6 Dec
2007
6 Dec
'07
11:56 p.m.
I'm having an issue with messages flowing between asterisk and openser and I believe
the issue is related to loose_route and having nulls for $du $dd $ds. Does anyone know
how to resolve this with my config? I have also referenced my asterisk config along with
some debug information. Thanks in advance for anyone that can help! Asterisk ends up
dropping the call after 20 seconds as it appears that openSer isnt responding to asterisk
OK message. Also BYE's and others arent working as well..
[Dec 7 04:45:55] WARNING[25642]: chan_sip.c:2334 retrans_pkt: Maximum retries exceeded on
transmission 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 for seqno 8992 (Critical
Response)
[Dec 7 04:45:55] WARNING[25642]: chan_sip.c:2361 retrans_pkt: Hanging up call
2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 - no reply to our critical packet.
Really destroying SIP dialog '2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115'
Method: INVITE
########################################################################
# This configuration is autogenerated by sip:wizard
# (http://www.sipwise.com/wizard) on Fri Dec 07 05:52:34 +0100 2007
# for OpenSER 1.2
#
# Copyright (C) 2007 Sipwise (support(a)sipwise.com)
########################################################################
########################################################################
# By obtaining, using, and/or copying this configuration and/or its
# associated documentation, you agree that you have read, understood,
# and will comply with the Terms of Usage provided at
# http://www.sipwise.com/news/?page_id=6 as well as the following
# additions:
#
# Permission to use, copy, modify, and distribute this configuration and
# its associated documentation for any purpose and without fee is hereby
# granted, provided that the above copyright notice appears in all
# copies, and that both that copyright notice and this permission notice
# appear in supporting documentation, and that the name of Sipwise or
# the author will not be used in advertising or publicity pertaining to
# distribution of the configuration without specific, written prior
# permission.
########################################################################
########################################################################
# Before using this configuration, read the following prerequisites in
# order to gain the designated functionallity:
#
# base:
# You have to insert all locally served domains (i.e.
# "openserctl domain add your.domain.com").
#
# nat-rtpproxy:
# You have to install RTPProxy
# (http://www.openser.org/downloads/snapshots/rtpproxy/) for relaying
# RTP traffic.
#
# offnet-pstn:
# You have to add a routing entry for lcr (i.e. "openserctl lcr
# addroute '' '' 1 1"). Additionally, you have to add your
gateways
# (i.e. "openserctl lcr addgw my-test-gw 1.2.3.4 5060 sip udp 1").
#
########################################################################
########################################################################
# Configuration 'sip:wizard - Fri Dec 07 05:52:34 +0100 2007'
########################################################################
listen = udp:127.0.0.1:5060
listen = udp:10.3.1.31:5060
mpath = "/usr/local/lib/openser/modules"
children = 8
debug = 3
fork = yes
group = "openser"
user = "openser"
disable_tcp = no
log_facility = LOG_DAEMON
log_stderror = no
tcp_children = 4
mhomed = no
server_signature = yes
sock_group = "openser"
sock_mode = 0600
sock_user = "openser"
unix_sock = "/tmp/openser.sock"
unix_sock_children = 1
reply_to_via = no
sip_warning = no
check_via = no
dns = no
rev_dns = no
disable_core_dump = no
dns_try_ipv6 = yes
dns_use_search_list = yes
loadmodule "usrloc.so"
modparam("usrloc", "user_column", "username")
modparam("usrloc", "domain_column", "domain")
modparam("usrloc", "contact_column", "contact")
modparam("usrloc", "expires_column", "expires")
modparam("usrloc", "q_column", "q")
modparam("usrloc", "callid_column", "callid")
modparam("usrloc", "cseq_column", "cseq")
modparam("usrloc", "methods_column", "methods")
modparam("usrloc", "flags_column", "flags")
modparam("usrloc", "user_agent_column", "user_agent")
modparam("usrloc", "received_column", "received")
modparam("usrloc", "socket_column", "socket")
modparam("usrloc", "use_domain", 0)
modparam("usrloc", "desc_time_order", 0)
modparam("usrloc", "timer_interval", 60)
modparam("usrloc", "db_url",
"mysql://openser:openserrw@localhost/openser")
modparam("usrloc", "db_mode", 1)
modparam("usrloc", "matching_mode", 0)
modparam("usrloc", "cseq_delay", 20)
modparam("usrloc", "nat_bflag", 6)
loadmodule "textops.so"
loadmodule "rr.so"
modparam("rr", "enable_full_lr", 0)
modparam("rr", "append_fromtag", 1)
modparam("rr", "enable_double_rr", 1)
modparam("rr", "add_username", 0)
loadmodule "tm.so"
modparam("tm", "fr_timer", 30)
modparam("tm", "fr_inv_timer", 120)
modparam("tm", "wt_timer", 5)
modparam("tm", "delete_timer", 2)
modparam("tm", "noisy_ctimer", 0)
modparam("tm", "ruri_matching", 1)
modparam("tm", "via1_matching", 1)
modparam("tm", "unix_tx_timeout", 2)
modparam("tm", "restart_fr_on_each_reply", 1)
modparam("tm", "pass_provisional_replies", 0)
loadmodule "xlog.so"
modparam("xlog", "buf_size", 4096)
modparam("xlog", "force_color", 0)
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")
modparam("mi_fifo", "fifo_mode", 0660)
modparam("mi_fifo", "fifo_group", "openser")
modparam("mi_fifo", "fifo_user", "openser")
modparam("mi_fifo", "reply_dir", "/tmp/")
modparam("mi_fifo", "reply_indent", "\t")
loadmodule "domain.so"
modparam("domain", "db_url",
"mysql://openser:openserrw@localhost/openser")
modparam("domain", "db_mode", 1)
modparam("domain", "domain_table", "domain")
modparam("domain", "domain_col", "domain")
loadmodule "nathelper.so"
modparam("nathelper", "natping_interval", 60)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "rtpproxy_sock",
"unix:/var/run/rtpproxy.sock")
modparam("nathelper", "rtpproxy_disable", 0)
modparam("nathelper", "rtpproxy_disable_tout", 60)
modparam("nathelper", "rtpproxy_tout", 1)
modparam("nathelper", "rtpproxy_retr", 5)
modparam("nathelper", "sipping_method", "OPTIONS")
modparam("nathelper", "received_avp", "$avp(i:801)")
loadmodule "sl.so"
modparam("sl", "enable_stats", 1)
loadmodule "uri.so"
loadmodule "registrar.so"
modparam("registrar", "default_expires", 3600)
modparam("registrar", "min_expires", 60)
modparam("registrar", "max_expires", 0)
modparam("registrar", "default_q", 0)
modparam("registrar", "append_branches", 1)
modparam("registrar", "case_sensitive", 0)
modparam("registrar", "received_param", "received")
modparam("registrar", "max_contacts", 0)
modparam("registrar", "retry_after", 0)
modparam("registrar", "method_filtering", 0)
modparam("registrar", "path_mode", 2)
modparam("registrar", "path_use_received", 0)
modparam("registrar", "received_avp", "$avp(i:801)")
loadmodule "maxfwd.so"
modparam("maxfwd", "max_limit", 256)
loadmodule "mysql.so"
modparam("mysql", "ping_interval", 300)
modparam("mysql", "auto_reconnect", 1)
loadmodule "auth.so"
modparam("auth", "nonce_expire", 300)
modparam("auth", "rpid_suffix",
";party=calling;id-type=subscriber;screen=yes")
modparam("auth", "rpid_avp", "$avp(s:rpid)")
loadmodule "auth_db.so"
modparam("auth_db", "db_url",
"mysql://openser:openserrw@localhost/openser")
modparam("auth_db", "user_column", "username")
modparam("auth_db", "domain_column", "domain")
modparam("auth_db", "password_column", "password")
modparam("auth_db", "password_column_2", "ha1b")
modparam("auth_db", "calculate_ha1", 1)
modparam("auth_db", "use_domain", 0)
modparam("auth_db", "load_credentials", "rpid")
loadmodule "uri_db.so"
modparam("uri_db", "db_url",
"mysql://openser:openserrw@localhost/openser")
modparam("uri_db", "uri_table", "uri")
modparam("uri_db", "uri_user_column", "username")
modparam("uri_db", "uri_domain_column", "domain")
modparam("uri_db", "uri_uriuser_column", "uri_user")
modparam("uri_db", "subscriber_table", "subscriber")
modparam("uri_db", "subscriber_user_column", "username")
modparam("uri_db", "subscriber_domain_column", "domain")
modparam("uri_db", "use_uri_table", 0)
modparam("uri_db", "use_domain", 0)
loadmodule "lcr.so"
modparam("lcr", "db_url",
"mysql://openser:openserrw@localhost/openser")
modparam("lcr", "gw_table", "gw")
modparam("lcr", "gw_name_column", "gw_name")
modparam("lcr", "ip_addr_column", "ip_addr")
modparam("lcr", "port_column", "port")
modparam("lcr", "uri_scheme_column", "uri_scheme")
modparam("lcr", "transport_column", "transport")
modparam("lcr", "grp_id_column", "grp_id")
modparam("lcr", "lcr_table", "lcr")
modparam("lcr", "strip_column", "strip")
modparam("lcr", "prefix_column", "prefix")
modparam("lcr", "from_uri_column", "from_uri")
modparam("lcr", "priority_column", "priority")
modparam("lcr", "gw_uri_avp", "1400")
modparam("lcr", "ruri_user_avp", "1402")
modparam("lcr", "contact_avp", "1401")
modparam("lcr", "fr_inv_timer_avp", "s:fr_inv_timer_avp")
modparam("lcr", "fr_inv_timer", 90)
modparam("lcr", "fr_inv_timer_next", 30)
modparam("lcr", "rpid_avp", "s:rpid")
########################################################################
# Request route 'main'
########################################################################
route[0]
{
xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci\n");
force_rport();
if(msg:len > max_len)
{
xlog("L_INFO", "Message too big - M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci\n");
sl_send_reply("513", "Message Too Big");
exit;
}
if (!mf_process_maxfwd_header("10"))
{
xlog("L_INFO", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci\n");
sl_send_reply("483", "Too Many Hops");
exit;
}
if(!is_method("REGISTER"))
{
if(nat_uac_test("19"))
{
record_route(";nat=yes");
}
else
{
record_route();
}
}
if(is_method("CANCEL") || is_method("BYE"))
{
unforce_rtp_proxy();
}
if(loose_route())
{
if(!has_totag())
{
xlog("L_INFO", "Initial loose-routing rejected - M=$rm RURI=$ru F=$fu
T=$tu IP=$si ID=$ci\n");
sl_send_reply("403", "Initial Loose-Routing Rejected");
exit;
}
if(nat_uac_test("19") || search("^Route:.*;nat=yes"))
{
fix_nated_contact();
setbflag(6);
}
route(3);
}
if(is_method("REGISTER"))
{
route(2);
}
if(is_method("INVITE"))
{
route(4);
}
if(is_method("CANCEL") || is_method("ACK"))
{
route(8);
}
route(9);
}
########################################################################
# Request route 'stop-rtp-proxy'
########################################################################
route[1]
{
if(isflagset(22))
{
unforce_rtp_proxy();
}
}
########################################################################
# Request route 'base-route-register'
########################################################################
route[2]
{
sl_send_reply("100", "Trying");
if(!www_authorize("", "subscriber"))
{
xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu
T=$tu IP=$si ID=$ci\n");
www_challenge("", "0");
exit;
}
if(!check_to())
{
xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
sl_send_reply("403", "Spoofed To-URI Detected");
exit;
}
consume_credentials();
if(!search("^Contact:[ ]*\*") && nat_uac_test("19"))
{
fix_nated_register();
setbflag(6);
}
if(!save("location"))
{
xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci\n");
sl_reply_error();
exit;
}
xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
exit;
}
########################################################################
# Request route 'base-outbound'
########################################################################
route[3]
{
if(isbflagset(6))
{
if(!isflagset(22) && !search("^Content-Length:[ ]*0"))
{
setflag(22);
force_rtp_proxy();
}
t_on_reply("2");
}
else
{
t_on_reply("1");
}
if(!isflagset(21))
{
t_on_failure("2");
}
if(isflagset(29))
{
append_branch();
}
if(is_present_hf("Proxy-Authorization"))
{
consume_credentials();
}
xlog("L_INFO", "Request leaving server, D-URI='$du' - M=$rm
RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
# no 100 (we already sent it) and no DNS blacklisting
if(!t_relay("0x05"))
{
sl_reply_error();
if(is_method("INVITE") && isbflagset(6))
{
unforce_rtp_proxy();
}
}
exit;
}
########################################################################
# Request route 'base-route-invite'
########################################################################
route[4]
{
sl_send_reply("100", "Trying");
if(from_gw())
{
xlog("L_INFO", "Call from PSTN' - M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci\n");
setflag(23);
}
else
{
if(!proxy_authorize("", "subscriber"))
{
xlog("L_INFO", "Proxy authentication failed - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
proxy_challenge("", "0");
exit;
}
if(!check_from())
{
xlog("L_INFO", "Spoofed From-URI detected - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
sl_send_reply("403", "Spoofed From-URI Detected");
exit;
}
}
if(nat_uac_test("19"))
{
fix_nated_contact();
setbflag(6);
}
route(5);
}
########################################################################
# Request route 'invite-find-callee'
########################################################################
route[5]
{
if(!is_domain_local("$rd"))
{
setflag(20);
route(7);
}
if(does_uri_exist())
{
xlog("L_INFO", "Callee is local - M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci\n");
route(6);
}
else
{
xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci\n");
route(7);
}
exit;
}
########################################################################
# Request route 'invite-to-internal'
########################################################################
route[6]
{
if(!lookup("location"))
{
xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci\n");
sl_send_reply("404", "User Offline");
}
else
{
xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci\n");
route(3);
}
exit;
}
########################################################################
# Request route 'invite-to-external'
########################################################################
route[7]
{
if(isflagset(20))
{
xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
route(3);
exit;
}
if(!isflagset(23))
{
# don't allow calls relaying from PSTN to PSTN, if not explicitely forwarded
if(uri =~ "^sip:[0-9]+@")
{
# only route numeric users to PSTN
if(!load_gws())
{
xlog("L_ERR", "Error loading PSTN gateways - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
sl_send_reply("503", "PSTN Termination Currently Unavailable");
exit;
}
if(!next_gw())
{
xlog("L_ERR", "No PSTN gateways available - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
sl_send_reply("503", "PSTN Termination Currently Unavailable");
exit;
}
setflag(21);
t_on_failure("1");
route(3);
}
}
xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci\n");
sl_send_reply("404", "User Not Found");
exit;
}
########################################################################
# Request route 'base-route-local'
########################################################################
route[8]
{
t_on_reply("1");
if(t_check_trans())
{
xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
if(!t_relay())
{
sl_reply_error();
}
}
else
{
xlog("L_INFO", "Dropping mis-routed request - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
}
exit;
}
########################################################################
# Request route 'base-route-generic'
########################################################################
route[9]
{
xlog("L_INFO", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci\n");
sl_send_reply("501", "Method Not Supported Here");
exit;
}
########################################################################
# Request route 'base-filter-failover'
########################################################################
route[10]
{
if(!t_check_status("408|500|503"))
{
xlog("L_INFO", "No failover routing needed for this response code - M=$rm
RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
route(1);
exit;
}
}
########################################################################
# Reply route 'base-standard-reply'
########################################################################
onreply_route[1]
{
xlog("L_INFO", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n");
exit;
}
########################################################################
# Reply route 'base-nat-reply'
########################################################################
onreply_route[2]
{
xlog("L_INFO", "NAT-Reply - S=$rs D=$rr F=$fu T=$tu IP=$si
ID=$ci\n");
if(nat_uac_test("1"))
{
fix_nated_contact();
}
if(isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]")
{
if(!search("^Content-Length:[ ]*0"))
{
force_rtp_proxy();
}
}
exit;
}
########################################################################
# Failure route 'pstn-failover'
########################################################################
failure_route[1]
{
xlog("L_INFO", "Failure route for PSTN entered - M=$rm RURI=$ru F=$fu
T=$tu IP=$si ID=$ci\n");
route(10);
if(!next_gw())
{
xlog("L_ERR", "Failed to select next PSTN gateway - M=$rm RURI=$ru F=$fu
T=$tu IP=$si ID=$ci\n");
route(1);
exit;
}
t_on_failure("1");
route(3);
}
########################################################################
# Failure route 'base-standard-failure'
########################################################################
failure_route[2]
{
route(10);
route(1);
}
##asterisk sip.conf##
[general]
matchexterniplocally=yes
canreinvite=no
externip=xxx.206.xxx.136
localnet=10.3.1.0/255.255.255.0
context=default
bindport=5061
bindaddr=0.0.0.0
sipdebug=yes
nat=yes
[openser]
type=friend
context=default
insecure=very
externalnotify=yes
allow=all
##ser log##
Dec 7 04:04:52 phonesys-slave openser[24602]: Request leaving server,
D-URI='<null>' - M=INVITE RURI=sip:500@10.3.1.31:5061;transport=udp
F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115
ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115
Dec 7 04:04:52 phonesys-slave openser[24597]: Request leaving server,
D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31
T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115
Dec 7 04:04:53 phonesys-slave openser[24602]: Request leaving server,
D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31
T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115
Dec 7 04:04:54 phonesys-slave openser[24597]: Request leaving server,
D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31
T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115
Dec 7 04:04:56 phonesys-slave openser[24602]: Request leaving server,
D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31
T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115
Dec 7 04:05:00 phonesys-slave openser[24597]: Request leaving server,
D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31
T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115
Dec 7 04:05:04 phonesys-slave openser[24602]: Request leaving server,
D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31
T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115
Dec 7 04:05:08 phonesys-slave openser[24597]: Request leaving server,
D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31
T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115
Dec 7 04:05:12 phonesys-slave openser[24589]: Request leaving server,
D-URI='<null>' - M=BYE RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31
T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115
Dec 7 04:05:13 phonesys-slave openser[24599]: Request leaving server,
D-URI='<null>' - M=BYE RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31
T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115
Dec 7 04:05:16 phonesys-slave openser[24597]: Request leaving server,
D-URI='<null>' - M=BYE RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31
T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115
##asterisk sip debug##
<--- SIP read from 10.3.1.31:5060 --->
INVITE sip:500@10.3.1.31:5061;transport=udp SIP/2.0
Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes>
Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0
Via: SIP/2.0/UDP
10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53
From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159
To: <sip:500@10.3.1.31>
Contact: <sip:pbaker2@10.3.1.115:5060>
Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115
CSeq: 8992 INVITE
Max-Forwards: 69
Content-Type: application/sdp
User-Agent: X-Lite release 1105d
Content-Length: 252
v=0
o=pbaker2 3001829617 3001829777 IN IP4 10.3.1.115
s=X-Lite
c=IN IP4 10.3.1.31
t=0 0
m=audio 35128 RTP/AVP 3 97 101
a=rtpmap:3 gsm/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=nortpproxy:yes
<------------->
--- (13 headers 12 lines) ---
Sending to 10.3.1.31 : 5060 (no NAT)
Using INVITE request as basis request - 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115
No user 'pbaker2' in SIP users list
Found peer 'openser' for 'pbaker2' from 10.3.1.31:5060
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 10.3.1.31:35128
Found audio description format gsm for ID 3
Found audio description format speex for ID 97
Found audio description format telephone-event for ID 101
Capabilities: us - 0x27f9fff
(g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|t140),
peer - audio=0x202 (gsm|speex)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x202
(gsm|speex)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event),
combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.3.1.31:35128
Looking for 500 in default (domain 10.3.1.31)
list_route: hop: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes>
<--- Transmitting (no NAT) to 10.3.1.31:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31
Via: SIP/2.0/UDP
10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53
Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes>
From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159
To: <sip:500@10.3.1.31>
Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115
CSeq: 8992 INVITE
User-Agent: Asterisk PBX SVN-trunk-r91598
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:500@167.206.216.136>
Content-Length: 0
<------------>
Audio is at 167.206.216.136 port 37712
Adding codec 0x2 (gsm) to SDP
Adding codec 0x200 (speex) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
phonesys-slave*CLI>
<--- Reliably Transmitting (no NAT) to 10.3.1.31:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31
Via: SIP/2.0/UDP
10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53
Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes>
From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159
To: <sip:500@10.3.1.31>;tag=as70a2356d
Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115
CSeq: 8992 INVITE
User-Agent: Asterisk PBX SVN-trunk-r91598
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:500@167.206.216.136>
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 87430933 87430933 IN IP4 167.206.216.136
s=Asterisk PBX SVN-trunk-r91598
c=IN IP4 167.206.216.136
t=0 0
m=audio 37712 RTP/AVP 3 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
Retransmitting #1 (no NAT) to 10.3.1.31:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31
Via: SIP/2.0/UDP
10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53
Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes>
From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159
To: <sip:500@10.3.1.31>;tag=as70a2356d
Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115
CSeq: 8992 INVITE
User-Agent: Asterisk PBX SVN-trunk-r91598
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:500@167.206.216.136>
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 87430933 87430933 IN IP4 167.206.216.136
s=Asterisk PBX SVN-trunk-r91598
c=IN IP4 167.206.216.136
t=0 0
m=audio 37712 RTP/AVP 3 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #2 (no NAT) to 10.3.1.31:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31
Via: SIP/2.0/UDP
10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53
Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes>
From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159
To: <sip:500@10.3.1.31>;tag=as70a2356d
Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115
CSeq: 8992 INVITE
User-Agent: Asterisk PBX SVN-trunk-r91598
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:500@167.206.216.136>
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 87430933 87430933 IN IP4 167.206.216.136
s=Asterisk PBX SVN-trunk-r91598
c=IN IP4 167.206.216.136
t=0 0
m=audio 37712 RTP/AVP 3 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #3 (no NAT) to 10.3.1.31:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31
Via: SIP/2.0/UDP
10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53
Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes>
From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159
To: <sip:500@10.3.1.31>;tag=as70a2356d
Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115
CSeq: 8992 INVITE
User-Agent: Asterisk PBX SVN-trunk-r91598
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:500@167.206.216.136>
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 87430933 87430933 IN IP4 167.206.216.136
s=Asterisk PBX SVN-trunk-r91598
c=IN IP4 167.206.216.136
t=0 0
m=audio 37712 RTP/AVP 3 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #4 (no NAT) to 10.3.1.31:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31
Via: SIP/2.0/UDP
10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53
Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes>
From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159
To: <sip:500@10.3.1.31>;tag=as70a2356d
Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115
CSeq: 8992 INVITE
User-Agent: Asterisk PBX SVN-trunk-r91598
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:500@167.206.216.136>
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 87430933 87430933 IN IP4 167.206.216.136
s=Asterisk PBX SVN-trunk-r91598
c=IN IP4 167.206.216.136
t=0 0
m=audio 37712 RTP/AVP 3 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #5 (no NAT) to 10.3.1.31:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31
Via: SIP/2.0/UDP
10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53
Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes>
From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159
To: <sip:500@10.3.1.31>;tag=as70a2356d
Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115
CSeq: 8992 INVITE
User-Agent: Asterisk PBX SVN-trunk-r91598
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:500@167.206.216.136>
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 87430933 87430933 IN IP4 167.206.216.136
s=Asterisk PBX SVN-trunk-r91598
c=IN IP4 167.206.216.136
t=0 0
m=audio 37712 RTP/AVP 3 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Dec 7 04:45:50] NOTICE[25642]: rtp.c:998 process_rfc3389: Comfort noise support
incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP:
10.3.1.31
Retransmitting #6 (no NAT) to 10.3.1.31:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31
Via: SIP/2.0/UDP
10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53
Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes>
From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159
To: <sip:500@10.3.1.31>;tag=as70a2356d
Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115
CSeq: 8992 INVITE
User-Agent: Asterisk PBX SVN-trunk-r91598
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:500@167.206.216.136>
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 87430933 87430933 IN IP4 167.206.216.136
s=Asterisk PBX SVN-trunk-r91598
c=IN IP4 167.206.216.136
t=0 0
m=audio 37712 RTP/AVP 3 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Dec 7 04:45:55] WARNING[25642]: chan_sip.c:2334 retrans_pkt: Maximum retries exceeded on
transmission 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 for seqno 8992 (Critical
Response)
[Dec 7 04:45:55] WARNING[25642]: chan_sip.c:2361 retrans_pkt: Hanging up call
2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 - no reply to our critical packet.
Really destroying SIP dialog '2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115'
Method: INVITE
7 Dec
7 Dec
3:04 a.m.
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(a)10.3.1.115 for seqno 8992 (Critical
Response)
[Dec 7 04:45:55] WARNING[25642]: chan_sip.c:2361 retrans_pkt: Hanging up call
2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 - no reply to our critical packet.
Really destroying SIP dialog '2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115'
Method: INVITE
########################################################################
# This configuration is autogenerated by sip:wizard
# (http://www.sipwise.com/wizard) on Fri Dec 07 05:52:34 +0100 2007
# for OpenSER 1.2
#
# Copyright (C) 2007 Sipwise (support(a)sipwise.com)
########################################################################
########################################################################
# By obtaining, using, and/or copying this configuration and/or its
# associated documentation, you agree that you have read, understood,
# and will comply with the Terms of Usage provided at
# http://www.sipwise.com/news/?page_id=6 as well as the following
# additions:
#
# Permission to use, copy, modify, and distribute this configuration and
# its associated documentation for any purpose and without fee is hereby
# granted, provided that the above copyright notice appears in all
# copies, and that both that copyright notice and this permission notice
# appear in supporting documentation, and that the name of Sipwise or
# the author will not be used in advertising or publicity pertaining to
# distribution of the configuration without specific, written prior
# permission.
########################################################################
########################################################################
# Before using this configuration, read the following prerequisites in
# order to gain the designated functionallity:
#
# base:
# You have to insert all locally served domains (i.e.
# "openserctl domain add your.domain.com").
#
# nat-rtpproxy:
# You have to install RTPProxy
# (http://www.openser.org/downloads/snapshots/rtpproxy/) for relaying
# RTP traffic.
#
# offnet-pstn:
# You have to add a routing entry for lcr (i.e. "openserctl lcr
# addroute '' '' 1 1"). Additionally, you have to add your
gateways
# (i.e. "openserctl lcr addgw my-test-gw 1.2.3.4 5060 sip udp 1").
#
########################################################################
########################################################################
# Configuration 'sip:wizard - Fri Dec 07 05:52:34 +0100 2007'
########################################################################
listen = udp:127.0.0.1:5060
listen = udp:10.3.1.31:5060
mpath = "/usr/local/lib/openser/modules"
children = 8
debug = 3
fork = yes
group = "openser"
user = "openser"
disable_tcp = no
log_facility = LOG_DAEMON
log_stderror = no
tcp_children = 4
mhomed = no
server_signature = yes
sock_group = "openser"
sock_mode = 0600
sock_user = "openser"
unix_sock = "/tmp/openser.sock"
unix_sock_children = 1
reply_to_via = no
sip_warning = no
check_via = no
dns = no
rev_dns = no
disable_core_dump = no
dns_try_ipv6 = yes
dns_use_search_list = yes
loadmodule "usrloc.so"
modparam("usrloc", "user_column", "username")
modparam("usrloc", "domain_column", "domain")
modparam("usrloc", "contact_column", "contact")
modparam("usrloc", "expires_column", "expires")
modparam("usrloc", "q_column", "q")
modparam("usrloc", "callid_column", "callid")
modparam("usrloc", "cseq_column", "cseq")
modparam("usrloc", "methods_column", "methods")
modparam("usrloc", "flags_column", "flags")
modparam("usrloc", "user_agent_column", "user_agent")
modparam("usrloc", "received_column", "received")
modparam("usrloc", "socket_column", "socket")
modparam("usrloc", "use_domain", 0)
modparam("usrloc", "desc_time_order", 0)
modparam("usrloc", "timer_interval", 60)
modparam("usrloc", "db_url",
"mysql://openser:openserrw@localhost/openser")
modparam("usrloc", "db_mode", 1)
modparam("usrloc", "matching_mode", 0)
modparam("usrloc", "cseq_delay", 20)
modparam("usrloc", "nat_bflag", 6)
loadmodule "textops.so"
loadmodule "rr.so"
modparam("rr", "enable_full_lr", 0)
modparam("rr", "append_fromtag", 1)
modparam("rr", "enable_double_rr", 1)
modparam("rr", "add_username", 0)
loadmodule "tm.so"
modparam("tm", "fr_timer", 30)
modparam("tm", "fr_inv_timer", 120)
modparam("tm", "wt_timer", 5)
modparam("tm", "delete_timer", 2)
modparam("tm", "noisy_ctimer", 0)
modparam("tm", "ruri_matching", 1)
modparam("tm", "via1_matching", 1)
modparam("tm", "unix_tx_timeout", 2)
modparam("tm", "restart_fr_on_each_reply", 1)
modparam("tm", "pass_provisional_replies", 0)
loadmodule "xlog.so"
modparam("xlog", "buf_size", 4096)
modparam("xlog", "force_color", 0)
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")
modparam("mi_fifo", "fifo_mode", 0660)
modparam("mi_fifo", "fifo_group", "openser")
modparam("mi_fifo", "fifo_user", "openser")
modparam("mi_fifo", "reply_dir", "/tmp/")
modparam("mi_fifo", "reply_indent", "\t")
loadmodule "domain.so"
modparam("domain", "db_url",
"mysql://openser:openserrw@localhost/openser")
modparam("domain", "db_mode", 1)
modparam("domain", "domain_table", "domain")
modparam("domain", "domain_col", "domain")
loadmodule "nathelper.so"
modparam("nathelper", "natping_interval", 60)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "rtpproxy_sock",
"unix:/var/run/rtpproxy.sock")
modparam("nathelper", "rtpproxy_disable", 0)
modparam("nathelper", "rtpproxy_disable_tout", 60)
modparam("nathelper", "rtpproxy_tout", 1)
modparam("nathelper", "rtpproxy_retr", 5)
modparam("nathelper", "sipping_method", "OPTIONS")
modparam("nathelper", "received_avp", "$avp(i:801)")
loadmodule "sl.so"
modparam("sl", "enable_stats", 1)
loadmodule "uri.so"
loadmodule "registrar.so"
modparam("registrar", "default_expires", 3600)
modparam("registrar", "min_expires", 60)
modparam("registrar", "max_expires", 0)
modparam("registrar", "default_q", 0)
modparam("registrar", "append_branches", 1)
modparam("registrar", "case_sensitive", 0)
modparam("registrar", "received_param", "received")
modparam("registrar", "max_contacts", 0)
modparam("registrar", "retry_after", 0)
modparam("registrar", "method_filtering", 0)
modparam("registrar", "path_mode", 2)
modparam("registrar", "path_use_received", 0)
modparam("registrar", "received_avp", "$avp(i:801)")
loadmodule "maxfwd.so"
modparam("maxfwd", "max_limit", 256)
loadmodule "mysql.so"
modparam("mysql", "ping_interval", 300)
modparam("mysql", "auto_reconnect", 1)
loadmodule "auth.so"
modparam("auth", "nonce_expire", 300)
modparam("auth", "rpid_suffix",
";party=calling;id-type=subscriber;screen=yes")
modparam("auth", "rpid_avp", "$avp(s:rpid)")
loadmodule "auth_db.so"
modparam("auth_db", "db_url",
"mysql://openser:openserrw@localhost/openser")
modparam("auth_db", "user_column", "username")
modparam("auth_db", "domain_column", "domain")
modparam("auth_db", "password_column", "password")
modparam("auth_db", "password_column_2", "ha1b")
modparam("auth_db", "calculate_ha1", 1)
modparam("auth_db", "use_domain", 0)
modparam("auth_db", "load_credentials", "rpid")
loadmodule "uri_db.so"
modparam("uri_db", "db_url",
"mysql://openser:openserrw@localhost/openser")
modparam("uri_db", "uri_table", "uri")
modparam("uri_db", "uri_user_column", "username")
modparam("uri_db", "uri_domain_column", "domain")
modparam("uri_db", "uri_uriuser_column", "uri_user")
modparam("uri_db", "subscriber_table", "subscriber")
modparam("uri_db", "subscriber_user_column", "username")
modparam("uri_db", "subscriber_domain_column", "domain")
modparam("uri_db", "use_uri_table", 0)
modparam("uri_db", "use_domain", 0)
loadmodule "lcr.so"
modparam("lcr", "db_url",
"mysql://openser:openserrw@localhost/openser")
modparam("lcr", "gw_table", "gw")
modparam("lcr", "gw_name_column", "gw_name")
modparam("lcr", "ip_addr_column", "ip_addr")
modparam("lcr", "port_column", "port")
modparam("lcr", "uri_scheme_column", "uri_scheme")
modparam("lcr", "transport_column", "transport")
modparam("lcr", "grp_id_column", "grp_id")
modparam("lcr", "lcr_table", "lcr")
modparam("lcr", "strip_column", "strip")
modparam("lcr", "prefix_column", "prefix")
modparam("lcr", "from_uri_column", "from_uri")
modparam("lcr", "priority_column", "priority")
modparam("lcr", "gw_uri_avp", "1400")
modparam("lcr", "ruri_user_avp", "1402")
modparam("lcr", "contact_avp", "1401")
modparam("lcr", "fr_inv_timer_avp", "s:fr_inv_timer_avp")
modparam("lcr", "fr_inv_timer", 90)
modparam("lcr", "fr_inv_timer_next", 30)
modparam("lcr", "rpid_avp", "s:rpid")
########################################################################
# Request route 'main'
########################################################################
route[0]
{
xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci\n");
force_rport();
if(msg:len > max_len)
{
xlog("L_INFO", "Message too big - M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci\n");
sl_send_reply("513", "Message Too Big");
exit;
}
if (!mf_process_maxfwd_header("10"))
{
xlog("L_INFO", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci\n");
sl_send_reply("483", "Too Many Hops");
exit;
}
if(!is_method("REGISTER"))
{
if(nat_uac_test("19"))
{
record_route(";nat=yes");
}
else
{
record_route();
}
}
if(is_method("CANCEL") || is_method("BYE"))
{
unforce_rtp_proxy();
}
if(loose_route())
{
if(!has_totag())
{
xlog("L_INFO", "Initial loose-routing rejected - M=$rm RURI=$ru F=$fu
T=$tu IP=$si ID=$ci\n");
sl_send_reply("403", "Initial Loose-Routing Rejected");
exit;
}
if(nat_uac_test("19") || search("^Route:.*;nat=yes"))
{
fix_nated_contact();
setbflag(6);
}
route(3);
}
if(is_method("REGISTER"))
{
route(2);
}
if(is_method("INVITE"))
{
route(4);
}
if(is_method("CANCEL") || is_method("ACK"))
{
route(8);
}
route(9);
}
########################################################################
# Request route 'stop-rtp-proxy'
########################################################################
route[1]
{
if(isflagset(22))
{
unforce_rtp_proxy();
}
}
########################################################################
# Request route 'base-route-register'
########################################################################
route[2]
{
sl_send_reply("100", "Trying");
if(!www_authorize("", "subscriber"))
{
xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu
T=$tu IP=$si ID=$ci\n");
www_challenge("", "0");
exit;
}
if(!check_to())
{
xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
sl_send_reply("403", "Spoofed To-URI Detected");
exit;
}
consume_credentials();
if(!search("^Contact:[ ]*\*") && nat_uac_test("19"))
{
fix_nated_register();
setbflag(6);
}
if(!save("location"))
{
xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci\n");
sl_reply_error();
exit;
}
xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
exit;
}
########################################################################
# Request route 'base-outbound'
########################################################################
route[3]
{
if(isbflagset(6))
{
if(!isflagset(22) && !search("^Content-Length:[ ]*0"))
{
setflag(22);
force_rtp_proxy();
}
t_on_reply("2");
}
else
{
t_on_reply("1");
}
if(!isflagset(21))
{
t_on_failure("2");
}
if(isflagset(29))
{
append_branch();
}
if(is_present_hf("Proxy-Authorization"))
{
consume_credentials();
}
xlog("L_INFO", "Request leaving server, D-URI='$du' - M=$rm
RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
# no 100 (we already sent it) and no DNS blacklisting
if(!t_relay("0x05"))
{
sl_reply_error();
if(is_method("INVITE") && isbflagset(6))
{
unforce_rtp_proxy();
}
}
exit;
}
########################################################################
# Request route 'base-route-invite'
########################################################################
route[4]
{
sl_send_reply("100", "Trying");
if(from_gw())
{
xlog("L_INFO", "Call from PSTN' - M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci\n");
setflag(23);
}
else
{
if(!proxy_authorize("", "subscriber"))
{
xlog("L_INFO", "Proxy authentication failed - M=$rm RURI=$ru F=$fu
T=$tu IP=$si ID=$ci\n");
proxy_challenge("", "0");
exit;
}
if(!check_from())
{
xlog("L_INFO", "Spoofed From-URI detected - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
sl_send_reply("403", "Spoofed From-URI Detected");
exit;
}
}
if(nat_uac_test("19"))
{
fix_nated_contact();
setbflag(6);
}
route(5);
}
########################################################################
# Request route 'invite-find-callee'
########################################################################
route[5]
{
if(!is_domain_local("$rd"))
{
setflag(20);
route(7);
}
if(does_uri_exist())
{
xlog("L_INFO", "Callee is local - M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci\n");
route(6);
}
else
{
xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci\n");
route(7);
}
exit;
}
########################################################################
# Request route 'invite-to-internal'
########################################################################
route[6]
{
if(!lookup("location"))
{
xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci\n");
sl_send_reply("404", "User Offline");
}
else
{
xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci\n");
route(3);
}
exit;
}
########################################################################
# Request route 'invite-to-external'
########################################################################
route[7]
{
if(isflagset(20))
{
xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
route(3);
exit;
}
if(!isflagset(23))
{
# don't allow calls relaying from PSTN to PSTN, if not explicitely forwarded
if(uri =~ "^sip:[0-9]+@")
{
# only route numeric users to PSTN
if(!load_gws())
{
xlog("L_ERR", "Error loading PSTN gateways - M=$rm RURI=$ru F=$fu
T=$tu IP=$si ID=$ci\n");
sl_send_reply("503", "PSTN Termination Currently Unavailable");
exit;
}
if(!next_gw())
{
xlog("L_ERR", "No PSTN gateways available - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
sl_send_reply("503", "PSTN Termination Currently Unavailable");
exit;
}
setflag(21);
t_on_failure("1");
route(3);
}
}
xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci\n");
sl_send_reply("404", "User Not Found");
exit;
}
########################################################################
# Request route 'base-route-local'
########################################################################
route[8]
{
t_on_reply("1");
if(t_check_trans())
{
xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
if(!t_relay())
{
sl_reply_error();
}
}
else
{
xlog("L_INFO", "Dropping mis-routed request - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
}
exit;
}
########################################################################
# Request route 'base-route-generic'
########################################################################
route[9]
{
xlog("L_INFO", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci\n");
sl_send_reply("501", "Method Not Supported Here");
exit;
}
########################################################################
# Request route 'base-filter-failover'
########################################################################
route[10]
{
if(!t_check_status("408|500|503"))
{
xlog("L_INFO", "No failover routing needed for this response code -
M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
route(1);
exit;
}
}
########################################################################
# Reply route 'base-standard-reply'
########################################################################
onreply_route[1]
{
xlog("L_INFO", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n");
exit;
}
########################################################################
# Reply route 'base-nat-reply'
########################################################################
onreply_route[2]
{
xlog("L_INFO", "NAT-Reply - S=$rs D=$rr F=$fu T=$tu IP=$si
ID=$ci\n");
if(nat_uac_test("1"))
{
fix_nated_contact();
}
if(isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]")
{
if(!search("^Content-Length:[ ]*0"))
{
force_rtp_proxy();
}
}
exit;
}
########################################################################
# Failure route 'pstn-failover'
########################################################################
failure_route[1]
{
xlog("L_INFO", "Failure route for PSTN entered - M=$rm RURI=$ru F=$fu
T=$tu IP=$si ID=$ci\n");
route(10);
if(!next_gw())
{
xlog("L_ERR", "Failed to select next PSTN gateway - M=$rm RURI=$ru F=$fu
T=$tu IP=$si ID=$ci\n");
route(1);
exit;
}
t_on_failure("1");
route(3);
}
########################################################################
# Failure route 'base-standard-failure'
########################################################################
failure_route[2]
{
route(10);
route(1);
}
##asterisk sip.conf##
[general]
matchexterniplocally=yes
canreinvite=no
externip=xxx.206.xxx.136
localnet=10.3.1.0/255.255.255.0
context=default
bindport=5061
bindaddr=0.0.0.0
sipdebug=yes
nat=yes
[openser]
type=friend
context=default
insecure=very
externalnotify=yes
allow=all
##ser log##
Dec 7 04:04:52 phonesys-slave openser[24602]: Request leaving server,
D-URI='<null>' - M=INVITE RURI=sip:500@10.3.1.31:5061;transport=udp
F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115
ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115
Dec 7 04:04:52 phonesys-slave openser[24597]: Request leaving server,
D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31
T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115
Dec 7 04:04:53 phonesys-slave openser[24602]: Request leaving server,
D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31
T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115
Dec 7 04:04:54 phonesys-slave openser[24597]: Request leaving server,
D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31
T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115
Dec 7 04:04:56 phonesys-slave openser[24602]: Request leaving server,
D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31
T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115
Dec 7 04:05:00 phonesys-slave openser[24597]: Request leaving server,
D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31
T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115
Dec 7 04:05:04 phonesys-slave openser[24602]: Request leaving server,
D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31
T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115
Dec 7 04:05:08 phonesys-slave openser[24597]: Request leaving server,
D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31
T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115
Dec 7 04:05:12 phonesys-slave openser[24589]: Request leaving server,
D-URI='<null>' - M=BYE RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31
T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115
Dec 7 04:05:13 phonesys-slave openser[24599]: Request leaving server,
D-URI='<null>' - M=BYE RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31
T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115
Dec 7 04:05:16 phonesys-slave openser[24597]: Request leaving server,
D-URI='<null>' - M=BYE RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31
T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115
##asterisk sip debug##
<--- SIP read from 10.3.1.31:5060 --->
INVITE sip:500@10.3.1.31:5061;transport=udp SIP/2.0
Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes>
Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0
Via: SIP/2.0/UDP
10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53
From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159
To: <sip:500@10.3.1.31>
Contact: <sip:pbaker2@10.3.1.115:5060>
Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115
CSeq: 8992 INVITE
Max-Forwards: 69
Content-Type: application/sdp
User-Agent: X-Lite release 1105d
Content-Length: 252
v=0
o=pbaker2 3001829617 3001829777 IN IP4 10.3.1.115
s=X-Lite
c=IN IP4 10.3.1.31
t=0 0
m=audio 35128 RTP/AVP 3 97 101
a=rtpmap:3 gsm/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=nortpproxy:yes
<------------->
--- (13 headers 12 lines) ---
Sending to 10.3.1.31 : 5060 (no NAT)
Using INVITE request as basis request - 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115
No user 'pbaker2' in SIP users list
Found peer 'openser' for 'pbaker2' from 10.3.1.31:5060
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 10.3.1.31:35128
Found audio description format gsm for ID 3
Found audio description format speex for ID 97
Found audio description format telephone-event for ID 101
Capabilities: us - 0x27f9fff
(g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|t140),
peer - audio=0x202 (gsm|speex)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x202
(gsm|speex)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event),
combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.3.1.31:35128
Looking for 500 in default (domain 10.3.1.31)
list_route: hop: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes>
<--- Transmitting (no NAT) to 10.3.1.31:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31
Via: SIP/2.0/UDP
10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53
Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes>
From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159
To: <sip:500@10.3.1.31>
Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115
CSeq: 8992 INVITE
User-Agent: Asterisk PBX SVN-trunk-r91598
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:500@167.206.216.136>
Content-Length: 0
<------------>
Audio is at 167.206.216.136 port 37712
Adding codec 0x2 (gsm) to SDP
Adding codec 0x200 (speex) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
phonesys-slave*CLI>
<--- Reliably Transmitting (no NAT) to 10.3.1.31:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31
Via: SIP/2.0/UDP
10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53
Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes>
From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159
To: <sip:500@10.3.1.31>;tag=as70a2356d
Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115
CSeq: 8992 INVITE
User-Agent: Asterisk PBX SVN-trunk-r91598
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:500@167.206.216.136>
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 87430933 87430933 IN IP4 167.206.216.136
s=Asterisk PBX SVN-trunk-r91598
c=IN IP4 167.206.216.136
t=0 0
m=audio 37712 RTP/AVP 3 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
Retransmitting #1 (no NAT) to 10.3.1.31:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31
Via: SIP/2.0/UDP
10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53
Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes>
From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159
To: <sip:500@10.3.1.31>;tag=as70a2356d
Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115
CSeq: 8992 INVITE
User-Agent: Asterisk PBX SVN-trunk-r91598
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:500@167.206.216.136>
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 87430933 87430933 IN IP4 167.206.216.136
s=Asterisk PBX SVN-trunk-r91598
c=IN IP4 167.206.216.136
t=0 0
m=audio 37712 RTP/AVP 3 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #2 (no NAT) to 10.3.1.31:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31
Via: SIP/2.0/UDP
10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53
Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes>
From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159
To: <sip:500@10.3.1.31>;tag=as70a2356d
Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115
CSeq: 8992 INVITE
User-Agent: Asterisk PBX SVN-trunk-r91598
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:500@167.206.216.136>
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 87430933 87430933 IN IP4 167.206.216.136
s=Asterisk PBX SVN-trunk-r91598
c=IN IP4 167.206.216.136
t=0 0
m=audio 37712 RTP/AVP 3 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #3 (no NAT) to 10.3.1.31:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31
Via: SIP/2.0/UDP
10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53
Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes>
From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159
To: <sip:500@10.3.1.31>;tag=as70a2356d
Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115
CSeq: 8992 INVITE
User-Agent: Asterisk PBX SVN-trunk-r91598
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:500@167.206.216.136>
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 87430933 87430933 IN IP4 167.206.216.136
s=Asterisk PBX SVN-trunk-r91598
c=IN IP4 167.206.216.136
t=0 0
m=audio 37712 RTP/AVP 3 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #4 (no NAT) to 10.3.1.31:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31
Via: SIP/2.0/UDP
10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53
Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes>
From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159
To: <sip:500@10.3.1.31>;tag=as70a2356d
Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115
CSeq: 8992 INVITE
User-Agent: Asterisk PBX SVN-trunk-r91598
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:500@167.206.216.136>
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 87430933 87430933 IN IP4 167.206.216.136
s=Asterisk PBX SVN-trunk-r91598
c=IN IP4 167.206.216.136
t=0 0
m=audio 37712 RTP/AVP 3 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #5 (no NAT) to 10.3.1.31:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31
Via: SIP/2.0/UDP
10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53
Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes>
From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159
To: <sip:500@10.3.1.31>;tag=as70a2356d
Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115
CSeq: 8992 INVITE
User-Agent: Asterisk PBX SVN-trunk-r91598
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:500@167.206.216.136>
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 87430933 87430933 IN IP4 167.206.216.136
s=Asterisk PBX SVN-trunk-r91598
c=IN IP4 167.206.216.136
t=0 0
m=audio 37712 RTP/AVP 3 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Dec 7 04:45:50] NOTICE[25642]: rtp.c:998 process_rfc3389: Comfort noise support
incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP:
10.3.1.31
Retransmitting #6 (no NAT) to 10.3.1.31:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31
Via: SIP/2.0/UDP
10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53
Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes>
From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159
To: <sip:500@10.3.1.31>;tag=as70a2356d
Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115
CSeq: 8992 INVITE
User-Agent: Asterisk PBX SVN-trunk-r91598
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:500@167.206.216.136>
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 87430933 87430933 IN IP4 167.206.216.136
s=Asterisk PBX SVN-trunk-r91598
c=IN IP4 167.206.216.136
t=0 0
m=audio 37712 RTP/AVP 3 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Dec 7 04:45:55] WARNING[25642]: chan_sip.c:2334 retrans_pkt: Maximum retries exceeded
on transmission 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 for seqno 8992 (Critical
Response)
[Dec 7 04:45:55] WARNING[25642]: chan_sip.c:2361 retrans_pkt: Hanging up call
2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 - no reply to our critical packet.
Really destroying SIP dialog '2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115'
Method: INVITE
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11:58 a.m.
Daniel,
Thanks for your response...
Interestingly enough, I'm not sure why asterisk is telling SER to respond to the
public IP?? When I commented out externip, everything works as it should, but this will
obviously break external SIP clients! Does anyone else have an issue like this?
### ASTERISK CONFIG ### 1.4 SVN HEAD
[general]
matchexterniplocally=yes
canreinvite=no
externip=xxx.206.xxx.136
localnet=10.3.1.0/255.255.255.0
context=default ; Default context for incoming calls
bindport=5061 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
sipdebug=yes ; Turn on SIP debugging by default, from
nat=no
[openser]
type=friend
context=default
insecure=very
externalnotify=yes
allow=all
host=10.3.1.31
Sip Client = 10.3.1.115
OpenSer = 10.3.1.31:5060
Asterisk = 10.3.1.31:5061
Outside IP (NAT) = xxx.206.xxx.136
|Time | 10.3.1.115 | 10.3.1.31 | xxx.206.xxx.136 |
|6.791 | Request: INVITE sip | |SIP/SDP: Request:
INVITE sip:500@10.3.1.31, with session description
| |(5060) ------------------> (5060) | |
|6.791 | Status: 100 Trying | |SIP: Status: 100
Trying
| |(5060) <------------------ (5060) | |
|6.791 | Status: 407 Proxy A | |SIP: Status: 407
Proxy Authentication Required
| |(5060) <------------------ (5060) | |
|6.794 | Request: ACK sip:50 | |SIP: Request: ACK
sip:500@10.3.1.31
| |(5060) ------------------> (5060) | |
|6.799 | Request: INVITE sip | |SIP/SDP: Request:
INVITE sip:500@10.3.1.31, with session description
| |(5060) ------------------> (5060) | |
|6.800 | Status: 100 Trying | |SIP: Status: 100
Trying
| |(5060) <------------------ (5060) | |
|6.803 | Status: 200 OK, wit | |SIP/SDP: Status:
200 OK, with session description
| |(5060) <------------------ (5060) | |
|6.817 | Request: ACK sip:50 | |SIP: Request: ACK
sip:500@xxx.206.xxx.136
| |(5060) ------------------> (5060) | |
|6.817 | | Request: ACK sip:50 |SIP: Request: ACK
sip:500@xxx.206.xxx.136
| | |(5060) ------------------> (5060) |
|7.804 | Status: 200 OK, wit | |SIP/SDP: Status:
200 OK, with session description
| |(5060) <------------------ (5060) | |
|7.806 | Request: ACK sip:50 | |SIP: Request: ACK
sip:500@xxx.206.xxx.136
| |(5060) ------------------> (5060) | |
|7.807 | | Request: ACK sip:50 |SIP: Request: ACK
sip:500@xxx.206.xxx.136
| | |(5060) ------------------> (5060) |
|8.803 | Status: 200 OK, wit | |SIP/SDP: Status:
200 OK, with session description
| |(5060) <------------------ (5060) | |
|8.805 | Request: ACK sip:50 | |SIP: Request: ACK
sip:500@xxx.206.xxx.136
| |(5060) ------------------> (5060) | |
|8.805 | | Request: ACK sip:50 |SIP: Request: ACK
sip:500@xxx.206.xxx.136
| | |(5060) ------------------> (5060) |
|10.803 | Status: 200 OK, wit | |SIP/SDP: Status:
200 OK, with session description
| |(5060) <------------------ (5060) | |
|10.806 | Request: ACK sip:50 | |SIP: Request: ACK
sip:500@xxx.206.xxx.136
| |(5060) ------------------> (5060) | |
|10.807 | | Request: ACK sip:50 |SIP: Request: ACK
sip:500@xxx.206.xxx.136
| | |(5060) ------------------> (5060) |
|14.803 | Status: 200 OK, wit | |SIP/SDP: Status:
200 OK, with session description
| |(5060) <------------------ (5060) | |
|14.805 | Request: ACK sip:50 | |SIP: Request: ACK
sip:500@xxx.206.xxx.136
| |(5060) ------------------> (5060) | |
|14.806 | | Request: ACK sip:50 |SIP: Request: ACK
sip:500@xxx.206.xxx.136
| | |(5060) ------------------> (5060) |
|18.803 | Status: 200 OK, wit | |SIP/SDP: Status:
200 OK, with session description
| |(5060) <------------------ (5060) | |
|18.806 | Request: ACK sip:50 | |SIP: Request: ACK
sip:500@xxx.206.xxx.136
| |(5060) ------------------> (5060) | |
|18.806 | | Request: ACK sip:50 |SIP: Request: ACK
sip:500@xxx.206.xxx.136
| | |(5060) ------------------> (5060) |
|22.802 | Status: 200 OK, wit | |SIP/SDP: Status:
200 OK, with session description
| |(5060) <------------------ (5060) | |
|22.805 | Request: ACK sip:50 | |SIP: Request: ACK
sip:500@xxx.206.xxx.136
| |(5060) ------------------> (5060) | |
|22.805 | | Request: ACK sip:50 |SIP: Request: ACK
sip:500@xxx.206.xxx.136
| | |(5060) ------------------> (5060) |
|38.046 | Request: BYE sip:50 | |SIP: Request: BYE
sip:500@xxx.206.xxx.136
| |(5060) ------------------> (5060) | |
|38.046 | | Request: BYE sip:50 |SIP: Request: BYE
sip:500@xxx.206.xxx.136
| | |(5060) ------------------> (5060) |
|38.491 | | Request: BYE sip:50 |SIP: Request: BYE
sip:500@xxx.206.xxx.136
| | |(5060) ------------------> (5060) |
|39.491 | | Request: BYE sip:50 |SIP: Request: BYE
sip:500@xxx.206.xxx.136
| | |(5060) ------------------> (5060) |
|39.548 | Request: BYE sip:50 | |SIP: Request: BYE
sip:500@xxx.206.xxx.136
| |(5060) ------------------> (5060) | |
|41.491 | | Request: BYE sip:50 |SIP: Request: BYE
sip:500@xxx.206.xxx.136
| | |(5060) ------------------> (5060) |
|42.788 | Request: BYE sip:50 | |SIP: Request: BYE
sip:500@xxx.206.xxx.136
| |(5060) ------------------> (5060) | |
|45.494 | | Request: BYE sip:50 |SIP: Request: BYE
sip:500@xxx.206.xxx.136
| | |(5060) ------------------> (5060) |
|49.495 | | Request: BYE sip:50 |SIP: Request: BYE
sip:500@xxx.206.xxx.136
| | |(5060) ------------------> (5060) |
6202
days inactive
6202
days old
2 comments
2 participants
participants (2)
-
Daniel-Constantin Mierla -
Patrick Baker