- sr-users - kamailio.org

Kamailio not updating CSeq after uac_auth() for ACK
by Terry Tailor 18 Dec '20

18 Dec '20

Hi, Here is setup details Kamailio Proxy a <-----> Kamailio Proxy b <-----> Asterisk Kamailio proxy b auth based on IP Kamailio Proxy a 1. Kamailio proxy sends an Invite request to Kamailio proxy b with CSeq: 102 INVITE 2. Kamailio proxy b check again IP auth and sends invite request to asterisk with CSeq: 102 INVITE 4 asterisk response with 401 Unauthorized with CSeq: 102 5. Kamailio proxy b sends ACK request with CSeq: 102 ACK 6. Kamailio proxy b sends INVITE request with CSeq: 103 INVITE 7. Asterisk sends 100 trying and 180 ringing with CSeq: 103 INVITE 8. Kamailio proxy b relay 180 ringing with CSeq: 102 INVITE 9. Asterisk send 200 Ok with CSeq: 103 INVITE and proxy b relay it to proxy a with CSeq: 102 INVITE 10. Now Proxy a send an ACK with CSeq: 102 ACK and proxy b relay ACK to asterisk with CSeq: 102 ACK Now asterisk is retransmitting 200 ok with CSeq: 103 INVITE and proxy a is sending ACK with Cseq: 102 ACK Same happing in case of 487 Request Terminated Here is my config modparam("dialog", "default_timeout", 10800) # 3h modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "db_mode", 0) modparam("dialog", "send_bye", 1) modparam("dialog", "track_cseq_updates", 1) modparam("uac","auth_username_avp","$avp(auser)") modparam("uac","auth_password_avp","$avp(apass)") modparam("uac","auth_realm_avp","$avp(arealm)") modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 3 sec modparam("tm", "fr_timer", 2000) modparam("tm", "retr_timer1", 250) modparam("tm", "retr_timer2", 500) # default invite retransmission timeout after 1xx: 60 sec modparam("tm", "fr_inv_timer", 60000) modparam("tm", "reparse_on_dns_failover", 0) if (is_method("INVITE") && !has_totag()) { dlg_setflag(FLD_RTPPROXY); dlg_manage(); } route(RELAY); route[RELAY] { # At this point all checks are passed t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); t_on_failure("MANAGE_FAILURE"); if(is_method("INVITE") && isflagset(FLT_FROMDID)) { #$fd = "sip.comms.kuflink.com"; t_on_failure("RELAY_AUTH"); } if (!t_relay()) { send_reply("500", "Cannot relay to $du/$ru"); } exit; } failure_route[RELAY_AUTH] { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); if (t_is_canceled()) { exit; } if(t_check_status("401|407")) { #$fd = "sip.comms.kuflink.com"; $avp(auser) = "DID_UA_USERNAME"; $avp(apass) = "DID_UA_PASSWORD"; if (uac_auth()) { t_relay(); } exit; } } route[WITHINDLG] { if (!has_totag()) { return; } if (is_method("BYE")) { rtpengine_delete(); } # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("ACK")) { # ACK is forwarded statelessly route(NATMANAGE); } else if (is_method("NOTIFY")) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if (is_method("ACK") ) { if (t_check_trans()) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } send_reply("404","Not here"); exit; } kamailio -v version: kamailio 5.3.7 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 6.3.0 uname -a Linux example.com 4.15.0-124-generic #127-Ubuntu SMP Fri Nov 6 10:54:43 UTC 2020 x86_64 GNU/Linux SIP logs 2020/12/17 10:08:18.483907 proxy-a.com:5060 -> 10.0.14.118:5060 INVITE sip:123456@proxy-a.com SIP/2.0 Record-Route: <sip:proxy-a.com;lr> Via: SIP/2.0/UDP proxy-a.com:5060 ;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com> Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com User-Agent: ProxyA CSeq: 102 INVITE User-Agent: proxy Date: Thu, 17 Dec 2020 10:08:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "7654321" <sip:7654321@did.com >;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 885 2020/12/17 10:08:18.485063 10.0.14.118:5060 -> proxy-a.com:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP proxy-a.com:5060 ;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1;rport=5060 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: kamailio (5.3.7 (x86_64/linux)) Content-Length: 0 2020/12/17 10:08:18.485576 10.0.14.118:5060 -> 10.0.14.118:6000 INVITE sip:123456@proxy-a.com SIP/2.0 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.0 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com> Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com User-Agent: ProxyA CSeq: 102 INVITE User-Agent: proxy Date: Thu, 17 Dec 2020 10:08:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "7654321" <sip:7654321@did.com >;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 885 2020/12/17 10:08:18.622867 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.0;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as4107645c Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="sip1.comms.kuflink.com", nonce="11139d33" Content-Length: 0 2020/12/17 10:08:18.624034 10.0.14.118:5060 -> 10.0.14.118:6000 ACK sip:123456@proxy-a.com SIP/2.0 Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.0 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as4107645c Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 ACK Content-Length: 0 2020/12/17 10:08:18.625557 10.0.14.118:5060 -> 10.0.14.118:6000 INVITE sip:123456@proxy-a.com SIP/2.0 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com> Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com User-Agent: ProxyA CSeq: 103 INVITE User-Agent: proxy Date: Thu, 17 Dec 2020 10:08:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "7654321" <sip:7654321@did.com >;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 885 Authorization: Digest username="xxxxx", realm="xxxx", nonce="11139d33", uri="sip:123456@proxy-a.com", response="qwewqeqwe", algorithm=MD5 2020/12/17 10:08:18.779483 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 103 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000> Content-Length: 0 2020/12/17 10:08:20.937756 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 103 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000> Content-Length: 0 2020/12/17 10:08:20.938593 10.0.14.118:5060 -> proxy-a.com:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1> Content-Length: 0 2020/12/17 10:08:22.075814 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 103 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000> Content-Type: application/sdp Require: timer Content-Length: 296 2020/12/17 10:08:22.077049 10.0.14.118:5060 -> proxy-a.com:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1> Content-Type: application/sdp Require: timer Content-Length: 316 2020/12/17 10:08:22.103365 proxy-a.com:5060 -> 10.0.14.118:5060 ACK sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1;alias=109.72.216.251~5060~1 SIP/2.0 Via: SIP/2.0/UDP proxy-a.com:5060 ;branch=z9hG4bKbe6d.0fc1940efc770391762d7d67094c24f8.0 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK2a5fae72;rport=5060 Route: <sip:proxy-b.com;lr;did=d18.7a9> Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 ACK User-Agent: proxy Content-Length: 0 2020/12/17 10:08:22.103877 10.0.14.118:5060 -> 10.0.14.118:6000 ACK sip:123456@10.0.14.118:6000;alias=109.72.216.251~5060~1 SIP/2.0 Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.79f351d031912aae585207d2f26d5650.0 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.0fc1940efc770391762d7d67094c24f8.0 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK2a5fae72;rport=5060 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 ACK User-Agent: proxy Content-Length: 0 P-K-CSeq-Refresh: 103 2020/12/17 10:08:22.575533 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 103 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000> Content-Type: application/sdp Require: timer Content-Length: 296 2020/12/17 10:08:22.576777 10.0.14.118:5060 -> proxy-a.com:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1> Content-Type: application/sdp Require: timer Content-Length: 316 2020/12/17 10:08:22.581649 proxy-a.com:5060 -> 10.0.14.118:5060 ACK sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1;alias=109.72.216.251~5060~1 SIP/2.0 Via: SIP/2.0/UDP proxy-a.com:5060 ;branch=z9hG4bKbe6d.b9c7eeccd4cd7b44f05b173308dbaf0a.0 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK219933f6;rport=5060 Route: <sip:proxy-b.com;lr;did=d18.7a9> Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 ACK User-Agent: proxy Content-Length: 0 2020/12/17 10:08:22.582313 10.0.14.118:5060 -> 10.0.14.118:6000 ACK sip:123456@10.0.14.118:6000;alias=109.72.216.251~5060~1 SIP/2.0 Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.33626b5695ad4648ba393bc62bed6a8c.0 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.b9c7eeccd4cd7b44f05b173308dbaf0a.0 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK219933f6;rport=5060 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 ACK User-Agent: proxy Content-Length: 0 P-K-CSeq-Refresh: 103 2020/12/17 10:08:23.575828 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 103 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000> Content-Type: application/sdp Require: timer Content-Length: 296 2020/12/17 10:08:23.577863 10.0.14.118:5060 -> proxy-a.com:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1> Content-Type: application/sdp Require: timer Content-Length: 316

4 7

Kamailio RTP proxy require?
by Mihai Cezar 17 Dec '20

17 Dec '20

Hi, Can't figure it out when to use RTPproxy, i am forwarding everything to asterisk in seem to work without RTPproxy. The scenario is like this: Softphone -> INTERNET -> Kamailio - > Asterisk (private ip) Now it seems to work without any rtpproxy but is it ok?

5 10

DLGLOAD stuck in Dispatcher Module(HA Mode)
by harneet singh 16 Dec '20

16 Dec '20

Hi All, I am using Kamailio in HA mode with Keepalived providing the VIP(Virtual IP) functionality, and have run into a rather peculiar issue. *Setup:* Caller ------------ KamailioVIP(*Primary *and *Secondary* Kamailio)------- Callee HA provided by Keepalived DMQ is used between the Primary and Secondary Kamailio for dialog sync *Issue Seen:* *Suppose the Primary Kamailio has been brought down* and the Secondary one is actually active and tied to the VIP. When a call is fired from the Caller, it traverses through the Secondary Kamailio and lands onto the callee. The dialogs are updated properly. At this point, the Primary Kamailio is brought up, and dialog state is synced due to the DMQ module. The Keepalived will now attach the VIP to the Primary Instance. If the caller hangs up the phone at this point, the BYE sip message traverses through the Primary Kamailio instance to the callee and the call is cleared, but there are two issues here. 1. The Primary Kamailio throws an error in *ds_load_remove() *saying that it cannot find the load for the specific call id. This is apparently due to the fact that the dispatcher data is not synced between the two modules but dialog data is. So dialog wise, things are fine. Can this be fixed somehow? 2. The above is still as grave a problem as the dialogs are cleared. BUT, if we check the '*kamcmd dispatcher.list*' on the *SECONDARY* kamailio even well after the call is cleared, the *DLGLOAD* shows 1. Since we are using the *Call Load based distribution* for the dispatching, this is effectively one call stuck on the dispatcher, which leads to resource leak. Is this a known issue, and if so, do we have a way to circumvent this? Regards, Harneet Singh -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

2 1

INFO: RTC Devroom Fosdem 2021 - Online Event
by Daniel-Constantin Mierla 16 Dec '20

16 Dec '20

Hello, (cross-posting to kamailio public groups, a few may get this message duplicated). Fosdem 2021 is going to be online, meaning that it opens the possibility for everyone to do a presentation on the RTC Devroom, no longer being required to travel to Brussels. Therefore I am sending here the link to CFP in case someone would be interested to present about Kamailio or something else RTC-related: * https://github.com/FreeRTC/fosdem/blob/master/2021/cfp.md New speakers are always welcome! Only have in mind that the event is targeting developers and source code development, therefore your proposals have to be made from this perspective. Note also that submission deadline is 20th of December, very short time frame from now. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

ERROR: dmq [notification_peer.c:337]: add_server_and_notify(): error adding notification node
by Syed Shahryar 16 Dec '20

16 Dec '20

Here is my setting: modparam("dmq", "server_address", "sip:0.0.0.0:5090" ) modparam("dmq", "notification_address", "sip:0.0.0.0:5090") modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 2) modparam("dmq", "worker_usleep", 0) modparam("dmq", "ping_interval", 90) I have tried localhost and I believe other options but to no avail. I also get the following message: ERROR: dmq [dmq_funcs.c:516]: ping_servers(): cannot retrieve initial nodelist from sip:0.0.0.0:5090 How do I set up DMQ for adding and removing any Kamailio instance to/from the cluster and any Kamailio instance could serve any request coming from any client? ie. all Kamailio instances keep the current state in DMQ. I also have dmq_usrloc enabled. -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 1

Kamailio dtmf signaling
by Amit Pal 16 Dec '20

16 Dec '20

Dear Team, I am using kamailio + rtpengine, how can I processed dtmf from cfg scripts. even thought I can able to see dtmf over syslog. but can not able to capture in kamailio.cfg. how can I prepared ivr using kamalio? Please find the attached syslog. Regards Amit Pal

3 2

ERROR: <core> [core/parser/parse_uri.c:1326]: _parse_ruri(): bad uri </sip?jwt=..
by Syed Shahryar 15 Dec '20

15 Dec '20

I am sending JWT tokens via SIP URL through JSSIP for authentication before websocket connection is made. However, I am getting errors like: ERROR: <core> [core/parser/parse_uri.c:1326]: _parse_ruri(): bad uri < */sip?jwt=..* I have the following setting in kamailio config: modparam("xhttp", "url_match", "^/sip") // only match this path Is this a bug or some configuration issue? How do I resolve it? -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 1

X-Forwarded-for IP to asterisk
by Mihai Cezar 15 Dec '20

15 Dec '20

Hello, Is it something like the XFF for Kamailio so I can see in asterisk (sip show peers) the ip of the registered softphone and not Kamailio's IP ? Thanks!

4 5

How to use port number using $env in listen
by Syed Shahryar 15 Dec '20

15 Dec '20

I tried many combinations of the following but I keep getting syntax errors: listen=tls:TLS_IP:$env(TLS_PORT) advertise PUBLIC_IP:$env(TLS_PORT) TLS_IP and PUBLIC_IP are constants defined at the top using #!define. And, TLS_PORT is supposed to be the env variable. -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 1

Losing MySQL connections
by Duarte Rocha 15 Dec '20

15 Dec '20

Greetings all, I'm having some dropped MySQL connections in my Kamailio in times of low or no activity in the proxy. These are the errors i'm getting : db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server during query (2013) <core> [db_query.c:348]: db_do_update(): error while submitting query usrloc [ucontact.c:1147]: db_update_ucontact_ruid(): updating database failed usrloc [ucontact.c:1663]: update_contact_db(): failed to update database registrar [save.c:784]: update_contacts(): failed to update contact sl [sl_funcs.c:414]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) In order to try to fix this i've added timeout_interval and ping_interval from the db_mysql module. Is this the best approach? I've restarted my Kamailio but i don't see any ping going to my BD. Is there any other config that I need to add so that the pings are sent? Is it possible to do a ping to verify connection and reconnect mannualy on the config file? Best Regards,

2 1

Problems debugging memory issues
by Sheldon L 15 Dec '20

15 Dec '20

Hi, I'm trying to investigate a possible memory leak issue in Kamailio and I'm following the instructions indicated in https://www.kamailio.org/wiki/tutorials/troubleshooting/memory I'm using Kamailo 4.4 with the following kamailio internals: Print out of kamailio internals Version: kamailio 4.4.2 (x86_64/linux) 892ad6 Default config: /apollo/env/TachyonPlatformKamailio/etc/kamailio/kamailio.cfg Default paths to modules: /apollo/env/TachyonPlatformKamailio/lib64/kamailio/modules Compile flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES MAX_RECV_BUFFER_SIZE=262144 MAX_LISTEN=16 MAX_URI_SIZE=1024 BUF_SIZE=65535 DEFAULT PKG_SIZE=8MB DEFAULT SHM_SIZE=64MB ADAPTIVE_WAIT_LOOPS=1024 TCP poll methods: poll, epoll_lt, epoll_et, sigio_rt, select Source code revision ID: 892ad6 Compiled with: x86_64-unknown-linux-gnu-gcc 4.9.4 Compiled on: 23:36:48 Dec 8 2020 Thank you for flying kamailio! I have attached to one of the kamailio tcp receiver processes via GDB and I am trying to dump memory blocks from the memory manager. However, mem_block is not in the current_context. The following is the output from my gdb session: Copyright (C) 2013 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html > This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-amazon-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... Reading symbols from /opt/codedeploy-agent/deployment-root/95bb1634-e653-4cec-b327-99b18b9227a8/d-7AJ1XZP58/deployment-archive/sbin/kamailio...done. Attaching to program: /apollo/env/TachyonPlatformKamailio/sbin/kamailio, process 8946 ptrace: Operation not permitted. (gdb) set $i=0 (gdb) set $a = mem_block->first_frag No symbol "mem_block" in current context. (gdb) info list Undefined info command: "list". Try "help info". (gdb) info threads No threads. Is there something I need to do before accessing the mem_block symbol? Also do you have any other suggestions for debugging memory leaks that may not be documented? Note: i can't upgrade to the latest version of Kamailio as we have custom modules and would require considerable effort to upgrade. Thanks, Sheldon.

2 1

sngrep TLS connection
by Agiftel 15 Dec '20

15 Dec '20

Hi all, does anyone was able to use sngrep (with -k option) to decrypt TLS connections? I'am doing what help says; used private key is the correct one; but no traffic is seen on sngrep. If i use not ecrypted call i can see everything, so sngrep is working good. Any hint? regards -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

5 8

Kamilio+asterisk = deregister after 2-3 minute
by Mihai 15 Dec '20

15 Dec '20

Hi, I am forwarding all the traffic to my asterisk server ( including registers) everything seems to work when i first register but if stay idle 2-3 minute it gets unregistered and then i can't register back unless i restart kamailio see attached print screen

6 6

voltdb support
by Ali Taher 14 Dec '20

14 Dec '20

Hello, I just need to know if kamailio can communicate with Voltdb for sqlops operations for example. Regards, Ali Taher

2 8

Kamailio v5.4.3 Released
by Daniel-Constantin Mierla 14 Dec '20

14 Dec '20

Hello, Kamailio SIP Server v5.4.3 stable release is out. This is a maintenance release of the latest stable branch, 5.4, that includes fixes since the release of v5.4.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.4.x. Deployments running previous v5.4.x versions are strongly recommended to be upgraded to v5.4.3. For more details about version 5.4.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/12/kamailio-v5-4-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Wrong To tag in t_reply
by Ivan Ribakov 14 Dec '20

14 Dec '20

Hi all, I’m trying to implement following routing behaviour: - ingress INVITE is sequentially forked to 1 or more endpoints - under certain conditions sequential forking may be stopped and pre-defined error returned to the caller Here’s is part of the config responsible for handling errors: failure_route[ROUTE_ELEMENT_FINAL_FAILURE_HELPER] { route(ROUTE_ELEMENT_FINAL_FAILURE); } route[ROUTE_ELEMENT_FINAL_FAILURE] { # We need to check if the error happens before the first egress, because T-state is not defined yet so we can not use t_reply if ($var(before_first_egress) == 0) { t_reply("$var(sipCode)", "$var(sipMessage)"); } else { sl_send_reply("$var(sipCode)", "$var(sipMessage)"); } } Here is a simple scenario where delivering call to first endpoint times out after receiving 180 Ringing (using custom fr_inv_timer value) where: - .33 is caller - .208 is Kamailio - .138 is callee Notice that 504 is retransmitted by Kamailio. Here is a comparison of 180 and 504 returned by Kamailio: As can be seen, 180 was proxied from the callee and has callee’s to-tag (tag=CALLEE1), however 504 that was generated by Kamailio using t_reply() has a different to-tag (presumably generated by Kamailio). How can I tell Kamailio not to generate new to-tag but to use the last one that proxied back to the caller? Thanks in advance for any help! Regards, Ivan

2 2

Releasing Kamailio v5.4.3
by Daniel-Constantin Mierla 14 Dec '20

14 Dec '20

Hello, I am considering to release Kamailio v5.4.3 sometime next week, likely on Monday, December 14, 2020. This is the usual heads up notification to see if anyone is aware of issues not yet reported to bug tracker and if yes, do it as soon as possible to give them a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 1

how to use environment variables to set listen address
by Syed Shahryar 13 Dec '20

13 Dec '20

This is a generic question about using environment variables in a config file. If I want to use below where TLS_IP itself is a constant defined somewhere else. TLS_IP might be set via an env variable as well. #!define TLS_TO_LISTEN tls:TLS_IP:$env(TLS_PORT) Will that work? I see syntax errors and other weird issues. What's the right syntax? -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 1

DMQ using Consul
by Syed Shahryar 12 Dec '20

12 Dec '20

Can someone share their configuration file explaining how they used consul to achieve high availability? -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 1

empty body on first NOTIFY on REGINFO subscription - bug ?
by Adrian Tabacioiu 11 Dec '20

11 Dec '20

Hello, If the user never registered (or few hours passed from its unregister) the subscription to REGINFO will return an empty body NOTIFY (no xml). I don't see any flag in module to force a body for this case, as in the presence module is force_dummy_presence who can force a body on a first NOTIFY. The RFC3680 says : o Notifications triggered from a SUBSCRIBE contain full state (the list of all contacts bound to the address-of-record). It appears that in this case the response is not conform to RFC. Regards, Adrian

1 0

porting keepalive module new functions on 5.4 on 5.3?
by Benoit Chesneau 11 Dec '20

11 Dec '20

2 new functions have been added to the keepalive module: - [ka_add_destination(sip_uri, owner)](https://kamailio.org/docs/modules/5.4.x/modules/keepalive.html#keep…) - [ka_del_destination(sip_uri, owner)](https://kamailio.org/docs/modules/5.4.x/modules/keepalive.html#keep…) Is there any possibility to do the same on 5.3? It seems that on 5.3 adding a destination can only be done statically: https://kamailio.org/docs/modules/5.3.x/modules/keepalive.html#idm53 or can it also be done dynamically when receiving an event using the kazoo module for example? Benoît Chesneau, Enki Multimedia — t. +33608655490 Sent with [ProtonMail](https://protonmail.com) Secure Email.

2 1

Dispatcher algorithm 11 possible priority drifting
by Володимир Іванець 10 Dec '20

10 Dec '20

Hello all! We are running Kamailio version 5.3.3 with 16 small Asterisk servers. Kamailio uses a dispatcher module to distribute calls. Algorithm #11 is selected. All destinations are configured with priority set to 50 and attribute to rweight=50. We noticed that one server constantly receives more calls than others. I run a few tests. I was sending 1000 calls to the system. 4 per second. All servers except one were getting around 60 calls while the last one - around 100. I then noticed that the server which receives most calls is always last in the "kamcmd dispatcher.list" command output. I then changed the order in the dispatcher DB table and repeat the test. The other server that now was last was getting the most calls. Does anyone else use algorithm #11 and finds the same thing? Is there something additional that I can provide to help with the investigation? Thanks!

3 3

kamailio.cfg fomatter
by Juha Heinanen 10 Dec '20

10 Dec '20

After noticing that emacs mode is missing, below is my first cut of kamailio-mode.el. -- Juha ----------------------------------------------------------------------- ;; kamailio mode for emacs (function list is not complete) ;; activate by starting your kamailio.cfg with this kind of line: ;; #-*-mode: kamailio; tab-width: 4; c-basic-offset: 4; (setq kamailio-font-lock-keywords (let* ( ;; define several category of keywords (x-keywords '("branch_route" "break" "case" "default" "else" "event_route" "exit" "failure_route" "for" "if" "loadmodule" "onreply_route" "onsend_route" "reply_route" "request_route" "return" "route" "switch" "while")) (x-types '()) (x-constants '()) (x-events '()) (x-functions '("acc_db_request" "add_path_received" "add_rr_param" "allow_routing" "allow_trusted" "append_hf" "append_to_reply" "avp_check" "avp_delete" "change_reply_status" "check_route_param" "defined" "dp_translate" "ERROR" "fix_nated_sdp" "force_send_socket" "forward" "get_redirects" "handle_ruri_alias" "inactivate_gw" "INFO" "in_list" "isflagset" "is_avp_set" "isbflagset" "is_direction" "is_domain_local" "is_e164" "is_gflag" "is_gruu" "is_method" "is_numeric" "is_present_hf" "jsonrpc_dispatch" "load_gws" "lookup_domain" "lookup" "loose_route" "modparam" "mt_match" "NOTICE" "proxy_challenge" "record_route" "redirecting" "remove_hf" "resetflag" "rtpengine_answer" "rtpengine_delete" "rtpengine_offer" "save" "sdp_content" "sdp_transport" "send_reply_mode" "send_reply" "setbflag" "setflag" "setxflag" "set_reply_close" "set_reply_no_connect" "set_rtpengine_set" "set_uri_host" "set_uri_user" "set_forward_no_connect" "sql_query" "sql_result_free" "starts_with" "subst_uri" "tel2sip" "t_check_status" "t_is_canceled" "t_load_contacts" "t_next_contacts" "t_on_branch" "t_on_failure" "t_on_reply" "t_relay_cancel" "t_relay" "t_set_fr" "unregister" "WARN" "ws_handle_handshake" "www_challenge" "xhttp_reply")) (x-keywords-regexp (regexp-opt x-keywords 'words)) (x-types-regexp (regexp-opt x-types 'words)) (x-constants-regexp (regexp-opt x-constants 'words)) (x-events-regexp (regexp-opt x-events 'words)) (x-functions-regexp (regexp-opt x-functions 'words))) `( (,x-types-regexp . font-lock-type-face) (,x-constants-regexp . font-lock-constant-face) (,x-events-regexp . font-lock-builtin-face) (,x-functions-regexp . font-lock-function-name-face) (,x-keywords-regexp . font-lock-keyword-face) ))) (defvar kamailio-mode-syntax-table nil "Syntax table for `kamailio-mode'.") (setq kamailio-mode-syntax-table (let ( (synTable (make-syntax-table))) ;; python style comment: “# …” (modify-syntax-entry ?# "<" synTable) (modify-syntax-entry ?\n ">" synTable) synTable)) (define-derived-mode kamailio-mode c-mode "kamailio mode" "Major mode for editing Kamailio configuration files" (setq font-lock-defaults '((kamailio-font-lock-keywords))) (set-syntax-table kamailio-mode-syntax-table)) (provide 'kamailio-mode)

2 5

Re: [SR-Users] Dialog profile not defined
by Gertjan Wolzak 10 Dec '20

10 Dec '20

Hello Kamailions, Well, I think I found the cause of my issue where the Dialog profile was not defined.. Well it was. I had not enabled antiflood, by not enabling antiflood, the htable module was also not loaded... I guess the us of profiles_with_value parameter needs htable? Is that correct? If so, that means the htable is a dependency for the dialog module...., then it would be sometihng that could be added to the documentation of the dialog module. When this is not the case, then I have to dig deeper, as I do not get an error anymore about the profile not being known... Thank you. Gertjan

2 2

REGINFO events not sending
by Adrian Tabacioiu 10 Dec '20

10 Dec '20

Hello, Does anybody has used with success the reginfo modules ? I added lines bellow in configuration, and tried different variations, Even if my subscribe for "reginfo" events is accepted (with 202 OK), I receive no notification when respective user registers. Nor do I get the xml body in the first notification. Added in my config: loadmodule "presence_reginfo.so" loadmodule "pua_reginfo.so" modparam("pua_reginfo", "default_domain", "192.168.60.65") modparam("pua_reginfo", "server_address", "sip:reginfo@192.168.60.65") modparam("pua_reginfo", "publish_reginfo", 1) Thank you in advance, Adrian

2 3

SLA presence not working proper with (SCA module)
by Aatif Shaikh 10 Dec '20

10 Dec '20

Aatif Shaikh <aatif.shaikh(a)ecosmob.com> Mon, Dec 7, 11:00 AM (3 days ago) to sr-users Hello All I configure my kamailio.cfg to handle SLA using (SCA module) and shared presence is working but still facing some issue in case of hold invite and BYE and the reason behind of that issue is following error Nov 26 07:38:40 SBC-4-1 /usr/local/sbin/kamailio[104107]: ERROR: sca [sca_call_info.c:1008]: sca_call_info_invite_request_handler(): Failed to update sip:4569@test.sip.abcd.com appearance-index 0 to active actually, in Kamailio appearances, I can see the entry for this call but the index is 1 instead of 0, and Kamailio going to update with a 0 index [root@SBC-4-1 ntcarfte-kamailio]# kamcmd sca.all_appearances sip:4569@test.sip.abcd.com 1 active 1606376289 sip:4569@10.xx.xx.xx :5070;transport=udp sip:2003@test.sip.abcd.com 186161_mobile-rel120MTQ5OTcyZjFhMjMyNmI1ZGE1MWY4ODc2M2RkN2VmZmQ 45dc18d8 H6cS9vecZ248B if I manually update appearances using RPC command with index 1 then the phone started blinking as I change state to held # kamcmd sca.update_appearance sip:4569@sc4test.sip.teledge.com 1 held # kamcmd sca.all_appearances sip:4569@test.sip.abcd.com 1 held 1606376516 sip:4569@10.xx.xx.xx :5070;transport=udp sip:2003@test.sip.abcd.com 186161_mobile-rel120NzE5MDgyZDc1NjJiYjcwMWFlYmI3NzM3NjE2OTRhZjU 08f005cb eK9ratm83g92e kamailio.cfg route[SCA] { if(is_method("SUBSCRIBE")) { if ($hdr(Event) == "call-info" || $hdr(Event) == "line-seize") { xlog("L_INFO","(MAIN) :HELLO ($avp(uuid)) : $avp(rsi) $sp $hdr(Event)"); xdbg("SCA: $hdr(Event) SUBSCRIBE $ru from $si:$sp"); sca_handle_subscribe(); exit; } return; } if (!is_method("BYE|CANCEL|INVITE|PRACK|REFER")) { return; } sca_call_info_update(); } route[RELAY_OUTBOUND_FS] { xlog("L_INFO","(RELAY_OUTBOUND_FS) : ($avp(uuid)) : INSIDE ROUTE "); if (!has_totag()){ $avp(set_h) = 1; } t_on_reply("REPLY_OUTBOUND_FS"); route(SCA); if (!t_relay()) { sl_reply_error(); } exit; } route[RELAY_INBOUND_FS] { if(is_method("INVITE|BYE|UPDATE|CANCEL|ACK")) { $avp(s:puburis_caller) = $fu; setflag(8); dlg_manage(); } if(is_method("REGISTER")) { t_on_reply("REPLY_REGISTER"); } else { t_on_reply("REPLY_INBOUND_FS"); } route(SCA); if (!t_relay()) { sl_reply_error(); } exit; } onreply_route[REPLY_INBOUND_FS] { xdbg("incoming reply\n"); if (status =~ "[456][0-9][0-9]") { # don't update SCA state here, since there may be # failure route processing (e.g., call forwarding). # update state in failure route instead. return; } route(SCA); } onreply_route[REPLY_OUTBOUND_FS] { xdbg("incoming reply\n"); if (status =~ "[456][0-9][0-9]") { # don't update SCA state here, since there may be # failure route processing (e.g., call forwarding). # update state in failure route instead. return; } route(SCA); } can someone help me with this Did I missed something to handle SLA?

1 0

NAT problem with recvonly calls
by David Cunningham 10 Dec '20

10 Dec '20

Hello, We have a problem with a SIP doorbell device which sends media one way only, and NAT at the receiving device. When the doorbell button is pressed it makes a call to a configured destination. Since the doorbell only sends and doesn't receive it sends the INVITE with sendonly in the SDP, and the destination then replies with a 200 OK with recvonly in the SDP. The problem is that the destination is behind NAT, and its reply contains a private network IP in the SDP. Normally Asterisk when nat=yes works around that by adjusting the destination for RTP to be the address it actually receives audio from, however because this device is recvonly Asterisk never receives audio from it. This means Asterisk keeps trying to send the doorbell's RTP to the private network IP which of course fails, and the destination never gets the RTP from the doorbell. We haven't found a solution in Asterisk to this, so are now looking to Kamailio which acts as a load-balancing proxy in front of Asterisk for one. For example, maybe we could use fix_nated_sdp, but only on 200 OK's with recvonly. Has anyone else encountered this, and are there any recommended solutions? Thank you in advance! -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

4 8

Re: [SR-Users] NAT problem with recvonly calls
by Ovidiu Sas 09 Dec '20

09 Dec '20

The doorbell is the caller. ICE support would be required on the callee side (between NATed callee and asterisk). There's no need for ICE support on the doorbell/caller side. -ovidiu On Wed, Dec 9, 2020 at 2:42 AM Alex Balashov <abalashov(a)evaristesys.com> wrote: > > ICE support just doesn't seem to me to be something a SIP doorbell would > have. > > On 12/9/20 2:39 AM, Olle E. Johansson wrote: > > > > > >> On 8 Dec 2020, at 18:55, Richard Fuchs <rfuchs(a)sipwise.com > >> <mailto:rfuchs@sipwise.com>> wrote: > >> > >> Use IPv6 > > > > While I like that proposal, assuming that IPv6 will not have a stateful > > firewall is propably not a good assumption. > > So I suspect that an IPv6 firewall would not let the packets in if there > > is not outbound traffic first. I do like Ovidius > > ICE-based solution and would like to add RTCP - if muxed it would open > > the port, but the likelyhood that it’s > > implemented is propably low. > > > > IPv6 greetings! > > /O :-) > >> > >> Cheers > >> > >> On 07/12/2020 23.01, David Cunningham wrote: > >>> Hello, > >>> > >>> We have a problem with a SIP doorbell device which sends media one > >>> way only, and NAT at the receiving device. > >>> > >>> When the doorbell button is pressed it makes a call to a configured > >>> destination. Since the doorbell only sends and doesn't receive it > >>> sends the INVITE with sendonly in the SDP, and the destination then > >>> replies with a 200 OK with recvonly in the SDP. The problem is that > >>> the destination is behind NAT, and its reply contains a private > >>> network IP in the SDP. > >>> > >>> Normally Asterisk when nat=yes works around that by adjusting the > >>> destination for RTP to be the address it actually receives audio > >>> from, however because this device is recvonly Asterisk never receives > >>> audio from it. This means Asterisk keeps trying to send the > >>> doorbell's RTP to the private network IP which of course fails, and > >>> the destination never gets the RTP from the doorbell. > >>> > >>> We haven't found a solution in Asterisk to this, so are now looking > >>> to Kamailio which acts as a load-balancing proxy in front of Asterisk > >>> for one. For example, maybe we could use fix_nated_sdp, but only on > >>> 200 OK's with recvonly. > >>> > >>> Has anyone else encountered this, and are there any recommended > >>> solutions? > >>> > >>> Thank you in advance! > >>> > >>> -- > >>> David Cunningham, Voisonics Limited > >>> http://voisonics.com/ <http://voisonics.com/> > >>> USA: +1 213 221 1092 > >>> New Zealand: +64 (0)28 2558 3782 > >>> > >>> _______________________________________________ > >>> Kamailio (SER) - Users Mailing List > >>> sr-users(a)lists.kamailio.org > >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > >> _______________________________________________ > >> Kamailio (SER) - Users Mailing List > >> sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org> > >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > -- > Alex Balashov | Principal | Evariste Systems LLC > > Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) > Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- VoIP Embedded, Inc. http://www.voipembedded.com

1 0

GitHub discussion
by Sergey Safarov 09 Dec '20

09 Dec '20

GitHub release new feature - discussion. Example https://github.com/vercel/vercel/discussions

2 1

Kamailio as an LCR based on prices but also signaling quality indicators
by Abdoul Osséni 08 Dec '20

08 Dec '20

Hello, I am trying to configure kamailio as an LCR based on prices but also signaling quality indicators like ASR and ACDR. Can you please help me and tell me where to start? Regards Abdoul

4 3

kamailio-5.2.0 Crash - udp_send failed
by Hamid Hashmi 07 Dec '20

07 Dec '20

I am using "kamailio 5.2.0-dev6 (x86_64/linux) 48ed62-dirty" what could be the reason of the crash its usually crash with the following logs Sep 25 16:58:41 pcscf [Proxy-CSCF][20771]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36eee8c560,418,0,10.100.227.202:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20771]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20771]: ERROR: tm [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20771]: WARNING: tm [t_reply.c:1581]: t_retransmit_reply(): send pr buffer failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20756]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8eb948,358,0,10.100.177.91:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20756]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20756]: ERROR: sl [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20744]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8b4a80,358,0,10.100.72.230:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20744]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20744]: ERROR: sl [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20783]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8ac740,416,0,10.100.72.230:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20783]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20783]: ERROR: tm [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20752]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8b5098,356,0,10.100.56.38:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20805]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8e8f18,416,0,10.100.177.91:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20805]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20805]: ERROR: tm [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20752]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20752]: ERROR: sl [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20777]: ERROR: <script>: Re-Transmit: REGISTER (sip:410060269353579@ims.mnc006.mcc410.3gppnetwork.org (10.100.236.57:5060) to sip:410060269353579@ims.mnc006.mcc410.3gppnetwork.org, 1SDq5cuY3iDYFKLATTp) Sep 25 16:58:41 pcscf [Proxy-CSCF][20761]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8e8a08,354,0,10.100.2.76:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20761]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20761]: ERROR: sl [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20767]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8ad100,360,0,10.100.144.137:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20767]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20767]: ERROR: sl [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20777]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8eb860,360,0,10.100.140.226:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20777]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20777]: ERROR: sl [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:43 pcscf [Proxy-CSCF][20861]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 77 Sep 25 16:58:44 pcscf [Proxy-CSCF][20853]: ERROR: <core> [core/tcp_read.c:301]: tcp_read_data(): error reading: Connection timed out (110) ([10.100.64.49]:32888 -> Sep 25 16:58:44 pcscf [Proxy-CSCF][20853]: ERROR: <core> [core/tcp_read.c:302]: tcp_read_data(): -> [10.161.224.228]:5060) Sep 25 16:58:44 pcscf [Proxy-CSCF][20853]: ERROR: <core> [core/tcp_read.c:1485]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f36a739fe78 r: 0x7f36a739fef8 Sep 25 16:58:44 pcscf [Proxy-CSCF][20742]: ALERT: <core> [main.c:748]: handle_sigs(): child process 20815 exited by a signal 11 Sep 25 16:58:44 pcscf [Proxy-CSCF][20742]: ALERT: <core> [main.c:751]: handle_sigs(): core was generated Sep 25 16:58:44 pcscf kernel: [52607827.581845] net_ratelimit: 16 callbacks suppressed Sep 25 16:58:44 pcscf [Proxy-CSCF][20742]: CRITICAL: cdp [diameter_peer.c:423]: diameter_peer_destroy(): destroy_diameter_peer(): Bye Bye from C Diameter Peer test Sep 25 16:59:01 pcscf CRON[24805]: (root) CMD (/bin/bash /root/volumes/monitoring/src/services_status.sh -p http://10.161.224.25:9091 -s kamailio_pcscf,sems -n pcscf) Sep 25 16:59:01 pcscf CRON[24804]: (CRON) info (No MTA installed, discarding output) Sep 25 16:59:44 pcscf [Proxy-CSCF][20742]: CRITICAL: <core> [main.c:654]: sig_alarm_abort(): shutdown timeout triggered, dying... but last time its also crashed with the following logs as well Sep 25 18:39:59 pcscf [Proxy-CSCF][28015]: ERROR: <core> [core/tcp_read.c:301]: tcp_read_data(): error reading: Connection timed out (110) ([10.100.183.34]:39022 -> Sep 25 18:39:59 pcscf [Proxy-CSCF][28015]: ERROR: <core> [core/tcp_read.c:302]: tcp_read_data(): -> [10.161.224.228]:5060) Sep 25 18:39:59 pcscf [Proxy-CSCF][28015]: ERROR: <core> [core/tcp_read.c:1485]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f92ca8d0150 r: 0x7f92ca8d01d0 Sep 25 18:39:59 pcscf [Proxy-CSCF][28010]: ERROR: cdp [authstatemachine.c:404]: auth_client_statefull_sm_process(): auth_client_statefull_sm_process(): Received invalid event 6 while in state Open! Sep 25 18:39:59 pcscf [Proxy-CSCF][28010]: ERROR: cdp [authstatemachine.c:404]: auth_client_statefull_sm_process(): auth_client_statefull_sm_process(): Received invalid event 6 while in state Open! Sep 25 18:40:00 pcscf [Proxy-CSCF][28034]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 77 Sep 25 18:40:01 pcscf [Proxy-CSCF][28026]: ERROR: <core> [core/tcp_read.c:301]: tcp_read_data(): error reading: Connection timed out (110) ([10.100.100.218]:1337 -> Sep 25 18:40:01 pcscf [Proxy-CSCF][28026]: ERROR: <core> [core/tcp_read.c:302]: tcp_read_data(): -> [10.161.224.228]:5060) Sep 25 18:40:01 pcscf [Proxy-CSCF][28026]: ERROR: <core> [core/tcp_read.c:1485]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f92ca87f070 r: 0x7f92ca87f0f0 Sep 25 18:40:01 pcscf [Proxy-CSCF][27919]: ALERT: <core> [main.c:748]: handle_sigs(): child process 27989 exited by a signal 11 Sep 25 18:40:01 pcscf [Proxy-CSCF][27919]: ALERT: <core> [main.c:751]: handle_sigs(): core was generated Sep 25 18:40:01 pcscf kernel: [52613904.372973] net_ratelimit: 29 callbacks suppressed Sep 25 18:40:01 pcscf [Proxy-CSCF][27919]: CRITICAL: cdp [diameter_peer.c:423]: diameter_peer_destroy(): destroy_diameter_peer(): Bye Bye from C Diameter Peer test Sep 25 18:40:01 pcscf CRON[30581]: (root) CMD (/bin/bash /root/volumes/monitoring/src/services_status.sh -p http://10.161.224.25:9091 -s kamailio_pcscf,sems -n pcscf) Sep 25 18:40:01 pcscf CRON[30580]: (CRON) info (No MTA installed, discarding output) Sep 25 18:40:02 pcscf sems[2858]: [#7f2080044700] [update_uac_request, trans_layer.cpp:2080] WARNING: While sending 200 ACK: no matching transaction Sep 25 18:41:01 pcscf [Proxy-CSCF][27919]: CRITICAL: <core> [main.c:654]: sig_alarm_abort(): shutdown timeout triggered, dying... PS: Kamailio is not listening on localhost. Regards Hamid R. Hashmi

3 4

Kamilio dispatcher not registering with asterisk
by Mihai 07 Dec '20

07 Dec '20

Hi all, i have a problem with kamailio 5.3.x and asterisk 13, i basically forward everything, but i cant manage to make it register. The register arrives on asterisk see the capture bellow: Any help is much appreciated! ---------------- REGISTER sip:oro.sips.nl;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 104.201.100.15;branch=z9hG4bK4d5e.997621d931a2ad9f6423e2ac7782a319.1 Via: SIP/2.0/UDP 34.12.55.102:34626 ;received=34.12.55.102;branch=z9hG4bK-524287-1---0a0bffd64bbd9941;rport=34626 Max-Forwards: 69 Contact: <sip:test-user1@34.12.55.102:34626 ;transport=UDP;rinstance=85c33e976be3dad1> To: <sip:test-user1@oro.sips.nl;transport=UDP> From: <sip:test-user1@oro.sips.nl;transport=UDP>;tag=6d8fd914 Call-ID: h6lkViFgxurpVRkw4Ly-GQ.. CSeq: 1 REGISTER Expires: 60 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Z 5.4.9 rv2.10.11.7 Allow-Events: presence, kpml, talk Content-Length: 0 Path: <sip:104.201.100.15;lr;received=sip:34.12.55.102:34626> -------------------the config ------- # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_FS 10 ####### Global Parameters ######### debug=4 log_stderror=yes 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="mysipserver.com" 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:127.0.0.1:5060 sip_warning=no ####### Modules Section ######## # set module path #mpath="/usr/local/lib/kamailio/modules/" #loadmodule "db_mysql.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.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 "acc.so" loadmodule "dispatcher.so" loadmodule "path.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) # ----- 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", FLT_ACC) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) 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", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") modparam("dispatcher", "ds_ping_from", "sip:proxy@mykamailio") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_threshold", 10) modparam("dispatcher", "ds_inactive_threshold", 5) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=488;class=3") modparam("dispatcher", "ds_probing_mode", 1) modparam("path", "use_received", 1) ####### Routing Logic ######## # main request routing logic request_route { # per request initial checks route(REQINIT); if(ds_is_from_list()) { setflag(FLT_FS); } # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); if (isflagset(FLT_FS)) { route(FROM_FS); exit; } ### only initial requests (no To tag) # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # 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[FROM_FS] { record_route(); route(RELAY); exit; } 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] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... 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 e.g. 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; add_path_received(); route(DISPATCH); } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; sl_send_reply("404", "Not here"); exit; } # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' switch ($fd){ case "oro.sips.nl": send_reply(483, "Ok, Skipper!"); if (!ds_select_dst("1", "4")) { log("ERROR: no active destinations found!\n"); send_reply(503, "Service Unavailable"); exit; } ds_select_dst("1", "4"); log("---- flag value is $dd -- branch $fd \n"); xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); break; case "oro2.sips.nl": send_reply(483, "Ok, Skipper!"); if (!ds_select_dst("2", "4")) { log("ERROR: no active destinations found!\n"); send_reply(503, "Service Unavailable"); exit; } ds_select_dst("2", "4"); log("---- flag value is $dd -- branch $fd \n"); xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); break; default: log("unknow destination?"); send_reply(503, "No service defined"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } }

3 4

SLA presence not working proper with (SCA module)
by Aatif Shaikh 07 Dec '20

07 Dec '20

Hello All I configure my kamailio.cfg to handle SLA using (SCA module) and shared presence is working but still facing some issue in case of hold invite and BYE and the reason behind of that issue is following error Nov 26 07:38:40 SBC-4-1 /usr/local/sbin/kamailio[104107]: ERROR: sca [sca_call_info.c:1008]: sca_call_info_invite_request_handler(): Failed to update sip:4569@test.sip.abcd.com appearance-index 0 to active actually, in Kamailio appearances, I can see the entry for this call but the index is 1 instead of 0, and Kamailio going to update with a 0 index [root@SBC-4-1 ntcarfte-kamailio]# kamcmd sca.all_appearances sip:4569@test.sip.abcd.com 1 active 1606376289 sip:4569@10.xx.xx.xx:5070;transport=udp sip:2003@test.sip.abcd.com 186161_mobile-rel120MTQ5OTcyZjFhMjMyNmI1ZGE1MWY4ODc2M2RkN2VmZmQ 45dc18d8 H6cS9vecZ248B if I manually update appearances using RPC command with index 1 then the phone started blinking as I change state to held # kamcmd sca.update_appearance sip:4569@sc4test.sip.teledge.com 1 held # kamcmd sca.all_appearances sip:4569@test.sip.abcd.com 1 held 1606376516 sip:4569@10.xx.xx.xx:5070;transport=udp sip:2003@test.sip.abcd.com 186161_mobile-rel120NzE5MDgyZDc1NjJiYjcwMWFlYmI3NzM3NjE2OTRhZjU 08f005cb eK9ratm83g92e kamailio.cfg route[SCA] { if(is_method("SUBSCRIBE")) { if ($hdr(Event) == "call-info" || $hdr(Event) == "line-seize") { xlog("L_INFO","(MAIN) :HELLO ($avp(uuid)) : $avp(rsi) $sp $hdr(Event)"); xdbg("SCA: $hdr(Event) SUBSCRIBE $ru from $si:$sp"); sca_handle_subscribe(); exit; } return; } if (!is_method("BYE|CANCEL|INVITE|PRACK|REFER")) { return; } sca_call_info_update(); } route[RELAY_OUTBOUND_FS] { xlog("L_INFO","(RELAY_OUTBOUND_FS) : ($avp(uuid)) : INSIDE ROUTE "); if (!has_totag()){ $avp(set_h) = 1; } t_on_reply("REPLY_OUTBOUND_FS"); route(SCA); if (!t_relay()) { sl_reply_error(); } exit; } route[RELAY_INBOUND_FS] { if(is_method("INVITE|BYE|UPDATE|CANCEL|ACK")) { $avp(s:puburis_caller) = $fu; setflag(8); dlg_manage(); } if(is_method("REGISTER")) { t_on_reply("REPLY_REGISTER"); } else { t_on_reply("REPLY_INBOUND_FS"); } route(SCA); if (!t_relay()) { sl_reply_error(); } exit; } onreply_route[REPLY_INBOUND_FS] { xdbg("incoming reply\n"); if (status =~ "[456][0-9][0-9]") { # don't update SCA state here, since there may be # failure route processing (e.g., call forwarding). # update state in failure route instead. return; } route(SCA); } onreply_route[REPLY_OUTBOUND_FS] { xdbg("incoming reply\n"); if (status =~ "[456][0-9][0-9]") { # don't update SCA state here, since there may be # failure route processing (e.g., call forwarding). # update state in failure route instead. return; } route(SCA); } can someone help me with this Did I missed something to handle SLA? -- -- Regards, *Aatif Shaikh *| Software Developer +91 9033976268Hangout: aatif.shaikh(a)ecosmob.comSkype : aatif.shaikh(a)ecosmob.com *Ecosmob Technologies Pvt. 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1 0

Re: [SR-Users] Dialog profile not defined
by Gertjan Wolzak 06 Dec '20

06 Dec '20

Thank you Joel for the tip. But that is also not it..... I tried of course, but when you look at the Dialog module documentation is says: modparam("dialog", "profiles_with_value", "caller ; my_profile") So there are spaces in the modparam config for the dialog profiles. Its just so strange that when starting Kamailio I get the error that the Profile outbound is not configured, while it is... Log line: Dec 6 15:04:04 proxy01 /usr/local/sbin/kamailio[76664]: CRITICAL: dialog [dialog.c:391]: fixup_profile(): profile <inbound> not defined kamailo.cfg # ---- Dialog params ------------- modparam("dialog", "default_timeout", 7200) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", DLG_FLAG) modparam("dialog", "hash_size", 4096) #modparam("dialog", "detect_spirals", 1) modparam("dialog", "profiles_with_value", "inbound") Im starting to think that I am missing some configuration in the kamailio.cfg file... Rgds, Gertjan

1 0

Re: [SR-Users] Dialog profile not defined
by Gertjan Wolzak 05 Dec '20

05 Dec '20

Thanks again Sergiu, /Hmmm... if dialog params are wrapped in a ifdef condition, then do you have something like #!define WITH_DIALOG at the top?/ But the #!define WITH_DIALOG is to choose between loading module parameters or not loading module parameters... When you do not add #!ifdef WITH_XXX before the parameters and #!endif after, then those parameters will be loaded when starting Kamailio. So, I still have the issue that the profile is not being loaded. I have tried renaming the profiles, in the route and the parameters, but that does not make a difference.. # ---- Dialog params ------------- modparam("dialog", "default_timeout", 7200) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", DLG_FLAG) modparam("dialog", "hash_size", 4096) #modparam("dialog", "detect_spirals", 1) modparam("dialog", "profiles_with_value", "inbound ; outbound") route[CONCURRENT_IN] { #Add call to customer profile #Check if customer has not reached inbound call limit #Get max concurrent calls sql_query("cc", "select max_calls_in from calllimit where cust_id='$avp(custid)'", "ra"); $avp(maxcalls)=$dbr(ra=>[0,0]); sql_result_free("ra"); $avp(concurrent) = 0; get_profile_size("inbound", "$avp(custid)", "$avp(concurrent)"); if( $avp(concurrent) >= $var(max) ) { xlog("L_INFO"," Call limit reached for customer $avp(custid)\r\n "); sl_send_reply("503", "No Lines Available"); exit; } set_dlg_profile("inbound", "$avp(custid)"); return; } But this still results in: Dec 5 09:34:33 proxy01 /usr/local/sbin/kamailio[75755]: CRITICAL: dialog [dialog.c:391]: fixup_profile(): profile <inbound> not defined Dec 5 09:34:33 proxy01 /usr/local/sbin/kamailio[75755]: ERROR: <core> [core/route.c:1166]: fix_actions(): fixing failed (code=-6) at cfg:/usr/local/etc/kamailio/kamailio.cfg:1578

2 1

question regarding rtpengine and ICE candidate selection
by Andrew Chen 04 Dec '20

04 Dec '20

Hi all, I was wondering if someone can help me understand how the ICE parameter works in the rtpengine module works. So basically our client does an ICE candidate lookup and grabs a list of them and applies it to the INVITE that gets sent to the Kamailio. The list looks like this: 2020-12-03T18:05:46.456030+00:00 ashmainkama51 kamailio[22147]: a=candidate:3373280875 1 udp 2122262783 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 59827 typ host generation 0 network-id 2 network-cost 10 2020-12-03T18:05:46.456057+00:00 ashmainkama51 kamailio[22147]: a=candidate:3040609428 1 udp 2122197247 2001:8a0:78fc:7000:d979:bf75:dbc0:69f 59828 typ host generation 0 network-id 3 network-cost 10 2020-12-03T18:05:46.456081+00:00 ashmainkama51 kamailio[22147]: a=candidate:1681997092 1 udp 2122129151 192.168.1.66 59829 typ host generation 0 network-id 1 network-cost 10 2020-12-03T18:05:46.456106+00:00 ashmainkama51 kamailio[22147]: a=candidate:2274611867 1 tcp 1518283007 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 9 typ host tcptype active generation 0 network-id 2 network-cost 10 2020-12-03T18:05:46.456131+00:00 ashmainkama51 kamailio[22147]: a=candidate:4223662180 1 tcp 1518217471 2001:8a0:78fc:7000:d979:bf75:dbc0:69f 9 typ host tcptype active generation 0 network-id 3 network-cost 10 2020-12-03T18:05:46.456155+00:00 ashmainkama51 kamailio[22147]: a=candidate:717406676 1 tcp 1518149375 192.168.1.66 9 typ host tcptype active generation 0 network-id 1 network-cost 10 2020-12-03T18:05:46.456180+00:00 ashmainkama51 kamailio[22147]: a=candidate:2130547417 1 udp 8199935 206.81.191.27 61165 typ relay raddr 85.247.0.121 rport 53091 generation 0 network-id 1 network-cost 10 In the rtpengine_offer, I have this: rtpengine_offer("ICE=force DTLS=passive replace-session-connection replace-origin external internal"); What i am trying to do is to tell the freeswitch endpoint to use rtpengine as ICE candidate and I see in the SDP this is happening: a=candidate:6296910676 1 udp 659136 206.81.191.52 52766 typ host generation 0 Btw..206.81.191.52 is a separate AWS instance running rtpengine binary. On the rtpengine_answer, I tell the remote client side the same thing..use rtpengine as your ICE candidate: rtpengine_answer("ICE=force DTLS=passive replace-session-connection replace-origin internal external"); and this is snippet from 200 OK: a=candidate:6296910676 1 udp 659136 206.81.191.52 52766 typ host generation 0 Question: Why does the rtpengine logs still show that it's trying to use 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 as RTP candidate in this scenario? Dec 3 18:05:47 ashmainrtpe42 rtpengine[8505]: DEBUG: [ep1sbnkk9tikhg4kpmot]: Forward to sink endpoint: 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71:59827 (RTP seq 25423 TS 0) I thought ICE=force will handle such adjustments so that all rtp is handled by the rtpengine to our client and bypass the STUN server? Thanks. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

3 22

REFER from MS teams to Kamailio.
by Yuriy Nasida 04 Dec '20

04 Dec '20

Well, I saw similar questions in the list already but looks like nobody has answer. Please look at REFER below. Kamilio get REFER from MS and sends it to FS node. Next, FS node try to make 3th call for some reason.I expect that FS will not do 3th call and just will connect Alice and Bob itself. 2020/05/14 12:32:00.637027 KAM_IP:5060 -> FS_IP:5060 REFER sip:Alice_number@FS_IP:5060;transport=udp SIP/2.0 FROM: Customer1<sip:MS_TRUNK_NUMBER@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=a860c50a3fb54d08b4e5740fa2dfb3d6 TO: <sip:Alice_number@FQDN_OF_TRUNK:5061>;user=phone;tag=e8ct9S6ty13va CSEQ: 4 REFER CALL-ID: 2c71b2a6669b5343a231e1244b19c945 MAX-FORWARDS: 50 Via: SIP/2.0/UDP FQDN_OF_TRUNK:5060;branch=z9hG4bK10ae.2c42897feca117121a23bf0c8d54cd19.0;i=c VIA: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bK7e3e8998 CONTACT: <sip:api-du-a-euwe.pstnhub.microsoft.com:443 ;x-i=6b68e7aa-f5e2-44ec-9edf-0bacbabfce07;x-c=2c71b2a6669b5343a231e1244b19c945/d/8/b68f86794a8e44d19543f8edbee6b2fc CONTENT-LENGTH: 0 REFER-TO: <sip:Bob_number@sip.pstnhub.microsoft.com:5061 ;user=phone;transport=tls> REFERRED-BY: <sip:sip.pstnhub.microsoft.com:5061 ;x-m=8:orgid:21bc47d3-c050-4292-8234-46f7005b97aa;x-t=fb788ef8-3c4c-455a-8d62-f3c20832c0d3;x-ti=6b68e7aa-f5e2-44ec-9edf- acbabfce07;x-tt=aHR0cHM6Ly9hcGktZHUtYS1ldXdlLnBzdG5odWIubWljcm9zb2Z0LmNvbS92MS9uZ2MvY2FsbG5vdGlmaWNhdGlvbj9kY2k9YzIxMjE3MzEyNTQ2NDk1ZjlhYTcwODliYTkwNGIxZGQ%3D> USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.5.6.2 i.EUWE.4 ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY P-ASSERTED-IDENTITY: <tel:MS_TRUNK_NUMBER>,< sip:customer1@m365x587912.onmicrosoft.com> PRIVACY: id X-AUTH-IP: 52.114.75.24 X-AUTH-PORT: 3136 Any advice?

4 11

Re: [SR-Users] Dialog profile not defined
by Gertjan Wolzak 04 Dec '20

04 Dec '20

Hello Sergiu, Thank you for the pointer, the line 1576 is: get_profile_size("callee", "$avp(custid)", "$avp(concurrent)"); That is where Kamailio refuses to start.... And as you mentioned my error in the IF statement after correction you pointed out, Kamailio still refuses on the fact that the profile <callee> was not defined... Dec 4 13:42:50 proxy01 /usr/local/sbin/kamailio[75008]: CRITICAL: dialog [dialog.c:391]: fixup_profile(): profile <callee> not defined Dec 4 13:42:50 proxy01 /usr/local/sbin/kamailio[75008]: ERROR: <core> [core/route.c:1166]: fix_actions(): fixing failed (code=-6) at cfg:/usr/local/etc/kamailio/kamailio.cfg:1576 Dec 4 13:42:50 proxy01 systemd[1]: kamailio.service: Main process exited, code=exited, status=255/n/a Dec 4 13:42:50 proxy01 systemd[1]: kamailio.service: Failed with result 'exit-code'. Does anyone else maybe have an idea on where I go wrong? Rgds, Gertjan

2 1

Fix Sdp Error
by Duarte Rocha 04 Dec '20

04 Dec '20

Hello, I noticed that when i do calls behind NAT i'm getting this error when i call fix_nated_sdp() : "nathelper [nathelper.c:1659]: replace_sdp_ip(): can't extract 'a=rtcp' IP from the SDP" As far as i can see, all the ips on SDP are correctly replaced to a public IP, so i don't understand the origin of this error. This is the SDP received by Kamailio : Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3421914885 1 IN IP4 192.168.24.89 Session Name (s): SIPPER for PhonerLite Connection Information (c): IN IP4 192.168.24.89 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 5062 RTP/AVP 107 8 0 2 3 97 110 111 9 18 11 118 101 Media Attribute (a): rtpmap:107 opus/48000/2 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:2 G726-32/8000 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:97 iLBC/8000 Media Attribute (a): rtpmap:110 speex/8000 Media Attribute (a): rtpmap:111 speex/16000 Media Attribute (a): rtpmap:9 G722/8000 Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): fmtp:18 annexb=yes Media Attribute (a): rtpmap:11 L16/44100 Media Attribute (a): rtpmap:118 L16/16000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): ssrc:2716178988 Media Attribute (a): sendrecv And this is the SDP sent by Kamailio after NAT being treated : Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3421914885 1 IN IP4 1.1.1.1 (Public IP) Session Name (s): SIPPER for PhonerLite Connection Information (c): IN IP4 1.1.1.1 (Public IP) Time Description, active time (t): 0 0 Media Description, name and address (m): audio 5062 RTP/AVP 107 8 0 2 3 97 110 111 9 18 11 118 101 Media Attribute (a): rtpmap:107 opus/48000/2 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:2 G726-32/8000 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:97 iLBC/8000 Media Attribute (a): rtpmap:110 speex/8000 Media Attribute (a): rtpmap:111 speex/16000 Media Attribute (a): rtpmap:9 G722/8000 Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): fmtp:18 annexb=yes Media Attribute (a): rtpmap:11 L16/44100 Media Attribute (a): rtpmap:118 L16/16000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): ssrc:2716178988 Media Attribute (a): sendrecv Media Attribute (a): oldmediaip:192.168.24.89 Media Attribute (a): oldmediaip:192.168.24.89 Is there something failing in the SDP fix? Best Regards,

1 0

TOPOS replication with KDMQ
by James Browne 04 Dec '20

04 Dec '20

Has anyone ever implemented DMQ replication of TOPOS data between kamailio servers? I'm trying to find the easiest (to maintain) way to implement this replication, possibly without using an external database. James

3 3

presence not work proper with subs_db_mode = 2
by Aatif Shaikh 04 Dec '20

04 Dec '20

Hello, all have a problem with Kamailio presence, I have configured my kamailio.cfg to handle dialog presence and facing issue in case of subs_db_mode = 2 in case of mode 2 facing the following presence issue. 1. after hangup presence is still up 2. sometimes stuck in a ringing state 3. sometimes presence not came..etc but in the case of subs_db_mode = 3 (DB ONLY) mode presence working fine not facing any issue with this mode. I noticed that sometimes Kamailio will throw the following message during PUBLISH processing and it will not generate appropriate NOTIFY I also verify inactive watcher subscription is there for that user and also a proper entry in the dialog table. MSG during PUBLISH processesing and it will not generate appropriate NOTIFY Oct 30 07:42:10 NC-HOSTED-1 /usr/local/sbin/kamailio[2409]: DEBUG: presence [notify.c:1444]: publ_notify(): Could not find subs_dialog kamailio.cfg param modparam("pua", "db_url", DBURL_KAMAILIO) modparam("pua", "db_mode", 2) modparam("pua", "update_period", 100) modparam("pua", "dlginfo_increase_version", 1) modparam("pua", "default_expires", 3600) modparam("pua", "fetch_rows", 1000) modparam("pua", "outbound_proxy", SERVER_IP) modparam("dialog", "db_url", DBURL_KAMAILIO ) modparam("dialog", "db_mode", 1) modparam("dialog", "db_update_period", 260) modparam("dialog", "db_fetch_rows", 500) modparam("pua_dialoginfo", "include_callid", 1) modparam("pua_dialoginfo", "send_publish_flag", 8) modparam("pua_dialoginfo", "caller_confirmed", 0) modparam("pua_dialoginfo", "include_tags", 1) modparam("pua_dialoginfo", "use_pubruri_avps", 1) modparam("pua_dialoginfo", "include_localremote", 1) modparam("pua_dialoginfo", "override_lifetime", 300) modparam("pua_dialoginfo", "pubruri_caller_avp", "$avp(s:puburis_caller)") modparam("pua_dialoginfo", "pubruri_caller_dlg_var", "pubruri_caller") modparam("pua_dialoginfo", "pubruri_callee_dlg_var", "pubruri_callee") modparam("pua_dialoginfo", "pubruri_callee_avp", "$avp(s:puburis_callee)") modparam("presence_dialoginfo", "force_single_dialog", 1) modparam("presence", "db_table_lock_type", 0) modparam("presence", "db_url", DBURL_KAMAILIO) modparam("presence", "db_update_period", 5) modparam("presence", "send_fast_notify", 1) modparam("presence", "clean_period", 100) modparam("presence", "subs_db_mode", 2) //modparam("presence", "subs_db_mode", 3) modparam("presence", "timeout_rm_subs", 1) //new/ //modparam("presence", "notifier_processes", 2) //new// modparam("presence", "notifier_poll_rate", 20) //new// modparam("presence", "waitn_time",1) modparam("presence", "fetch_rows", 1000) modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "max_expires", 3600) modparam("presence_xml", "db_url", DBURL_KAMAILIO) modparam("presence_xml", "force_active", 1) logic for presence request_route { if(is_method("PUBLISH") || is_method("SUBSCRIBE")) { route(HANDLE_PRESENCE); } } route[HANDLE_PRESENCE] { 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; } route[RELAY] { if(is_method("INVITE|BYE|UPDATE|CANCEL|ACK")) { $avp(t_ext) = $tU ; $avp(domain) = $fd ; $avp(et) = $_s(sip:$avp(t_ext)@$avp(domain)) ; $avp(s:puburis_callee) = $avp(et); setflag(8); dlg_manage(); } } I already report this issue in Kamailio git https://github.com/kamailio/kamailio/issues/2540 -- -- Regards, *Aatif Shaikh *| Software Developer +91 9033976268Hangout: aatif.shaikh(a)ecosmob.comSkype : aatif.shaikh(a)ecosmob.com *Ecosmob Technologies Pvt. Ltd. * https://www.ecosmob.com VoIP | Web | Mobile | IoT | Big Data <https://twitter.com/ecosmob> <https://www.facebook.com/Ecosmob> <https://plus.google.com/+Ecosmob> <https://www.linkedin.com/company/ecosmob> <https://in.pinterest.com/ecosmob/> This e-mail message may contain confidential or legally privileged information and is intended only for the use of the intended recipient(s). Any unauthorized disclosure, dissemination, distribution, copying or the taking of any action in reliance on the information herein is prohibited. Ecosmob Technologies is not responsible for errors or omissions in this message and denies any responsibility for any damage arising from the use of e-mail. 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1 0

Dialog profile not defined
by Gertjan Wolzak 03 Dec '20

03 Dec '20

Hello Kamailions, I am running into the following issue. The same setup I had working on 5.0, now I am using a 5.4 version of Kamailio, but I do not believe it has to do with the version, more with my ability to make errors... I am trying to limit incoming calls by using the dialog module. The max concurrent calls value I retrieve from a database. But when I try to start Kamailio it will not because the dialog profile is not defined... These are my configured Dialog parameters: # ---- Dialog params ------------- modparam("dialog", "default_timeout", 7200) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", DLG_FLAG) modparam("dialog", "hash_size", 4096) #modparam("dialog", "detect_spirals", 1) modparam("dialog", "profiles_with_value", "callee") My check on the concurrent inbound calls: route[CONCURRENT_IN] { #Add call to callee profile #Check if callee has not reached inbound call limit #Get max concurrent calls sql_query("cc", "select max_calls_in from calllimit where cust_id='$avp(custid)'", "ra"); $avp(maxcalls)=$dbr(ra=>[0,0]); sql_result_free("ra"); $avp(concurrent) = 0; get_profile_size("callee", "$avp(custid)", "$avp(concurrent)"); if( $avp(concurrent) = $var(max) ) { xlog("L_INFO"," Call limit reached for customer $avp(custid)\r\n "); sl_send_reply("503", "No Lines Available"); exit; } set_dlg_profile("callee", "$avp(custid)"); return; } And the kamailio log error: Dec 3 09:12:08 proxy01 kamailio[74102]: INFO: <core> [main.c:2833]: main(): processes (at least): 21 - shm size: 67108864 - pkg size: 8388608 Dec 3 09:12:08 proxy01 kamailio[74102]: CRITICAL: dialog [dialog.c:391]: fixup_profile(): profile <callee> not defined Dec 3 09:12:08 proxy01 kamailio[74102]: ERROR: <core> [core/route.c:1166]: fix_actions(): fixing failed (code=-6) at cfg:/usr/local/etc/kamailio/kamailio.cfg:1576 Is someone able to see where I go wrong and point me in the right direction? Thank you. Rgds, Gertjan Wolzak

2 2

$TF pseudo variable can not be used
by Володимир Іванець 03 Dec '20

03 Dec '20

Hello! I'm using a $TF variable in my config for some logging purposes. It was providing the correct string with Kamailio version 5.3.8 on CentOS 6 and 8. After updating to Kamailio version 5.4.1, the variable started to show something else. I also tried versions 5.4.2 and development 5.5.0 on CentOS 6 and the problem still persists. With version 5.5.0_dev3 this variable shows "pG▒" for me. I was wondering if someone else is using it too with Kamailio version 5.4.1 or newer on different platforms and has this issue. Thanks!

2 2

kamailio51 installs 5.2.1?
by David Villasmil 03 Dec '20

03 Dec '20

Hello everyone, On a debian buster i added the repo deb http://deb.kamailio.org/kamailio51 buster main deb-src http://deb.kamailio.org/kamailio51 buster main But when i install i get 5.2, is this right? additional info: # apt-cache showpkg kamailio Package: kamailio Versions: 5.2.1-1 (/var/lib/apt/lists/deb.debian.org_debian_dists_buster_main_binary-i386_Packages) (/var/lib/dpkg/status) Description Language: File: /var/lib/apt/lists/deb.debian.org_debian_dists_buster_main_binary-i386_Packages MD5: df3e15f422439e08c305782f5650a98c Description Language: en File: /var/lib/apt/lists/deb.debian.org_debian_dists_buster_main_i18n_Translation-en MD5: df3e15f422439e08c305782f5650a98c 5.1.10.1+bpo10 (/var/lib/apt/lists/deb.kamailio.org_kamailio51_dists_buster_main_binary-i386_Packages) Description Language: File: /var/lib/apt/lists/deb.kamailio.org_kamailio51_dists_buster_main_binary-i386_Packages MD5: ebddf40d0dfbfde1c479419970f978c7 Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

1 0

kamailio not absorbing ACK to CANCEL when behind a Path proxy
by Anthony Alba 02 Dec '20

02 Dec '20

Folks, I have a weird situation, when I enable Path header, kamailio behind a Path proxy, that kamailio doesn't recognise the ACK to CANCEL and tries to forward it. Test scenario: - voice.example.com is a stateless proxy with TLS/UDP bridging to kamailio. - on the UDP leg to kamailio it will add a Path: header to REGISTER UA1 = david is calling UA2 = charles and UA2 is sending 603 1 . UA1: preloaded route set (outbound proxy) # this is the Path proxy Route: sip:voice.example.com;transport=tls;lr 2. UA2: sends CANCEL 603 - kamailio immediately sends its (per-hop?) ACK - kamailio forwards CANCEL to UA1 kamailio sent this (per-hop?) ACK - the Route header is the internal UDP interface of the Path proxy ACK sip:charles@192.168.1.7:37309;transport=TCP;ob SIP/2.0 Via: SIP/2.0/UDP 192.168.122.99:5064;branch=z9hG4bK735d.6b27c8d09e40462ef47eccf90eb10823.0 Max-Forwards: 70 From: sip:david@voice.example.com;tag=i1acAHsitf2gcN9uHH-ZSyJ5OdT8O5M0 To: <sip:charles@voice.example.com>;tag=LmUY79rIdBQiiSakML4F0lHl271VdUn8 Call-ID: qM3W-fMrzPsvcjRSD1okEp5nbgHCF5Hu CSeq: 16667 ACK Route: <sip:192.168.122.100;lr> Content-Length: 0 3. UA1 sends ACK kamailio sees this - top Via is path proxy - 2nd Via is UA (caller) - CSeq correctly matches the INVITE - kamailio does not recognise this ACK from UA1(caller), tries to forward it, and also resends 603 to UA1(caller) 3 times ACK sip:charles@voice.example.com SIP/2.0 Via: SIP/2.0/UDP 192.168.122.100:5060;branch=z9hG4bKb3c9a7a3b6ec9e78d0144cda709f7047 Via: SIP/2.0/TLS 192.168.1.17:44925;rport=44925;branch=z9hG4bKPjFM-9gpaJEghJvN19TjnvScCwCoAGzwEc;alias;received=192.168.1.17 Max-Forwards: 70 From: sip:david@voice.example.com;tag=i1acAHsitf2gcN9uHH-ZSyJ5OdT8O5M0 To: sip:charles@voice.example.com;tag=LmUY79rIdBQiiSakML4F0lHl271VdUn8 Call-ID: qM3W-fMrzPsvcjRSD1okEp5nbgHCF5Hu CSeq: 16667 ACK Route: <sip:voice.example.com;transport=tls;lr> Content-Length: 0 Now kamailio should absorb this ACK, as it has already sent its per-hop ACK. Instead I see ERROR: <core> [core/forward.c:541]: forward_request(): cannot forward to af 2, proto 3 no corresponding listening socket I don't expect kamailio to forward anything at this point. The route block is route[WITHINDLG] and it calls t_relay() normally. route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); # we should absorb this ACK, no? exit; } 4. If I remove the Path proxy and kamailio is the TLS proxy at voice.example.com the call flow works perfectly. The ACK from caller (david) is absorbed. 5. Regular calls, when UA2 accepts the call, work fine. In-dialog requests for BYE, from both UAs, work correctly with this Path proxy. Each UA has the complete 5 element route set( 2 x Path proxy + kamailio + 2 x Path proxy). The working route set looks like this: BYE sip:david@192.168.1.17:44925;transport=TCP;ob SIP/2.0 Via: SIP/2.0/TLS 192.168.1.7:37309;rport;branch=z9hG4bKPjHdUtaUL5AfRcV-SX1Puya4niJKOOQ4VQ;alias Max-Forwards: 70 From: <sip:charles@voice.example.com>;tag=6zOt.H7w8NL5njV7klwGLlSJy.U5yBC9 To: <sip:david@voice.example.com>;tag=Pa0ORUxA-7lK0BnR4pLJ5n8rn2OvQrJy Call-ID: 3gWy4Q7O.-zQ-I07lRKHyFyX6azyCSwG CSeq: 12855 BYE Route: <sip:voice.example.com;transport=tls;lr;r2=on> Route: <sip:192.168.122.100;lr;r2=on> Route: <sip:192.168.122.99:5064;lr> Route: <sip:192.168.122.100;lr;r2=on> Route: <sip:voice.example.com;transport=tls;lr;r2=on> User-Agent: basesip 1.0.0 Content-Length: 0 Any ideas? Anthony Alba

1 0

android client SIP TLS client autentication + kamailio
by beer Ll 02 Dec '20

02 Dec '20

Hello Has anybody use a android SIP client with TLS client authentication enabled with kamailio? Do you have suggestion ? Many thanks Luca

3 2

Kamailio 5.4 ndb_redis, selecting a database
by Daniel Donoghue 02 Dec '20

02 Dec '20

Hello, I note that I can use the ndb_redis server config parameter to specify connection details for my redis servers; I also note that the connection string can specify a “db” parameter which can be set from 0 to 9, with the default being 0. I am looking for a way to dynamically set the db parameter. I do not which to add the same server with different names N times just to select a different db and having N times more connections to it than necessary. It looks as though I can call redis_cmd(“svrN”, “SELECT dbN”, “init”); but will that propagate properly and set the correct db for all connections to my redis server for the lifetime of the request? Ideally, I’d like to be able to specify an avp for the db, or have a function to set the db per request. With every blessing, — Daniel Donoghue

1 0

Kamailio Remove Remove "Allow" features from header
by me.projects＠yahoo.com 02 Dec '20

02 Dec '20

Hi, I want to remove some "Allow" features from my Kamailio SBC like I want to keep following only Allow: OPTIONS, NOTIFY, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, How can I achieve that? Thanks,

2 1

How to DIY/Setup An Open Source IP PBX Appliance/Server?
by Turritopsis Dohrnii Teo En Ming 01 Dec '20

01 Dec '20

Subject: How to DIY/Setup An Open Source IP PBX Appliance/Server? Good day from Singapore, After reading recent reviews, I gather that Asterisk is the gold standard when it comes to open source VoIP systems and it is the most famous open source PBX out there. Article: Compare the Top 10 Best Open Source PBX Software of 2020 Link: https://www.voipreview.org/business-voip/best-open-source-pbx-software Article: Top 10 Free Open Source PBX Software Solutions Link: https://getvoip.com/blog/2016/09/23/best-open-source-pbx-software/ The following is an excerpt from Wikipedia: "Asterisk is a core component in many commercial products and open-source projects. Some of the commercial products are hardware and software bundles, for which the manufacturer supports and releases the software with an open-source distribution model. AskoziaPBX, a fork of the m0n0wall project, uses Asterisk PBX software to realize all telephony functions. AstLinux is a "Network Appliance for Communications" open-source software distribution.[15] FreePBX, an open-source graphical user interface, bundles Asterisk as the core of its FreePBX Distro[16] LinuxMCE bundles Asterisk to provide telephony; there is also an embedded version of Asterisk for OpenWrt routers. PBX in a Flash/Incredible PBX and trixbox are software PBXes based on Asterisk. Elastix previously used Asterisk, HylaFAX, Openfire and Postfix to offer PBX, fax, instant messaging and email functions, respectively, before switching to 3CX. Issabel is an open-source Unified Communications software which uses Asterisk for telephony functions. It was forked from the open-source versions of Elastix when 3CX acquired it. *astTECS uses Asterisk in its VoIP and mobile gateways." Link: https://en.wikipedia.org/wiki/Asterisk_(PBX) I would like to DIY/setup an IP PBX appliance/server using free open source projects. Which free open source project, mentioned in the list and links above, would you recommend to DIY my IP PBX appliance/server? Should I buy a desktop computer or get one of those appliances listed in the link below to serve as my IP PBX appliance/server? Link: https://www.lazada.sg/products/pfsense-iron-metal-case-fanless-intel-celero… Please also refer me to very good, detailed and well explained guides/tutorials/manuals on setting up open source IP PBX appliances/servers. Lastly, please recommend a cheap and affordable IP phone (suggest brand and model) to go along with my DIY open source IP PBX appliance/server. Mr. Turritopsis Dohrnii Teo En Ming, 42 years as of 1st December 2020 Tuesday, is a TARGETED INDIVIDUAL (TI) living in Singapore. Thank you very much. -----BEGIN EMAIL SIGNATURE----- The Gospel for all Targeted Individuals (TIs): [The New York Times] Microwave Weapons Are Prime Suspect in Ills of U.S. Embassy Workers Link: https://www.nytimes.com/2018/09/01/science/sonic-attack-cuba-microwave.html ******************************************************************************************** Singaporean Targeted Individual Mr. Turritopsis Dohrnii Teo En Ming's Academic Qualifications as at 14 Feb 2019 and refugee seeking attempts at the United Nations Refugee Agency Bangkok (21 Mar 2017), in Taiwan (5 Aug 2019) and Australia (25 Dec 2019 to 9 Jan 2020): [1] https://tdtemcerts.wordpress.com/ [2] https://tdtemcerts.blogspot.sg/ [3] https://www.scribd.com/user/270125049/Teo-En-Ming -----END EMAIL SIGNATURE-----

3 2

How to msg_apply_changes in a branch route?
by Anthony Alba 01 Dec '20

01 Dec '20

Is there a way to obtain the effect of msg_apply_changes in a branch route? I want to do: route[BRANCHMANAGE] { rtpengine_manage() msg_apply_changes() // further mangle the SDP with textops for // obtuse UAs } The reason for this, is that after forking, I have some UAs that are extremely picky about SDP, and I need to mangle the body to make them happy. (4000 char limit, 68 attribute limit on SDP body - ever seen that? Compared with the absolutely gigantic OFFERs from WebRTC signalling libraries...) Now rtpengine_manage() and textops are working on separate copies of the msg body and the results don't stack correctly. Cheers Anthony Alba

2 1

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