Hello,all: I am trying to use asterisk and opensips and follow the link: http://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration-1-8. when i register my sip phone, I got the 408 timeout and P-hint:outbound from ngrep -d lo -qt -W byline port 5060. when i use the example CFG to startup the opensips, i got many errors due to the module compatebility. so i change some modules for opensips-1.8, therefore I suspect the modules i loaded is wrong. --------------------------------------------CFG--------------------------------- # # $Id: opensips.cfg 8758 2012-02-29 11:59:26Z vladut-paiu $ # # OpenSIPS residential configuration script # by OpenSIPS Solutions <t...@opensips-solutions.com> # # This script was generated via "make menuconfig", from # the "Residential" scenario. # You can enable / disable more features / functionalities by # re-generating the scenario with different options.# # # Please refer to the Core CookBook at: # http://www.opensips.org/Resources/DocsCookbooks # for a explanation of possible statements, functions and parameters. #
####### Global Parameters ######### debug=5 log_stderror=yes log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ disable_dns_blacklist=yes /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* comment the next line to enable the auto discovery of local aliases based on revers DNS on IPs */ /* auto_aliases=yes */ /* alias=test.com */ # port = 5060 listen=udp:192.168.1.104:5060 # CUSTOMIZE ME disable_tcp=yes # disable_tls=yes ####### Modules Section ######## #set module path mpath="/usr/local/lib/opensips/modules/" loadmodule "db_mysql.so" loadmodule "signaling.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" #loadmodule "uri_db.so" loadmodule "uri.so" #loadmodule "xlog.so" loadmodule "acc.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "sipmsgops.so" loadmodule "domain.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_double_rr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- usrloc params ----- modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://opensips:opensipsrw@localhost/opensips") # ----- uri_db params ----- #modparam("uri_db", "use_uri_table", 0) #modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) #modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) /* uncomment the following lines to enable DB accounting also */ modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) # ----- auth_db params ----- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://opensips:opensipsrw@localhost/opensips") modparam("auth_db", "load_credentials", "") # ----- domain params ----- modparam("domain", "db_url", "mysql://opensips:opensipsrw@localhost/opensips") modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { send_reply("483","Too Many Hops"); exit; } if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } else if (is_method("INVITE")) { # even if in most of the cases is useless, do RR for # re-INVITEs alos, as some buggy clients do change route set # during the dialog. record_route(); } # route it out to whatever destination was set by loose_route() # in $du (destination URI). route(1); } else { if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after # a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction -> # ignore and discard exit; } } send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authenticate if from local subscriber if (!(method=="REGISTER") && is_from_local()) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (!db_check_from()) { send_reply("403","Forbidden auth ID"); exit; } consume_credentials(); # caller authenticated } # preloaded route checking if (loose_route()) { xlog("L_ERR", "Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]"); if (!is_method("ACK")) send_reply("403","Preload Route denied"); exit; } # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } # if not a targetting a local SIP domain, just send it out # based on DNS (calls to foreign SIP domains) if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); route(1); } # requests for my domain if (is_method("REGISTER")) { # authenticate the REGISTER requests if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if (!db_check_to()) { send_reply("403","Forbidden auth ID"); exit; } if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI send_reply("484","Address Incomplete"); exit; } # ASTERISK HOOK - BEGIN # media service number? (digits starting with *) if ($rU=~"^\*[1-9]+") { # we do provide access to media services only to our # subscribers, who were previously authenticated if (!is_from_local()) { send_reply("403","Forbidden access to media service"); exit; } #identify the services and translate to Asterisk extensions if ($rU=="*1111") { # access to own voicemail IVR seturi("sip:VM_pickup@192.168.1.104:5090"); } else if ($rU=="*2111") { # access to the "say time" announcement seturi("sip:AN_time@192.168.1.104:5090"); } else if ($rU=="*2112") { # access to the "say date" announcement seturi("sip:AN_date@192.168.1.104:5090"); } else if ($rU=="*2113") { # access to the "echo" service seturi("sip:AN_echo@192.168.1.104:5090"); } else if ($rU=~"\*3[0-9]{3}") { # access to the conference service # remove the "*3" prefix and place the "CR_" prefix strip(2); prefix("CR_"); rewritehostport("192.168.1.104:5090"); } else { # unknown service seturi("sip:AN_notavailable@192.168.1.104:5090"); } # after setting the proper RURI (to point to corresponding ASTERISK extension), # simply forward the call t_relay(); exit; } # ASTERISK HOOK - END # do lookup if (!lookup("location")) { # ASTERISK HOOK - BEGIN # callee is not registered, so different to Voicemail # First add the VM recording prefix to the RURI prefix("VMR_"); # forward to the call to Asterisk (replace below with real IP and port) rewritehostport("192.168.1.104:5090"); route(1); # ASTERISK HOOK - END exit; } # when routing via usrloc, log the missed calls also setflag(2); # arm a failure route in order to catch failed calls # targeting local subscribers; if we fail to deliver # the call to the user, we send the call to voicemail t_on_failure("1"); route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } failure_route[1] { if (t_was_cancelled()) { exit; } # if the failure code is "408 - timeout" or "486 - busy", # forward the calls to voicemail recording if (t_check_status("486|408")) { # ASTERISK HOOK - BEGIN # First revert the RURI to get the original user in RURI # Then add the VM recording prefix to the RURI revert_uri(); prefix("VMR_"); # forward to the call to Asterisk (replace below with real IP and port) rewritehostport("192.168.1.104:5090"); t_relay(); # ASTERISK HOOK - END exit; } } ============================ngrep debug======================================================== Via: SIP/2.0/UDP 192.168.1.200:16377;received=192.168.1.200;branch=z9hG4bK-d87543-e92fd965fb6c5113-1--d87543-;rport=16377. Max-Forwards: 29. Contact: <sip:101@192.168.1.200:16377;rinstance=624d83088d4ed92e>. To: "101"<sip:101@192.168.1.104>. From: "101"<sip:101@192.168.1.104>;tag=de6d7016. Call-ID: ODMyNTg1MmU2YTQzN2Q5MzAyNDJiYjY2Njk3NWE0MWI.. CSeq: 2 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. User-Agent: X-Lite release 1011s stamp 41150. Content-Length: 0. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. . U 2014/05/27 15:20:54.894213 192.168.1.104:5060 -> 192.168.1.104:5060 REGISTER sip:192.168.1.104 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.a2d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.92d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.82d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.72d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.62d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.52d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.42d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.32d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.22d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.12d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.02d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.f1d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.e1d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.d1d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.c1d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.b1d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.a1d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.91d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.81d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.71d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.61d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.51d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.41d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.31d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.21d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.11d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.01d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.f0d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.e0d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.d0d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.c0d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.b0d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.a0d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.90d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.80d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.70d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.60d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.50d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.40d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.30d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.20d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.10d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.00d88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.ffc88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.efc88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.dfc88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.cfc88ec4.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK6ed.bfc88ec4.0. Via: SIP/2.0/UDP 192.168.1.103:5060;received=192.168.1.103;branch=z9hG4bK112230595;rport=5060. From: <sip:bob@192.168.1.104>;tag=1244885899. To: <sip:bob@192.168.1.104>. Call-ID: 1761427266-506...@bjc.bgi.b.bad. CSeq: 2045 REGISTER. Contact: <sip:bob@192.168.1.103:5060>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-1000-8000-000B824017A0>". X-Grandstream-PBX: true. Max-Forwards: 22. User-Agent: Grandstream GXP2124 1.0.4.10. Supported: path. Expires: 480. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Length: 0. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. P-hint: outbound. Cheers!
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