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 [EMAIL PROTECTED] for seqno 8992 (Critical Response) [Dec 7 04:45:55] WARNING[25642]: chan_sip.c:2361 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. Really destroying SIP dialog '[EMAIL PROTECTED]' 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 ([EMAIL PROTECTED]) ######################################################################## ######################################################################## # 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:[EMAIL PROTECTED]/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:[EMAIL PROTECTED]/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:[EMAIL PROTECTED]/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:[EMAIL PROTECTED]/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:[EMAIL PROTECTED]/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:[EMAIL PROTECTED]:5061;transport=udp F=sip:[EMAIL PROTECTED] T=sip:[EMAIL PROTECTED] IP=10.3.1.115 [EMAIL PROTECTED] Dec 7 04:04:52 phonesys-slave openser[24597]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:[EMAIL PROTECTED] F=sip:[EMAIL PROTECTED] T=sip:[EMAIL PROTECTED] IP=10.3.1.115 [EMAIL PROTECTED] Dec 7 04:04:53 phonesys-slave openser[24602]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:[EMAIL PROTECTED] F=sip:[EMAIL PROTECTED] T=sip:[EMAIL PROTECTED] IP=10.3.1.115 [EMAIL PROTECTED] Dec 7 04:04:54 phonesys-slave openser[24597]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:[EMAIL PROTECTED] F=sip:[EMAIL PROTECTED] T=sip:[EMAIL PROTECTED] IP=10.3.1.115 [EMAIL PROTECTED] Dec 7 04:04:56 phonesys-slave openser[24602]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:[EMAIL PROTECTED] F=sip:[EMAIL PROTECTED] T=sip:[EMAIL PROTECTED] IP=10.3.1.115 [EMAIL PROTECTED] Dec 7 04:05:00 phonesys-slave openser[24597]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:[EMAIL PROTECTED] F=sip:[EMAIL PROTECTED] T=sip:[EMAIL PROTECTED] IP=10.3.1.115 [EMAIL PROTECTED] Dec 7 04:05:04 phonesys-slave openser[24602]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:[EMAIL PROTECTED] F=sip:[EMAIL PROTECTED] T=sip:[EMAIL PROTECTED] IP=10.3.1.115 [EMAIL PROTECTED] Dec 7 04:05:08 phonesys-slave openser[24597]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:[EMAIL PROTECTED] F=sip:[EMAIL PROTECTED] T=sip:[EMAIL PROTECTED] IP=10.3.1.115 [EMAIL PROTECTED] Dec 7 04:05:12 phonesys-slave openser[24589]: Request leaving server, D-URI='<null>' - M=BYE RURI=sip:[EMAIL PROTECTED] F=sip:[EMAIL PROTECTED] T=sip:[EMAIL PROTECTED] IP=10.3.1.115 [EMAIL PROTECTED] Dec 7 04:05:13 phonesys-slave openser[24599]: Request leaving server, D-URI='<null>' - M=BYE RURI=sip:[EMAIL PROTECTED] F=sip:[EMAIL PROTECTED] T=sip:[EMAIL PROTECTED] IP=10.3.1.115 [EMAIL PROTECTED] Dec 7 04:05:16 phonesys-slave openser[24597]: Request leaving server, D-URI='<null>' - M=BYE RURI=sip:[EMAIL PROTECTED] F=sip:[EMAIL PROTECTED] T=sip:[EMAIL PROTECTED] IP=10.3.1.115 [EMAIL PROTECTED] ##asterisk sip debug## <--- SIP read from 10.3.1.31:5060 ---> INVITE sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=1516093159 To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]:5060> Call-ID: [EMAIL PROTECTED] 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 - [EMAIL PROTECTED] 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:[EMAIL PROTECTED]>;tag=1516093159 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]>;tag=1516093159 To: <sip:[EMAIL PROTECTED]>;tag=as70a2356d Call-ID: [EMAIL PROTECTED] CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]>;tag=1516093159 To: <sip:[EMAIL PROTECTED]>;tag=as70a2356d Call-ID: [EMAIL PROTECTED] CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]>;tag=1516093159 To: <sip:[EMAIL PROTECTED]>;tag=as70a2356d Call-ID: [EMAIL PROTECTED] CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]>;tag=1516093159 To: <sip:[EMAIL PROTECTED]>;tag=as70a2356d Call-ID: [EMAIL PROTECTED] CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]>;tag=1516093159 To: <sip:[EMAIL PROTECTED]>;tag=as70a2356d Call-ID: [EMAIL PROTECTED] CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]>;tag=1516093159 To: <sip:[EMAIL PROTECTED]>;tag=as70a2356d Call-ID: [EMAIL PROTECTED] CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]>;tag=1516093159 To: <sip:[EMAIL PROTECTED]>;tag=as70a2356d Call-ID: [EMAIL PROTECTED] CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[EMAIL PROTECTED]> 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 [EMAIL PROTECTED] for seqno 8992 (Critical Response) [Dec 7 04:45:55] WARNING[25642]: chan_sip.c:2361 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. Really destroying SIP dialog '[EMAIL PROTECTED]' Method: INVITE _______________________________________________ Users mailing list [email protected] http://lists.openser.org/cgi-bin/mailman/listinfo/users
