[OpenSER-Users] loose_route not setting $du ??

Daniel-Constantin Mierla daniel at voice-system.ro
Fri Dec 7 09:04:15 CET 2007


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 2825F5A8-334C-7B59-8EDA-4C0602180528 at 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 at 10.3.1.115 - no reply to our critical packet.
> Really destroying SIP dialog '2825F5A8-334C-7B59-8EDA-4C0602180528 at 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 at 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 at 10.3.1.31:5061;transport=udp F=sip:pbaker2 at 10.3.1.31 T=sip:500 at 10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833 at 10.3.1.115 
> Dec  7 04:04:52 phonesys-slave openser[24597]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500 at 167.206.216.136 F=sip:pbaker2 at 10.3.1.31 T=sip:500 at 10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833 at 10.3.1.115 
> Dec  7 04:04:53 phonesys-slave openser[24602]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500 at 167.206.216.136 F=sip:pbaker2 at 10.3.1.31 T=sip:500 at 10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833 at 10.3.1.115 
> Dec  7 04:04:54 phonesys-slave openser[24597]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500 at 167.206.216.136 F=sip:pbaker2 at 10.3.1.31 T=sip:500 at 10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833 at 10.3.1.115 
> Dec  7 04:04:56 phonesys-slave openser[24602]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500 at 167.206.216.136 F=sip:pbaker2 at 10.3.1.31 T=sip:500 at 10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833 at 10.3.1.115 
> Dec  7 04:05:00 phonesys-slave openser[24597]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500 at 167.206.216.136 F=sip:pbaker2 at 10.3.1.31 T=sip:500 at 10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833 at 10.3.1.115 
> Dec  7 04:05:04 phonesys-slave openser[24602]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500 at 167.206.216.136 F=sip:pbaker2 at 10.3.1.31 T=sip:500 at 10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833 at 10.3.1.115 
> Dec  7 04:05:08 phonesys-slave openser[24597]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500 at 167.206.216.136 F=sip:pbaker2 at 10.3.1.31 T=sip:500 at 10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833 at 10.3.1.115 
> Dec  7 04:05:12 phonesys-slave openser[24589]: Request leaving server, D-URI='<null>' - M=BYE RURI=sip:500 at 167.206.216.136 F=sip:pbaker2 at 10.3.1.31 T=sip:500 at 10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833 at 10.3.1.115 
> Dec  7 04:05:13 phonesys-slave openser[24599]: Request leaving server, D-URI='<null>' - M=BYE RURI=sip:500 at 167.206.216.136 F=sip:pbaker2 at 10.3.1.31 T=sip:500 at 10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833 at 10.3.1.115 
> Dec  7 04:05:16 phonesys-slave openser[24597]: Request leaving server, D-URI='<null>' - M=BYE RURI=sip:500 at 167.206.216.136 F=sip:pbaker2 at 10.3.1.31 T=sip:500 at 10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833 at 10.3.1.115 
>
>
>
> ##asterisk sip debug##
> <--- SIP read from 10.3.1.31:5060 --->
> INVITE sip:500 at 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 at 10.3.1.31>;tag=1516093159
> To: <sip:500 at 10.3.1.31>
> Contact: <sip:pbaker2 at 10.3.1.115:5060>
> Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528 at 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 at 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 at 10.3.1.31>;tag=1516093159
> To: <sip:500 at 10.3.1.31>
> Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528 at 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 at 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 at 10.3.1.31>;tag=1516093159
> To: <sip:500 at 10.3.1.31>;tag=as70a2356d
> Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528 at 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 at 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 at 10.3.1.31>;tag=1516093159
> To: <sip:500 at 10.3.1.31>;tag=as70a2356d
> Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528 at 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 at 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 at 10.3.1.31>;tag=1516093159
> To: <sip:500 at 10.3.1.31>;tag=as70a2356d
> Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528 at 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 at 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 at 10.3.1.31>;tag=1516093159
> To: <sip:500 at 10.3.1.31>;tag=as70a2356d
> Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528 at 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 at 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 at 10.3.1.31>;tag=1516093159
> To: <sip:500 at 10.3.1.31>;tag=as70a2356d
> Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528 at 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 at 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 at 10.3.1.31>;tag=1516093159
> To: <sip:500 at 10.3.1.31>;tag=as70a2356d
> Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528 at 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 at 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 at 10.3.1.31>;tag=1516093159
> To: <sip:500 at 10.3.1.31>;tag=as70a2356d
> Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528 at 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 at 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 at 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 at 10.3.1.115 - no reply to our critical packet.
> Really destroying SIP dialog '2825F5A8-334C-7B59-8EDA-4C0602180528 at 10.3.1.115' Method: INVITE
>
> _______________________________________________
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