sr-users September 2009

sr-users@lists.kamailio.org

69 participants
105 discussions

[Kamailio-Users] Kamailio not starting - no clue in syslog

by Anders

Hi, I get an error when trying to start Kamailio: ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed I've searched around and found others with the same problem - but they usually found a lead to the problem in the syslog. All I get in the syslog when trying to start Kamailio is this line: Sep 28 05:34:01 server /USR/SBIN/CRON[10786]: (root) CMD (echo "call kamailio_cdrs();"|/usr/bin/mysql -u xxx -pyyy kamailio >/dev/null) Any ideas where and how to look? Thanks!

15 years, 1 month

[Kamailio-Users] DNS lookup for requests within dialog

by catalina oancea

Hi all, I tested the following scenario: astreisk kamailio provider(hostname) --------------------> dns lookup --------------------> REGISTER 1 REGISTER 1 <-------------------- <-------------------- 401 401 --------------------> dns lookup --------------------> REGISTER 2 REGISTER 2 (same dialog, increased cseq) It seems that kamailio does dns lookup again when receiving REGISTER 2, although both registers are part of the same dialog. Is this the expected behavior? Thanks, Catalina

15 years, 1 month

[Kamailio-Users] add_path_received and OBP problems

by Florian Meister

Hi, I'm trying to set up a HA-OBP for both natted and non-natted users. Atm my setup looks like that: [UAC] <---> (NAT) <---> [LB] <---> [REG + Proxy] <---> [PSTN GW] At the loadbalancer I use add_path and add_path_received to add a path-header to the request sent to the Registrar. (Of course only for register requests). I use a few parameters for telling the registrar module to use the path header. When looking in the mysql-db I can see the correct value in the path-field in the location-table. When getting an invite from the PSTN gateway for one of the UAC's the Proxy is adding a route-header according to the path-value in the mysql. That seems also correct. Now to my problem: The loadbalancer is sending out the INVITE to the private-ip, not to the public-nat-ip of the UAC. It seems that the received-parameter in the Route-header is somehow ignored. I am using the use_received parameter in the loadbalancer, but it does not work. In the following you can find the SIP trace (made on the loadbalancer) and the loadbalancer config, since I think there is the problem. Thanks, Florian Here is the SIP-trace of the INVITE: 09:49:07.785294 IP [Proxy+Registrar].5060 > [Loadbalancer].5060: SIP, length: 1101 E..i..@.?.Yb..,F..,+.....U.BINVITE sip:435572949012@[PRIVATE IP of UAC];user=phone SIP/2.0 Record-Route: <sip:[Proxy+Registrar];lr;ftag=as33339e2e> Via: SIP/2.0/UDP [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0 Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060 Route: <sip:[Loadbalancer];lr;received="sip:[PUBLIC NAT IP of UAC]:64224"> Max-Forwards: 69 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e To: <sip:+435572949012@[Proxy+Registrar]> Contact: <sip:4366480588134@[PSTN Gateway]> Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Fri, 25 Sep 2009 07:40:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 312 v=0 o=root 1940822246 1940822246 IN IP4 [PSTN Gateway] s=Asterisk PBX 1.6.1.5 c=IN IP4 [PSTN Gateway] t=0 0 m=audio 19178 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 09:49:07.786459 IP [Loadbalancer].5060 > [Proxy+Registrar].5060: SIP, length: 445 E.....@.@.Z...,+..,F........SIP/2.0 100 Giving a try Via: SIP/2.0/UDP [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0;rport=5060;received=[Proxy+Registrar] Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e To: <sip:+435572949012@[Proxy+Registrar]> Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102 INVITE Server: Kamailio (1.5.1-notls (i386/linux)) Content-Length: 0 09:49:07.786608 IP [Loadbalancer].5060 > [PRIVATE IP of UAC].5060: SIP, length: 1307 E..7..@.@..@..,+.........#>.INVITE sip:435572949012@[PRIVATE IP of UAC];user=phone SIP/2.0 Record-Route: <sip:[Loadbalancer];lr;ftag=as33339e2e;nat=yes> Record-Route: <sip:[Proxy+Registrar];lr;ftag=as33339e2e> Via: SIP/2.0/UDP [Loadbalancer];branch=z9hG4bK3d51.207ba0a5.0 Via: SIP/2.0/UDP [Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0 Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060 Route: <sip:[Loadbalancer];lr;received="sip:[PUBLIC NAT IP of UAC]:64224"> Max-Forwards: 68 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e To: <sip:+435572949012@[Proxy+Registrar]> Contact: <sip:4366480588134@[PSTN Gateway]> Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Fri, 25 Sep 2009 07:40:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 330 Path: <sip:[Loadbalancer];lr> v=0 o=root 1940822246 1940822246 IN IP4 [PSTN Gateway] s=Asterisk PBX 1.6.1.5 c=IN IP4 [Loadbalancer] t=0 0 m=audio 35480 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes 09:49:08.258116 IP [Loadbalancer].5060 > [PRIVATE IP of UAC].5060: SIP, length: 1307 E..7..@.@..@..,+.........#>.INVITE sip:435572949012@[PRIVATE IP of UAC];user=phone SIP/2.0 Record-Route: <sip:[Loadbalancer];lr;ftag=as33339e2e;nat=yes> Record-Route: <sip:[Proxy+Registrar];lr;ftag=as33339e2e> Via: SIP/2.0/UDP [Loadbalancer];branch=z9hG4bK3d51.207ba0a5.0 Via: SIP/2.0/UDP [Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0 Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060 Route: <sip:[Loadbalancer];lr;received="sip:[PUBLIC NAT IP of UAC]:64224"> Max-Forwards: 68 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e To: <sip:+435572949012@[Proxy+Registrar]> Contact: <sip:4366480588134@[PSTN Gateway]> Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Fri, 25 Sep 2009 07:40:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 330 Path: <sip:[Loadbalancer];lr> v=0 o=root 1940822246 1940822246 IN IP4 [PSTN Gateway] s=Asterisk PBX 1.6.1.5 c=IN IP4 [Loadbalancer] t=0 0 m=audio 35480 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes 09:49:09.258109 IP [Loadbalancer].5060 > [PRIVATE IP of UAC].5060: SIP, length: 1307 E..7..@.@..@..,+.........#>.INVITE sip:435572949012@[PRIVATE IP of UAC];user=phone SIP/2.0 Record-Route: <sip:[Loadbalancer];lr;ftag=as33339e2e;nat=yes> Record-Route: <sip:[Proxy+Registrar];lr;ftag=as33339e2e> Via: SIP/2.0/UDP [Loadbalancer];branch=z9hG4bK3d51.207ba0a5.0 Via: SIP/2.0/UDP [Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0 Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060 Route: <sip:[Loadbalancer];lr;received="sip:[PUBLIC NAT IP of UAC]:64224"> Max-Forwards: 68 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e To: <sip:+435572949012@[Proxy+Registrar]> Contact: <sip:4366480588134@[PSTN Gateway]> Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Fri, 25 Sep 2009 07:40:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 330 Path: <sip:[Loadbalancer];lr> v=0 o=root 1940822246 1940822246 IN IP4 [PSTN Gateway] s=Asterisk PBX 1.6.1.5 c=IN IP4 [Loadbalancer] t=0 0 m=audio 35480 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes 09:49:11.258107 IP [Loadbalancer].5060 > [PRIVATE IP of UAC].5060: SIP, length: 1307 E..7..@.@..@..,+.........#>.INVITE sip:435572949012@[PRIVATE IP of UAC];user=phone SIP/2.0 Record-Route: <sip:[Loadbalancer];lr;ftag=as33339e2e;nat=yes> Record-Route: <sip:[Proxy+Registrar];lr;ftag=as33339e2e> Via: SIP/2.0/UDP [Loadbalancer];branch=z9hG4bK3d51.207ba0a5.0 Via: SIP/2.0/UDP [Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0 Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060 Route: <sip:[Loadbalancer];lr;received="sip:[PUBLIC NAT IP of UAC]:64224"> Max-Forwards: 68 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e To: <sip:+435572949012@[Proxy+Registrar]> Contact: <sip:4366480588134@[PSTN Gateway]> Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Fri, 25 Sep 2009 07:40:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 330 Path: <sip:[Loadbalancer];lr> v=0 o=root 1940822246 1940822246 IN IP4 [PSTN Gateway] s=Asterisk PBX 1.6.1.5 c=IN IP4 [Loadbalancer] t=0 0 m=audio 35480 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes 09:49:15.023388 IP [Proxy+Registrar].5060 > [Loadbalancer].5060: SIP, length: 397 E.....@.?.\"..,F..,+......a.CANCEL sip:435572949012@[PRIVATE IP of UAC];user=phone SIP/2.0 Via: SIP/2.0/UDP [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] To: <sip:+435572949012@[Proxy+Registrar]> CSeq: 102 CANCEL Max-Forwards: 70 User-Agent: OpenSER (1.3.2-notls (x86_64/linux)) Content-Length: 0 09:49:15.024138 IP [Loadbalancer].5060 > [Proxy+Registrar].5060: SIP, length: 413 E.....@.@.[...,+..,F........SIP/2.0 200 canceling Via: SIP/2.0/UDP [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0;rport=5060;received=[Proxy+Registrar] From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] To: <sip:+435572949012@[Proxy+Registrar]>;tag=77b8b5897dbef36847302656cec9825e-db87 CSeq: 102 CANCEL Server: Kamailio (1.5.1-notls (i386/linux)) Content-Length: 0 09:49:15.024576 IP [Loadbalancer].5060 > [Proxy+Registrar].5060: SIP, length: 493 E.. ..@.@.Z...,+..,F........SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0;rport=5060;received=[Proxy+Registrar] Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e To: <sip:+435572949012@[Proxy+Registrar]>;tag=77b8b5897dbef36847302656cec9825e-db87 Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102 INVITE Server: Kamailio (1.5.1-notls (i386/linux)) Content-Length: 0 09:49:15.025180 IP [Proxy+Registrar].5060 > [Loadbalancer].5060: SIP, length: 433 E.....@.?.[...,F..,+.......MACK sip:435572949012@[PRIVATE IP of UAC];user=phone SIP/2.0 Via: SIP/2.0/UDP [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] To: <sip:+435572949012@[Proxy+Registrar]>;tag=77b8b5897dbef36847302656cec9825e-db87 CSeq: 102 ACK Max-Forwards: 70 User-Agent: OpenSER (1.3.2-notls (x86_64/linux)) Content-Length: 0 ----------------This is the config of my loadbalancer--------------------: ####### Global Parameters ######### debug=100 log_stderror=yes log_facility=LOG_LOCAL0 fork=no children=4 disable_tcp=yes port=5060 listen=[Loadbalancer] ####### Modules Section ######## mpath="/usr/lib/kamailio/modules/" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "nathelper.so" loadmodule "dispatcher.so" loadmodule "path.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) # -- nathelper modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy/rtpproxy.sock") modparam("nathelper", "received_avp", "$avp(i:80)") # -- dispatcher params -- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") # -- path params -- modparam("path", "use_received", 1) ####### Routing Logic ######## # main request routing logic route{ xlog("L_ALERT", "[$rm] from [$fu] to [$tu]\n"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # NAT detection route(4); if (has_totag()) { xlog("L_ALERT", "_____HASTOTAG [$rm] from [$fu] to [$tu]\n"); # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { xlog("L_ALERT", "_____LOOSE_ROUTE [$rm] from [$fu] to [$tu]\n"); t_relay(); exit; } else { if ( is_method("ACK") ) { if ( t_check_trans() ) { xlog("L_ALERT", "_____NON-LOOSE, STATEFUL-ACK [$rm] from [$fu] to [$tu]\n"); # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { xlog("L_ALERT", "_____ACK_WITHOUT_TRANSACTION [$rm] from [$fu] to [$tu]\n"); # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # record routing if (!is_method("REGISTER|MESSAGE")) xlog("L_ALERT", "_____RECORD_ROUTING [$rm] from [$fu] to [$tu]\n"); record_route(); #if ($rU==NULL) { # # request with no Username in RURI # sl_send_reply("484","Address Incomplete"); # exit; #} if (check_route_param("nat=yes")) { setbflag(6); } #if (isflagset(5) || isbflagset(6)) { route(5); #} /* example how to enable some additional event routes */ if (is_method("INVITE")) { t_on_reply("1"); t_on_failure("1"); } if(!ds_is_from_list()) { ds_select_domain("0", "0"); } if (!t_relay()) { sl_reply_error(); } } # Caller NAT detection route route[4]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { add_path_received(); fix_nated_register(); } else { fix_nated_contact(); } setflag(5); } else { add_path(); } return; } # RTPProxy control route[5] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); return; } onreply_route[1] { xdbg("incoming reply\n"); #if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { if (status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset(6)) { fix_nated_contact(); } } failure_route[1] { #if (is_method("INVITE") && (isbflagset(6) || isflagset(5))) { if (is_method("INVITE")) { unforce_rtp_proxy(); } if (t_was_cancelled()) { exit; } }

15 years, 1 month

Re: [Kamailio-Users] won't start - no error found in syslog

by Florian Meister

> There's rwx for root on /var/run - and I'm on with root trying to start > it... What about the user/group in the config (kamailio.conf) file ? Regards, Florian

15 years, 1 month

[Kamailio-Users] won't start - no error found in syslog

by Anders

15 years, 1 month

Re: [Kamailio-Users] content about kamailio start error

by zhangchao00001

Dear sir: I get the error during during processing of SIP message. The config file i used is below (the config works well on other kamailio server). By the way , the email system i used dosen't support:all the time please keep CC-ing to mailing list address ormaybe iadd the mailing list address in a wrong place, mails with a cc-ing added will be posted back. is it CC:users@lists.kamailio.org? debug=3 fork=yes log_stderror=yes port=5060 ####### Modules Section ######## mpath="/usr/local/lib/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "path.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "domain.so" loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") modparam("rr", "enable_full_lr", 1) modparam("rr", "append_fromtag", 0) modparam("registrar", "method_filtering", 1) modparam("registrar", "append_branches", 0) modparam("registrar", "max_contacts", 10) modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 0) modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("domain", "db_url", "mysql://root:123456@localhost/SIP") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://root:123456@localhost/SIP") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://root:123456@localhost/SIP") modparam("auth_db", "load_credentials", "") modparam("domain", "db_url", "mysql://root:123456@localhost/SIP") modparam("domain", "db_mode", 1) modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) modparam("presence|presence_xml", "db_url", "mysql://root:123456@localhost/SIP") modparam("presence_xml", "force_active", 1) modparam("presence", "server_address", "sip:26.28.29.253:5060") modparam("nathelper", "rtpproxy_sock", "udp:26.28.249.253:7722") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "rtpproxy_tout", 10) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") modparam("registrar|nathelper", "received_avp", "$avp(i:80)") modparam("usrloc", "nat_bflag", 6) modparam("path", "use_received", 1) ####### Routing Logic ######## route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(msg:len>=2048) { sl_send_reply("513","Message too big"); exit; } route(4); if (has_totag()) { if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); if(!uri==myself) { append_hf("P-hint: outbound\r\n"); if(is_method("REGISTER|MESSAGE")) { if(is_method("REGISTER")) { if(!add_path_received()) { sl_send_reply("503","Internal Path Error!"); exit(); } } } else { record_route(); } route(1); } # authentication route(3); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if( is_method("PUBLISH|SUBSCRIBE")) route(2); if (is_method("REGISTER")) { if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { if (check_route_param("nat=yes")) { setbflag(6); } if (isflagset(5) || isbflagset(6)) { route(5); } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) /*if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; }*/ } else { # authenticate if from local subscriber (uncomment to enable auth) /*if (from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated }*/ } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(5); } return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset(6)) { fix_nated_contact(); } } failure_route[1] { if (is_method("INVITE") && (isbflagset(6) || isflagset(5))) { unforce_rtp_proxy(); } if (t_was_cancelled()) { exit; } if (t_check_status("486|408")) { sethostport("26.28.249.125:5060"); append_branch(); # do not set the missed call flag again t_relay(); } } 在2009-09-25，"Daniel-Constantin Mierla" <miconda(a)gmail.com> 写道： >Hello, > >do keep mailing list CC-ed for every mail, otherwise I will stop replying. > >On 25.09.2009 3:33 Uhr, zhangchao00001 wrote: >> Dear sir: >> thank you for your reply. >> The Kamailio -V result : >> version: kamailio 1.5.2-notls (i386/linux) >> flags: STATISTICS, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, >> SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >> MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304 >> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >> svnrevision: unknown >> @(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $ >> main.c compiled on 10:23:04 Sep 23 2009 with gcc 4.3.0 >> when there's no user, it reports no error, when a UAC register, it >> reports the error below: >> Sep 25 09:24:21 [3323] ERROR:core:build_req_buf_from_sip_req: out of >> pkg memory >> Sep 25 09:24:21 [3323] ERROR:tm:print_uac_request: no more shm_mem >> Sep 25 09:24:21 [3323] ERROR:tm:t_forward_nonack: failure to add branches >> Sep 25 09:24:21 [3323] ERROR:tm:_reply_light: failed to allocate shmem >> buffer >> Sep 25 09:24:21 [3323] ERROR:tm:relay_reply: no more share memory >> what should i do to fix it? >does it happen after startup, during processing of SIP message? Is a >stress test? What is the list of loaded modules? If you can share the >config, would be good. > >Daniel > >-- >Daniel-Constantin Mierla >* Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin >* http://www.asipto.com/index.php/sip-router-masterclass/ > 网易邮箱用户购物独享现金返还网易邮箱用户购物独享现金返还网易邮箱用户购物独享现金返还 "中国制造",讲述中国60年往事 "中国制造",讲述中国60年往事

15 years, 1 month

[Kamailio-Users] Need Assistance Tracking Down A SIP Signaling Problem

by Geoffrey Mina

Hello, I am wondering if anyone can help me determine what the problem with my SIP signaling is here. I have attached two Excel files (can provide other formats if someone doesn't have Excel). The environments in both scenarios are almost identical... although I am dealing with two different carriers upstream. In my environment, I have Kamailio (1.5) sitting in front of a multitude of Asterisk machines. I am using the dispatcher module to distribute INVITE requests across the network. I am doing some interop with a new carrier and we are struggling a bit with some looping scenarios. They are sending invites to my Kamailio server, I am forwarding to asterisk. The two XLS files (Success.xls) and (Fail.xls) have sip_trace information for the transactions. As you can see in the Fail.xls, the ACK to the 200 OK gets looped back to the Kamailio server and never ends up reaching the Asterisk machine. In the Success.xls, this doesn't happen. The only thing I can see that's different is the Success carrier appears to be replacing the RURI in the ACK with the value of the Contact header... in the failed one, the RURI of the ACK is the IP of my Kamailio gateway. I have read through the RFC 3261 to try and figure out what is breaking this, but haven't been able to. I basically need to determine if the signalling is correct... what is wrong with my config... and if the signalling is wrong, what the carrier is doing incorrectly. Thanks in advance! Geoff

15 years, 1 month

[Kamailio-Users] Diameter

by Ghaith ALKAYYEM

Hello list, Does anybody have any idea or recommendation about integrating diameter with SER? As far as you know the available module is obsolete, i'm trying to implement the module "CDP" available with OpenIMS: http://www.openimscore.org/docs/ser_ims/CDP.html Thank you

15 years, 1 month

[Kamailio-Users] Unable to lookup host in c= line,

by Tincho ylm

Hi all! I'm trying to make a realtime environment openser-asterisk. At this moment, everything is working fine but I have problems with VM feature. What I'm doing is: A call B If B is busy and returns 486 Openser send this to Asterisk. failure_route[1] { if(!t_was_cancelled()) { if (t_check_status("487|486")) { rewritehostport("200.xx.xx.xx:5060"); append_branch(); xlog("L_INFO","---> To Voicemail...\n"); setflag(10); route(1); exit; } if (isbflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } } } Asterisk cant handle it because this error: chan_sip.c:5121 process_sdp: Unable to lookup host in c= line, 'IN IP4 200.xx.xx.xx200.xx.xx.xx' chan_sip.c:5121 process_sdp: Unable to lookup host in c= line, 'IN IP4 200.xx.xx.xx200.xx.xx.xx' chan_sip.c:5121 process_sdp: Unable to lookup host in c= line, 'IN IP4 200.xx.xx.xx200.xx.xx.xx' I'm using: Kamailio 1.4.4 Rtpproxy Asterisk 1.4.22 Is it a rtpproxy issue? it can fix it? If I make a ethereal capture, I can see in the MESSAGE BODY this: connection inrfomation (c): IN IP4 200.xx.xx.xx200.xx.xx.xx What can I do? thanks in advance.

15 years, 1 month

[Serusers] Search trough status line for specific reason phrase

by Josip Djuricic

Dear, is there any way to search inside Status-Line of response message? I have strange b2bua that always returns SIP 500 just with different reason phrase. I need to match on them so that I can rewrite failure messages. I thought about something like this: failure_route[1] { if (t_check_status("500")) { #here I need to search trough status line for specific reasons, but have no idea how to do that if (search("^Specific message")) { t_reply("480", "Temporarily Unavailable"); break; } } } Thank you J.

15 years, 1 month

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sr-users September 2009