0 ] on 1660" | [07-11-24]13:01:01.488 | Info (debug) RESIP:TRANSPORT | "Trying to form TLS connection - acting as client" | [07-11-24]13:01:02.405 | Info (more) RESIP:TRANSPORT | "Exception writing to socket 1660 code: 10061; closing connection" | [07-11-24]13:01:02.406 | Info (debug) RESIP:TRANSPORT | "ConnectionBase::~ConnectionBase 024AA840" | [07-11-24]13:01:02.406 | Info (more) RESIP:TRANSACTION | "Sending ConnectionTerminated 4 to TUs" | [07-11-24]13:01:02.406 | Info (more) RESIP:TRANSACTION | "Try sending request to a different dns result" | [07-11-24]13:01:02.406 | Info (more) RESIP:TRANSACTION | "Ran out of dns entries for xxx.xx.xx.xxx. Send 503" | [07-11-24]13:01:02.406 | Info (debug) RESIP | "Helper::makeResponse(SipReq: REGISTER xxx.xx.xx.xxx tid=a1596d01462f2d03 cseq=REGISTER contact= marcio@yyy.yy.yy.yy:35085 / 1 from(tu) code=503 reason=" | [07-11-24]13:01:02.408 | Warning (min) RESIP:DNS | "local hostname does not contain a domain part" | [07-11-24]13:01:02.408 | Info (debug) RESIP:TRANSACTION | "Send to TU: TU: DialogUsageManager size=1 SIP/2.0 503 Service Unavailable Via: SIP/2.0/TLS yyy.yy.yy.yy :35085;branch=z9hG4bK-d87543-a1596d01462f2d03-1--d87543-;rport To: ""Marcio""<sip:marcio@xxx.xx.xx.xxx>;tag=b76ee406 From: ""Marcio""<sip:marcio@xxx.xx.xx.xxx>;tag=bf4bff41 Call-ID: 507d385424726b5fOGFmYzRkOTMwZjFmOTNjNzkyYmE1MjI4MDY0YzAwNzg. CSeq: 1 REGISTER Warning: 499 DUMNONIA """" Content-Length: 0 searching on google doesn't give-me much stuf about this. I really need to implement secure communication... the softphone doesn't matter. I have no idea of what to do, and i really need help... thanks, Marcio Ribeiro

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[OpenSER-Users] error: use_media_proxy(): out of memory
by Tomasz Zieleniewski 26 Nov '07

26 Nov '07

Hi, I am using media proxy for nat handling when I invoke use_media_proxy() for some calls I get the following error: error: use_media_proxy(): out of memory please point me what is wrong?? (I know that some problem with memory:) but why?) Cheers Tomasz

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Re: [Serusers] error: use_media_proxy(): out of memory
by Jiri Kuthan 26 Nov '07

26 Nov '07

memory leak? nathelper may provide you with similar functionality and there is a good chance you will not see this problem. -jiri At 13:20 26/11/2007, Tomasz Zieleniewski wrote: >Hi, > >I am using media proxy for nat handling when I invoke >use_media_proxy() for some calls from my >pstn gateway I get the following error: >error: use_media_proxy(): out of memory >please point me what is wrong?? >(I know that some problem with memory:) but why?) > >cheers >tomasz >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[Serusers] Can SER support MySQL 5.0.22
by roayu 26 Nov '07

26 Nov '07

Hi there! Would like to ask, can SER 0.9.6 support MySQL 5.0.22 ? I got the following errors when I tried start SER and to connect to MySQL: [root@localhost ~]# /usr/local/sbin/ser 0(3291) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: libmysqlclient.so.14: cannot open shared object file: No such file or directory 0(3291) parse error (31,13-14): failed to load module ERROR: bad config file (1 errors) Thanks. Roa Yu

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