sr-users May 2017

sr-users@lists.kamailio.org

91 participants
116 discussions

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

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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

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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.

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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

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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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Auth_radius - digest auth problem
by Donat Zenichev 06 Jun '17

06 Jun '17

Hi all. Have a problem with radius authorization. I'm using auth_radius.so modparams, only path to client file: modparam("auth_radius", "radius_config", "/etc/radiusclient/radiusclient.conf") Freeradius installed and is working properly, radtest authentication from kamailio host succeed . How authorization block looks like: if (!is_present_hf("Authorization")) { xlog("L_NOTICE", "----- Athorization HF is not found - passing the challenge -----\n"); if (nat_uac_test("2")) { force_rport(); } www_challenge("$fd", "1"); exit; if (!radius_www_authorize("$fd","$fU")) { if (nat_uac_test("2")) { force_rport(); } xlog("L_NOTICE", "----- Registeration $au@$ar ($fU) from $si:$sp Rejected. Code: $rc -----\n"); sl_send_reply("401","Unauthorized"); exit; Radius log is filled by rows like: Auth: [digest] Cleartext-Password or Digest-HA1 is required for authentication. Tried to use radius_www_authorize without $fU - didn't change anything. Tried to use www_challenge without qop - didn't change anything. So, this solution is quite simple, but I have a fail while digest authentication. Any ideas? -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-808 http://wnet.ua

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topos module - possible bug
by Pete Kelly 02 Jun '17

02 Jun '17

Hi I am using the topos module when bridging 2 networks with Kamailio. The INVITE/200OK part of the transaction is working fine (i.e. the Contact on both sides matches correctly the corresponding network). However when the ACK is sent into Kamailio, instead of realising the next hop is myself and skipping it, Kamailio is sending the ACK directly to itself as a packet, causing the call setup to break. Does anyone have any advice for this situation?

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rtpengine nattede WAN address
by gerry kernan 01 Jun '17

01 Jun '17

Hi I'm looking to check if what I'm trying to configure is feasible . UA -> WAN(X.X.X.X) -> Kamailio(LAN 192.X.X.148) -> internal PBX(LAN 192.X.X.131) I have a WN IP forwarded to and internal Kamailio server which is sitting in front of an asterisk server. SIP requests are sent to asterisk from Kamailio via dispatcher . SIP client can register via WAN and Kamailio to asterisk , I have rtpenine configured as below. /usr/sbin/rtpengine --interface=192.X.X.148!X.X.X.X --listen-ng=7722 --timeout=60 --silent-timeout=3600 --pidfile=/var/run/ngcp-rtpengine-daemon.pid --table=0 --log-level=6 --log-facility=local7 --log-facility-rtcp=local7 --homer=192.168.200.5:9060 --homer-protocol=udp --homer-id=2002 But SIP SDP when sent to asterisk is still show the WAN IP of the Sip client in the SDP offer that goes to asterisk , ive tried adding parameter "via-branch=" and "auto-bridge" to rtpengine_offer but they don't seem to make a difference , is there anyone way I can force the SDP offer that goes to the backend PBX to use the IP of Kamailio server that rtpengine is running on rather that the WAN IP that I'm advertising(X.X.X.X) ? Example SIP SDP from WAN client to asterisk INVITE sip:*97@sip.domain.com;transport=UDP SIP/2.0 Via: SIP/2.0/UDP Client-IP:29345;branch=z9hG4bK-524287-1---e38082af1f110ad1;rport Max-Forwards: 70 Contact: <sip:21@Client-IP:29345;transport=UDP> To: <sip:*97@sip.domain.com;transport=UDP> From: <sip:21@sip.domain.com;transport=UDP>;tag=e637343b Call-ID: jvqeIB9f48_OGcTq0UxP7w.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rv2.8.40 o=Z 0 0 IN IP4 Client-IP s=Z c=IN IP4 Client-IP t=0 0 m=audio 29344 RTP/AVP 8 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv and SDP that Kamailio sends to asterisk INVITE sip:*97@sip.domain.com;transport=UDP SIP/2.0 Record-Route: <sip:192.X.X.148;lr> Via: SIP/2.0/UDP 192.X.X.148;branch=z9hG4bKd924.bcc4248735b53e9cf783c87c81dfee28.0 Via: SIP/2.0/UDP Client-IP:29345;received=Client-IP;branch=z9hG4bK-524287-1---e38082af1f110ad 1;rport=29345 Max-Forwards: 69 Contact: <sip:21@Client-IP:29345;transport=UDP> To: <sip:*97@sip.domain.com;transport=UDP> From: <sip:21@sip.domain.com;transport=UDP>;tag=e637343b Call-ID: jvqeIB9f48_OGcTq0UxP7w.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rv2.8.40 Path: <sip:192.X.X.148;lr;received=sip:Client-IP:29345> v=0 o=Z 0 0 IN IP4 Client-IP s=Z c=IN IP4 X.X.X.X t=0 0 m=audio 32550 RTP/AVP 8 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:32551 a=ice-ufrag:Dwd23Dem a=ice-pwd:9Itp6Cq7lkYJxMkeDeA1I2kkzw a=candidate:KZyzUR5lXJJpt1Et 1 UDP 2130706431 X.X.X.X 32550 typ host a=candidate:KZyzUR5lXJJpt1Et 2 UDP 2130706430 X.X.X.X 32551 typ host Gerry Kernan Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland Tel: +353 - (0)1 - 293 0090 | E-Mail: <mailto:gerry.kernan@infinityit.ie> gerry.kernan(a)infinityit.ie Managed IT Services Infinity IT - <http://www.infinityit.ie/> www.infinityit.ie IP Telephony Asterisk Consulting - <http://www.asteriskconsulting.com> www.asteriskconsulting.com Contact Centre Total Interact - <http://www.totalinteract.com> www.totalinteract.com

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sr-users May 2017