Hello,

the sip trace from the proxy will help more. Asterisk is doing 
retransmission to 200ok, no relation with $du, to give more hints, we 
should see what does the proxy with the 200ok.

Cheers,
Daniel


On 12/07/07 06:56, Patrick Baker wrote:
> 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
>
>   

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