如何设置kamailio代理服务器并将呼叫路由到Twilio?

我试图通过kamailio代理路由所有的电话twilio。 与我的configuration文件,呼叫获得连接,约30秒后自动下降。 这是因为发送到200 OK的ACK是不正确的。 Twilio期望与ruri相同的响应在200 OK响应,但kamailio发送是不同的。 如何解决这个错误?

200 OK response [email protected]|k SIP/2.0 200 OK To: <sip:[email protected]>;tag=52642973_6772d868_f5ff7ec8-2860-4963-894d-4686de291299 Via: SIP/2.0/UDP 54.69.159.69:5060;branch=z9hG4bKcdde.3a10daa4c4dfb45218165e7003bc1925.0 Via: SIP/2.0/UDP 64.2.142.90;branch=z9hG4bKcdde.0da1e6a5.0 Via: SIP/2.0/UDP 64.2.142.153:5060;received=64.2.142.153;branch=z9hG4bK0c18e9b5;rport=5060 Record-Route: <sip:107.21.211.20:5060;lr;ftag=as0b111d7b> Record-Route: <sip:54.69.159.69:5060;lr=on> Record-Route: <sip:64.2.142.90;lr=on> CSeq: 102 INVITE Call-ID: [email protected] From: "+14088271419" <sip:[email protected]>;tag=as0b111d7b Contact: <sip:10.108.170.11:5060> Content-Type: application/sdp X-Twilio-CallSid: CA2d519006d553f3acf89f1dd2fef77cc3 Content-Length: 247 v=0 o=- 1132985615 1132985615 IN IP4 54.82.64.195 s=session c=IN IP4 54.82.64.195 t=0 0 m=audio 13800 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 

ACK转发给twilio

 E@d kCACK sip:107.21.211.20:5060;lr;ftag=as0b111d7b SIP/2.0 Via: SIP/2.0/UDP 54.69.159.69:5060;branch=z9hG4bKcdde.95210bf832aa38aeeccc727ba583b66f.0 Via: SIP/2.0/UDP 54.69.159.69:5060;branch=z9hG4bKcdde.0a606545da03734a8b332edc9b21079f.0 Via: SIP/2.0/UDP 64.2.142.90;branch=z9hG4bKcdde.0da1e6a5.2 Via: SIP/2.0/UDP 64.2.142.153:5060;received=64.2.142.153;branch=z9hG4bK32a73a53;rport=5060 From: "+14088271419" <sip:[email protected]>;tag=as0b111d7b To: <sip:[email protected]:5060>;tag=52642973_6772d868_f5ff7ec8-2860-4963-894d-4686de291299 Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 ACK User-Agent: packetrino Max-Forwards: 67 Content-Length: 0 

这是我的configuration文件(IP和帐户名称已更改)

 #!KAMAILIO # # sample config file for dispatcher module # - load balancing of VoIP calls with round robin # - no TPC listening # - don't dispatch REGISTER and presence requests # Direct your questions about this file to: [email protected] # # # *** To run in debug mode: # - define WITH_DEBUG # #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* comment the next line to enable TCP */ disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ alias="54.69.159.69" port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:172.31.22.10:5060 advertise 54.69.159.69:5060 #advertised_address="54.69.159.69" #advertised_port=5060 sip_warning=no ####### Modules Section ######## #set module path mpath="/usr/local/lib64/kamailio/modules_k/:/usr/local/lib64/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "dispatcher.so" loadmodule "pdt.so" loadmodule "nathelper.so" loadmodule "avpops.so" loadmodule "htable.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- acc params ----- modparam("acc", "log_flag", 1) modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- tm params ----- modparam("tm", "fr_timer", 2000) modparam("tm", "fr_inv_timer", 40000) # ----- dispatcher params ----- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags_col", "flags") modparam("dispatcher", "priority_col", "priority") modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "force_dst", 1) #-------pdt-------------- modparam("pdt", "db_table", "pdt") modparam("pdt", "domain_column", "sdomain") modparam("pdt", "prefix_column", "prefix") modparam("pdt", "domain_column", "domain") #-----------------nat----------- modparam("nathelper", "natping_interval", 0) modparam("nathelper", "ping_nated_only", 0) #modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:172.31.22.10:5060") #-----------acc------------ modparam("acc", "report_ack", 1) ####### Routing Logic ######## # main request routing logic route { # per request initial checks route(REQINIT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) #add_rr_param(";nat=yes"); record_route_advertised_address("54.69.159.69:5060"); #record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations route(DISPATCH); } route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { t_check_trans(); 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 } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; # must be ACK after a 487 or eg 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard. exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if(!is_method("REGISTER")) return; sl_send_reply("404", "No registrar"); exit; } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; sl_send_reply("404", "Not here"); exit; } # Dispatch requests route[DISPATCH] { if(prefix2domain("2", "0")) { $ru = "sip:" + $rU + "@" + $rd; t_relay(); exit; } # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Sample failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } }