May 2017 - sr-users - kamailio.org

rtjson + uac + event - OK, ACK and BYE display name not updated
by Diego Nadares 30 Jan '18

30 Jan '18

Hi Guys, We are modifying "from" and "to" headers with rtjson module via a json struct. It works fine in INVITE, TRYING, RINGING and OK but not in the following messages. Kamailio It's changing ONLY "from" and "to" URI BUT NOT the DISPLAY NAME. I think it's the tm module but I can't figure out how to confirm this. Maybe I need to configure something else in my cfg or could it be a bug? My json struct that is handled by rtjson module response.routing = "serial"; //response.routing = "parallel"; response.routes = []; response.routes[0] = {}; response.routes[0].uri = "sip:1111111111@172.16.213.31:5060"; response.routes[0].headers = { // Headers to be modified "from": { "display": "1111111111", "uri": "sip:111111111@172.16.213.38" }, "to": { "display": "2222222222", "uri": "sip:2222222222@172.16.213.38" }, }; response.routes[0].headers.extra = {};//"X-Hdr-A: abc\r\nX-Hdr-B: bcd\r\n"; This is part of my cfg .... modparam("dialog","dlg_flag", FLD_START) modparam("uac","restore_mode","auto") modparam("uac", "restore_dlg", 1) .... # account only INVITEs if (is_method("INVITE") && !has_totag()) { setflag(FLT_ACC); # do accounting * setflag(FLD_START); # do accounting* setflag(FLT_ACCMISSED); setflag(FLT_ACCFAILED); } ..... >From source/Aleg: *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>.* To dest/Bleg: *From: 1111111111 <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: 2222222222 <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>.* Incorrect display name in BYE and ACK: *From: sipp <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: sut <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* interface: any filter: ( port 5060 ) and (ip or ip6) # U 2016/12/06 15:52:13.638731 172.16.213.21:5060 -> 172.16.213.38:5060 INVITE sip:5555555555@172.16.213.38:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-0. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Contact: "SIPP" <sip:sipp@172.16.213.21:5060>. Max-Forwards: 70. Subject: Performance Test. Content-Type: multipart/mixed;boundary=uniqueBoundary. Remote-Party-ID: <sip:sipp@172.16.213.21 >:5060;party=calling;id-type=subscriber;privacy=off;screen=no>. Content-Length: 549. . --uniqueBoundary. Content-Type: application/sdp. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.21. s=-. c=IN IP4 172.16.213.21. t=0 0. m=audio 6000 RTP/AVP 8. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. . --uniqueBoundary. Content-Type: application/gtd. Content-Disposition: signal;handling=optional. . IAM,. CGN,04,y,1,y,4,1143617400. CIC,0000000573. CPC,09. CPN,02,y,1,52381660. FCI,n,n,n,n,y,n,n,u. GCI,f7140cc78a611601838a002128d7e512. NOC,0,n,1,n. PRN,q761*,AR*,oper2,1993. TMR,02. . --uniqueBoundary--. ................ # U 2016/12/06 15:52:13.734159 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-0. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Server: kamailio (4.4.4 (x86_64/linux)). Content-Length: 0. . ................ # U 2016/12/06 15:52:13.878600 172.16.213.38:5060 -> 172.16.213.31:5060 INVITE sip:1111111111@172.16.213.31:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bKabaa.840cc83114975a34a37b7af020d37b65.1. *From: 1111111111 <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: 2222222222 <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Max-Forwards: 69. Subject: Performance Test. Content-Type: multipart/mixed;boundary=uniqueBoundary. Content-Length: 788. Contact: <sip:btpsh-58470879-6c61-1@172.16.213.38>. . --uniqueBoundary. Content-Type: application/sdp. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.38. s=-. c=IN IP4 172.16.213.38. t=0 0. m=audio 40236 RTP/AVP 8. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. a=sendrecv. a=rtcp:40237. a=ice-ufrag:QNCzXhUr. a=ice-pwd:Ci90BRpdZH51Hx5BaPgGMrGCeh. a=candidate:P1djl8LFNb5kexbG 1 UDP 2130706431 172.16.213.38 40236 typ host. a=candidate:P1djl8LFNb5kexbG 2 UDP 2130706430 172.16.213.38 40237 typ host. . --uniqueBoundary. Content-Type: application/gtd. Content-Disposition: signal;handling=optional. . IAM,. CGN,04,y,1,y,4,1143617400. CIC,0000000573. CPC,09. CPN,02,y,1,52381660. FCI,n,n,n,n,y,n,n,u. GCI,f7140cc78a611601838a002128d7e512. NOC,0,n,1,n. PRN,q761*,AR*,oper2,1993. TMR,02. . --uniqueBoundary--. ................ # U 2016/12/06 15:52:13.879441 172.16.213.31:5060 -> 172.16.213.38:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bKabaa.840cc83114975a34a37b7af020d37b65.1. *From: 1111111111 <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: 2222222222 <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Contact: <sip:172.16.213.31:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 137. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.31. s=-. c=IN IP4 172.16.213.31. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. ................ # U 2016/12/06 15:52:13.936565 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 180 Ringing. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Content-Type: application/sdp. Content-Length: 316. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-0. Contact: <sip:atpsh-58470879-6c61-2@172.16.213.38>. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.38. s=-. c=IN IP4 172.16.213.38. t=0 0. m=audio 40248 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=sendrecv. a=rtcp:40249. a=candidate:P1djl8LFNb5kexbG 1 UDP 2130706431 172.16.213.38 40248 typ host. a=candidate:P1djl8LFNb5kexbG 2 UDP 2130706430 172.16.213.38 40249 typ host. ................ # U 2016/12/06 15:52:20.882165 172.16.213.31:5060 -> 172.16.213.38:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bKabaa.840cc83114975a34a37b7af020d37b65.1. *From: 1111111111 <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: 2222222222 <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Contact: <sip:172.16.213.31:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 213. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.31. s=-. c=IN IP4 172.16.213.31. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. ................ # U 2016/12/06 15:52:21.173798 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 200 OK. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Content-Type: application/sdp. Content-Length: 380. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-0. Contact: <sip:atpsh-58470879-6c61-2@172.16.213.38>. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.38. s=-. c=IN IP4 172.16.213.38. t=0 0. m=audio 40248 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. a=rtcp:40249. a=candidate:P1djl8LFNb5kexbG 1 UDP 2130706431 172.16.213.38 40248 typ host. a=candidate:P1djl8LFNb5kexbG 2 UDP 2130706430 172.16.213.38 40249 typ host. ................ # U 2016/12/06 15:52:21.174390 172.16.213.21:5060 -> 172.16.213.38:5060 ACK sip:atpsh-58470879-6c61-2@172.16.213.38 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-4. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 ACK. Contact: sip:sipp@172.16.213.21:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . ................ # U 2016/12/06 15:52:21.248172 172.16.213.38:5060 -> 172.16.213.31:5060 ACK sip:172.16.213.31:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bKabaa.98116b5795d4c30f52156187a72059c8.0. *From: sipp <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: sut <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 ACK. Max-Forwards: 69. Subject: Performance Test. Content-Length: 0. Contact: <sip:btpsh-58470879-6c61-1@172.16.213.38>. . ................ # U 2016/12/06 15:52:51.178826 172.16.213.21:5060 -> 172.16.213.38:5060 BYE sip:atpsh-58470879-6c61-2@172.16.213.38 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-7. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 2 BYE. Contact: sip:sipp@172.16.213.21:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . ................ # U 2016/12/06 15:52:51.468639 172.16.213.38:5060 -> 172.16.213.31:5060 BYE sip:172.16.213.31:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bK7baa.3cb44935b506e759e3418501b0688b08.0. *From: sipp <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: sut <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 2 BYE. Max-Forwards: 69. Subject: Performance Test. Content-Length: 0. Contact: <sip:btpsh-58470879-6c61-1@172.16.213.38>. . ................ # U 2016/12/06 15:52:51.469153 172.16.213.31:5060 -> 172.16.213.38:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bK7baa.3cb44935b506e759e3418501b0688b08.0. *From: sipp <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: sut <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 2 BYE. Contact: <sip:172.16.213.31:5060;transport=UDP>. Content-Length: 0. . ................ # U 2016/12/06 15:52:51.519423 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 200 OK. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 2 BYE. Content-Length: 0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-7. Contact: <sip:atpsh-58470879-6c61-2@172.16.213.38>. . ................ exit 13 received, 0 dropped Sorry for my english. Thanks in advance! Diego.

2 7

pua.publish using RPC
by Alex Balashov 12 Dec '17

12 Dec '17

Hello, Is pua.publish strictly an MI function, or is it possible nowadays to call it via the RPC channel, and specifically, using jsonrpc_exec()? Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 5

fix_nated_contact and IPv6
by Sebastian Damm 27 Oct '17

27 Oct '17

Hi, I'm trying to set up a Kamailio (4.1.3) server which converts between IPv6 and IPv4. Everything looks pretty good. Now I have a test client which sends packets via IPv6, but in contact header, SDP etc. there are still IPv4 addresses. (This comes from some converting from v4 to v6 earlier.) Thus, Kamailio tries to do a fix_nated_contact(). But after fixing, the Contact URI is wrong. Example: inbound Contact: <sip:bob@192.168.8.132> outbound Contact: <sip:bob@1234:1234:0:1234:0:0:0:2> The correct way of transforming the Contact URI would be: Contact: <sip:bob@[1234:1234:0:1234:0:0:0:2]> This incorrect Contact URI doesn't hurt first, but when the called party wants to hang up the call, it addresses the URI from Contact header, and Kamailio can't parse the Request URI: Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: pv [pv_core.c:304]: pv_get_ruri_attr(): failed to parse the R-URI Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: rr [loose.c:934]: loose_route(): failed to parse Request URI Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: domain [domain.c:140]: is_uri_host_local(): error while parsing R-URI Is this a bug in the nathelper module? Was it never meant to handle IPv6 addresses? Or do I understand something wrong? Best Regards, Sebastian

4 4

NatHelper ignoring force_socket module parameter
by Asgaroth 24 Oct '17

24 Oct '17

Hi All, I have a strange issue, I have set the module parameter for nathelper's force_socket to a specfic ip/port, however, when I perform a sip trace I can see that all locally generated options messages are not sent from the socket defined in the modules parameters. I am not setting $fs anywhere in the script, so I am not over-writing it. Do any of the other module parameters have a bearing on if nathelper uses the force_socket parameter? My current nathelper settings are: modparam("nathelper", "received_avp", "$avp(RECEIVED)") modparam("nathelper", "natping_interval", 20) modparam("nathelper", "natping_processes", 4) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_from", "sip:keepalive@domain.com") modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "sipping_bflag", NAT_BFLAG) modparam("nathelper", "force_socket", "1.2.3.4:5060") I am using kamailio v4.3.1: # kamailio -V version: kamailio 4.3.1 (x86_64/linux) f38e67 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, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: f38e67 compiled on 18:15:23 Jul 20 2015 with gcc 4.4.7 Thanks

2 5

Kamailio IMS deployment
by Ramya Y 22 Oct '17

22 Oct '17

Hi, I am newbie to kamailio world and we would like to deploy Kamailio based IMS platform. Can any body share the link or Step by step procedure(user Guide) to deploy Kamailio IMS. Regards, Ramya

5 22

Kamailio 4.3.4: receive_msg(): no via found in reply
by Grant Bagdasarian 15 Aug '17

15 Aug '17

Hello, We're testing kamailio 4.3.4 with two kamailio's communicating with each other. One of them is a gateway with rtpengine at the carrier side, the other is a basic sip router. When calling a destination and cancelling the INVITE a 487 is received from the carrier and directly ACK'd by the first kamailio and in turn is forwarded to the second. But the 487 response received by the 2nd kamailio doesn't have a Via header which results in kamailio printing the error: receive_msg(): no via found in reply. Any ideas why the forwarded 487 doesn't contain a Via header and how to fix this? Note: IP's are changed. Carrier: 10.0.0.1 Kamailio 1: 10.14.0.1 Kamailio 2: 10.14.0.2 U 10.0.0.1:5060 -> 10.14.0.1:5060 SIP/2.0 487 Request Cancelled. Via: SIP/2.0/UDP 10.14.0.1;branch=z9hG4bK8e28.642c68037d059634308d4be8fe3f0cdc.0. From: "+31612345678" <sip:+31612345678@10.14.0.3:5060>;tag=109ac722. To: <sip:003112345678@10.14.0.3:5060>;tag=sbcsipuas_1_C22196_20160314080307791_b59sb10. Call-ID: 78100MjY3OGExZmMyODU2MzAyMzM5ODA3MmNhMDdlMzg0MzU. Contact: <sip:10.0.0.1:5060>. CSeq: 2 INVITE. Server: sbc_5. Content-Length: 0. . # U 10.14.0.1:5060 -> 10.0.0.1:5060 ACK sip:003112345678@10.0.0.1:5060 SIP/2.0. Via: SIP/2.0/UDP 10.14.0.1;branch=z9hG4bK8e28.642c68037d059634308d4be8fe3f0cdc.0. Max-Forwards: 67. To: <sip:003112345678@10.14.0.3:5060>;tag=sbcsipuas_1_C22196_20160314080307791_b59sb10. From: "+31612345678"<sip:+31612345678@10.14.0.3:5060>;tag=109ac722. Call-ID: 78100MjY3OGExZmMyODU2MzAyMzM5ODA3MmNhMDdlMzg0MzU. CSeq: 2 ACK. Content-Length: 0. . # U 10.14.0.1:5060 -> 10.14.0.2:5060 SIP/2.0 487 Request Cancelled. From: "+31612345678" <sip:+31612345678@10.14.0.3:5060>;tag=109ac722. To: <sip:003112345678@10.14.0.3:5060>;tag=sbcsipuas_1_C22196_20160314080307791_b59sb10. Call-ID: 78100MjY3OGExZmMyODU2MzAyMzM5ODA3MmNhMDdlMzg0MzU. Contact: <sip:10.0.0.1:5060>. CSeq: 2 INVITE. Server: sbc_5. Content-Length: 0.

4 16

Rate limiting
by NITESH BANSAL 31 Jul '17

31 Jul '17

Hello, I plan to add some rate limiting in the Kamailio. I'm already using Pike to rate limit based on the source IP. But in this case, i want to do the rate limiting per customer(regardless of the source IP), I can identify the customer based on a custom header in the INVITE, idea is to limit the customer to 'x' number of requests per second. Nitesh

5 12

Problem with t_relay() and VPN tunnel
by Carlos Ruiz Diaz 24 Jun '17

24 Jun '17

Hi, I have two SIP servers, *A* and *B*, connected each other though a OPENVPN tunnel. The server *B* needs to t_relay() every SIP message containing the method MESSAGE to the server *A* but these messages never reach destination. I have tested the tunnel connectivity and works fine. I wrote a Perl script (located in *B*) that sends SIP MESSAGES to Kamailio (located in *A*) trying to figure out what is happening but these messages are received by *A* and processed correctly but when *B* does the same from Kamailio, it is never received. Here is the route part of kamailio.cfg in *B*: Observation: ($rU == "1004") result is *true* * if(is_method("MESSAGE")) { if($rU == "1004") { xlog("L_INFO","En 1004"); rewritehost("10.8.0.1"); if (!t_relay()) xlog("L_INFO","MIO Error en t_relay"); t_reply("200", "Ok"); xlog("L_INFO","MIO despues rewrite"); exit; } .....* The perl script that WORKS: *$msg = 'MESSAGE sip:1004@192.168.1.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2 From: "2002" <sip:2002@192.168.1.2 <sip%3A2002(a)192.168.1.2>>;tag=1837944796 To: <sip:1004@192.168.1.2 <sip%3A1004(a)192.168.1.2>> Call-ID: 19722852989(a)192.168.1.2 CSeq: 15773 MESSAGE Contact: <sip:2002@**PublicIP-protected**:5060> Max-Forwards: 29 User-Agent: DBL Content-Type: text/plain Content-Length: 34 +595981[protected] hello from kamailio '; use IO::Socket; my $sock = IO::Socket::INET->new( Proto=>'udp', PeerHost=>'192.168.2.102', PeerPort=>'5060'); print "Sending msg $msg\n"; $sock->send($msg) or die "error sending $!\n"; *Please help! Thanks in advance. Carlos.

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REDIS load problem - Kamailio 5.0.1 - db_redis not found
by Marko Tirs 14 Jun '17

14 Jun '17

Hi all, I'm trying to use the Redis database for authentication and later for accounting. I installed Redis on the server, I compiled Kamailio with ndb_redis and the module exists. When I try to start Kamailio I get following errors: 0(16737) ERROR: <core> [db.c:203]: db_bind_mod(): Module db_redis not found. Missing loadmodule? 0(16737) ERROR: auth_db [auth_db_mod.c:175]: mod_init(): unable to bind to a database driver 0(16737) ERROR: <core> [core/sr_module.c:968]: init_mod(): Error while initializing module auth_db (/usr/local/lib64/kamailio/modules/auth_db.so) ERROR: error while initializing modules Bellow are the relevant parts from config files. Is there any example of kamailio.cfg with Redis configuration and commands for authentication and accounting? Please help me to use Kamailio with Redis. Thank youRegardsMarko kamctlrc --------- DBENGINE=DBTEXT kamailio.cfg ------------- #!define WITH_REDIS #!define WITH_AUTH #!ifdef WITH_REDIS #!define DBURL "redis://localhost:6379/" #!endif #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!endif #!ifdef WITH_REDIS loadmodule "ndb_redis.so" #!endif # ----- ndb_redis params ----- modparam("ndb_redis", "server", "name=srv1;addr=127.0.0.1;port=6379;db=1") # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) #!endif | | Virenfrei. www.avast.com |

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Issue with Dispatcher Failover algorithm
by Cristi Constantin 07 Jun '17

07 Jun '17

Hello, I have an issue with the behavior of ds_select_dst() method in case of using algorithm 8 (dispatch based on priority). In case both destinations are available, they are used alternatively, instead of always using the one with the highest priority. By searching through the mailing list I found that the lower the priority number, the higher the priority (1 is first, 2 is second, etc.). Based on this, I set the following in dispatcher.list: 7 sip:10.126.54.51:5060;transport=sctp 9 1 7 sip:10.126.55.51:5060;transport=sctp 9 2 After running "kamcmd dispatcher.reload" and "kamcmd dispatcher.list", I have the confirmation that the priorities are correctly set: SET: { ID: 7 TARGETS: { DEST: { URI: sip:10.126.55.51:5060 ;transport=sctp FLAGS: AP PRIORITY: 2 } DEST: { URI: sip:10.126.54.51:5060 ;transport=sctp FLAGS: AP PRIORITY: 1 } } } The dispatcher settings from kamailio.cfg is the following: modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:kamailio@sdp-t-lb2") modparam("dispatcher", "ds_probing_mode", 3) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)") And the code which should do the displatch is the following: if (ds_is_from_list(7, 3, "$ru")) { remove_hf("Route"); if(!ds_select_dst("7", "8")) { send_reply("404", "No destination"); exit; } } With the settings above, I was expecting to always have the messages delivered to 10.126.54.51. Instead, they are dispathed to 10.126.54.51 and 10.126.55.51 alternatively. Thanks a lot, Cristian

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