- sr-users - kamailio.org

Websocket kamailio.cfg doubt
by Austin Einter 23 Mar '15

23 Mar '15

The README of websocket suggests to modify kamailio.cfg as below *# Handle requests within SIP dialogsroute[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if ($du == "") { if (!handle_ruri_alias()) { xlog("L_ERR", "Bad alias <$ru>\n"); sl_send_reply("400", "Bad Request"); exit; } } route(RELAY); } else { if ( is_method("ACK") ) {...* But the original script was *# Handle requests within SIP dialogsroute[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 statelessy 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; } } sl_send_reply("404","Not here"); exit;}* Now I have to include the websocket README suggested script in original script. I am not finding logical way where shall I include websocket part. Can somebody please help me. Thanks Austin

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Kamailio sending NOTIFY messages to private IP
by Olli Attila 23 Mar '15

23 Mar '15

Hello, Im running kamailio 4.2.3 and I'm having troubles with reporting user presence data from Kamailio server to a registered UA behind nat. The NOTIFY message is being sent from Kamailio to the UA after receiving PUBLISH, but because the UA is behind nat the NOTIFY message is being sent to the UAs private (=unroutable) ip and therefor never reaching the real UA. My server is using nat helping modules so that REGISTER messages and INVITE messages are handled ok even if they are originated behind nat. Any suggestions how to solve this and make the NOTIFY messages to be sent via routabe ip addresses? my kamailio.cfg PRESENCE routing: #!ifdef WITH_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; #!endif if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } -- Olli

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Re: [SR-Users] kamailio asterisk NOTIFY
by Slava Bendersky 23 Mar '15

23 Mar '15

Hello Daniel, I still can't understand clearly why when asterisk send NOTIFY to client through kamailio it coming back with with 408. I I tried different case event change Contact:, but no luck can't I see what the issue all routes in first SUBSCRIBE looks ok . Any though or help will be really appreciate. This what I see on asterisk debug Reliably Transmitting (NAT) to 10.18.130.46:5060: NOTIFY sip:102@192.168.88.243:5068;alias=client_pub_ip~5068~2 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.51:5060;branch=z9hG4bK5dd3f266 Max-Forwards: 70 Route: <sip:10.18.130.46:5060;lr=on;ftag=3b27a2a409;nat=yes> --->Kamailio route From: "canlpbx02" <sip:canlpbx02@10.18.130.51>;tag=as56402f92 To: <sip:102@192.168.88.243:5068;alias=client_pub_ip~5068~2>;tag=3b27a2a409 Contact: <sip:canlpbx02@10.18.130.51:5060> Call-ID: b1edde040d695988 CSeq: 102 NOTIFY User-Agent: FPBX-12.0.43(12.8.1) Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 88 Messages-Waiting: yes Message-Account: sip:*97@10.18.130.51 Voice-Message: 5/0 (0/0) --- <--- SIP read from UDP:10.18.130.46:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.18.130.51:5060;rport=5060;branch=z9hG4bK5dd3f266 From: "canlpbx02" <sip:canlpbx02@10.18.130.51>;tag=as56402f92 To: <sip:102@192.168.88.243:5068;alias=client_pub_ip~5068~2>;tag=3b27a2a409 Call-ID: b1edde040d695988 CSeq: 102 NOTIFY Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE Server: Media5-fone/4.1.6.3283 Android/5.1 Supported: eventlist, replaces, timer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:10.18.130.46:5060 ---> SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.18.130.51:5060;branch=z9hG4bK6386387f;rport=5060 From: "canlpbx02" <sip:canlpbx02@10.18.130.51>;tag=as00936a6d To: <sip:102@192.168.88.243:5066;alias=client_pub_ip~5066~2>;tag=a3aea18741 Call-ID: 5a66b87a618d1042 CSeq: 103 NOTIFY Server: kamailio (4.2.3 (x86_64/linux)) Content-Length: 0 <-------------> Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca>, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, March 13, 2015 1:28:33 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, that function might be in textopsx module. Cheers, Daniel On 13/03/15 14:25, Slava Bendersky wrote: Hello Daniel, Where possible find updated version of options for textops. [sr_module.c:701]: find_mod_export_record(): find_export_record: <remove_hf_value> not found Specially Record-Route manipulation. Slava. From: "Slava Bendersky" <volga629(a)networklab.ca> To: miconda(a)gmail.com Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, March 13, 2015 8:54:26 AM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello Daniel, Here paste for first SUBSCRIBE. http://fpaste.org/197545/25118514/ Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, March 12, 2015 5:09:17 AM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, can you get the sip traffic from the network (from the first subscribe of this presence dialog)? It will help to see the Route set. Cheers, Daniel On 12/03/15 00:36, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Here debug of the notify which failing. Look like transaction miss match http://fpaste.org/196896/61168591/ Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 11, 2015 5:44:05 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, it doesn't matter if there is a mix of transports. If the UA is mirroring the Record-Route headers in the response to the NOTIFY, it should work fine. Cheers, Daniel On 11/03/15 21:40, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Is this matter that UDP and TCP transport in mix ? Slava. From: "Slava Bendersky" <volga629(a)networklab.ca> To: miconda(a)gmail.com Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 11, 2015 4:21:39 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello Daniel, I have record route in WITHINDLG for NOTIFY, but it not helped behaviour is the same it loosing route for NOTIFY. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 11, 2015 2:14:02 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, you have to do record_route() for all NOTIFY requests -- there are new specs requiring that (RFC 6665) and perhaps asterisk is following them. If you look in the default kamailio.cfg for v4.2, inside route[WITHINDLG], then you will see how it is done. Cheers, Daniel On 11/03/15 14:22, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Here fresh paste http://fpaste.org/196616/42608006/ I see proper routing headers on SUBSCRIBE, but then NOTIFY loose it. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 10, 2015 5:31:51 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, fpaste link doesn't show anything. Can you check if the NOTIFY coming to Kamailio has two Route headers? Cheers, Daniel On 10/03/15 18:44, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, Having issue with NOTIFY, is not forwarded to client properly. asterisk ---> private kamailio public ----> soft client http://fpaste.org/196257/99948714/ I see kamailio trying connect always with private IP to the client when NOTIFY is come is in. Is this something to do with rr_advertise_address ? Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.88.246:5064 failed (timeout) Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7fbbb64e2800 Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9dbf00, 33, -1, 0x10) fd_no=24 called Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4320]: tcpconn_main_timeout(): tcp_main: entering timer for 0x7fbbb64eb6c0 (ticks=839577436, timeout=839634956 (3595 s), wr_timeout=839577436 (0 s)), write queue: 798 bytes Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.88.246:5066 failed (timeout) Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7fbbb64eb6c0 Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9dbf00, 34, -1, 0x10) fd_no=23 called Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1294]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1812]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: <core> [mem/shm_mem.c:111]: _shm_resize(): WARNING:vqm_resize: resize(0) called Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1294]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1812]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 I have in configuration in WITHINDLG if (is_method("NOTIFY")) { record_route(); } and in NATDETECT if(is_method("INVITE|SUBSCRIBE|UPDATE|NOTIFY")) { set_contact_alias(); } Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

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Re: [SR-Users] remove_hf_value
by Slava Bendersky 23 Mar '15

23 Mar '15

Hello Daniel, I end up insert routes like this. And right now on client side I see only public ip route and private side only private ip routes. route[ROUTE_HDR_ORG] { # Getting record routes if(is_present_hf("Route")) { xlog("L_INFO", "[$rm], Inbound connection from <$si>\n"); if(is_method("NOTIFY")) { if(is_ip_rfc1918($si)) { xlog("L_INFO", "[$rm], Present Route: $(route_uri{uri.host})\n"); remove_hf("Route"); insert_hf("Record-Route: <"sip" + $sel(cfg_get.kamailio.bindpublicip) + ":" $sel(cfg_get.kamailio.bindpublicport);transport=tcp;lr=on;ftag=$ft;nat=yes>\r\n"); $fs = "tcp" + ":" + $sel(cfg_get.kamailio.bindpublicip) + ":" + $sel(cfg_get.kamailio.bindpublicport); } else { xlog("L_INFO", "[$rm], Present Route: $(route_uri{uri.host})\n"); remove_hf("Route"); insert_hf("Record-Route: <"sip" + ":" $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport);lr=on;ftag=$ft;nat=yes>\r\n"); $fs = "udp" + ":" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport); } } } } And for initial SUBSCRIBE INVITE if (is_method("INVITE|SUBSCRIBE")) { remove_hf("Route"); insert_hf("Record-Route: <"sip" + ":" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport);lr=on;ftag=$ft;nat=yes>\r\n"); #record_route(); } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca>, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 18, 2015 4:15:28 AM Subject: Re: [SR-Users] remove_hf_value Hello, but the INVITE request goes out with 2 Record Route headers, if you get them in the 200ok. Not sure how it is going to work overall, you will have to manually set local sockets for requests within dialog, because you are doing also trasport layer gatewaying (tcp-udp). Anyhow, as a solution, maybe you should just not be using record_route() function and just add the header with insert_hf() for the invite, remove the Record-Route header for 200ok with remove_hf() and insert a new one again with insert_hf() If you know the local address (IP/port/transport) you want to put in the Record-Route, you can build the header as: insert_hf("Record-Route: <sip:_LOCAL_IP_:_LOCAL_PORT_;transport=_LOCAL_TRANSPORT_;lr=on;ftag=$ft;nat=yes> \r\n"); Requests within the dialog can be handled with loose_route(), but again, you have to do more in config for selecting the local socket to send to next hop. Cheers, Daniel On 16/03/15 13:35, Slava Bendersky wrote: Hello Daniel, I want do clean up on routes from first SUBSCRIBE, that when traffic goes to public side of kamailio will be only public route and when traffic is internal the Record-Route will stay internal only. In SDP I see like this. I tired double routes set 0 in rr, but didn't clean enough. <--- Transmitting (NAT) to 10.18.130.46:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.18.130.46;branch=z9hG4bKf852.e0223f39c2bbad8366fdf1b7cb22b336.0;i=8;received=10.18.130.46;rport=5060 Via: SIP/2.0/TCP 192.168.88.252:5062;received=client_pub_ipbranch=z9hG4bK0bbe1f7d27257bba9;rport=5062 Record-Route: <sip:10.18.130.46;r2=on;lr=on;ftag=a185d974ec;nat=yes> On public side remote this Record-Route Record-Route: <sip:kamailio_pub_ip:5084;transport=tcp;r2=on;lr=on;ftag=a185d974ec;nat=yes> On private side remote this Record-Route From: "Slava Bendersky" <sips:10101@networklab.ca> ;tag=a185d974ec To: <sips:10101@networklab.ca> ;tag=as00757d3e Call-ID: b08adb1ad1804a83 CSeq: 236711034 SUBSCRIBE Server: FPBX-2.11.0(11.15.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 3600 Contact: <sip:10101@10.18.130.50:5060> ;expires=3600 Content-Length: 0 Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Monday, March 16, 2015 7:01:54 AM Subject: Re: [SR-Users] remove_hf_value Hello, can you give an example of incoming header and how you want to be after modification? It is not clear for me what end result you want/expect. Cheers, Daniel On 14/03/15 06:41, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, Here updated version. Trying remove Record-Route after the coma. My log shows Mar 14 01:25:21 canlvprx01 /usr/sbin/kamailio[751]: INFO: <script>: [SUBSCRIBE], Inbound connection from <Client Public IP> Mar 14 01:25:21 canlvprx01 /usr/sbin/kamailio[751]: ERROR: <script>: <null> Mar 14 01:25:21 canlvprx01 /usr/sbin/kamailio[751]: INFO: <script>: [SUBSCRIBE], Route for removal: '<null>' route snippet. route[ROUTE_HDR_ORG] { # Getting record routes if(is_present_hf("Route")) { xlog("L_INFO", "[$rm], Inbound connection from <$si>\n"); if(is_method("SUBSCRIBE")) { if(is_ip_rfc1918($si)) { xlog( "$sel((a)hf_value.Record_Route[-1])" ); xlog("L_INFO", "[$rm], Route for removal: '$sel((a)hf_value.Record_Route[-1])'\n"); remove_hf_value("Record-Route[-1]"); } else { xlog( "$sel((a)hf_value.Record_Route[1])" ); xlog("L_INFO", "[$rm], Route for removal: '$sel((a)hf_value.Record_Route[1])'\n"); remove_hf_value("Record-Route[1]"); } } } return; } Slava, From: "Slava Bendersky" <volga629(a)networklab.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, March 13, 2015 1:00:44 PM Subject: [SR-Users] remove_hf_value Hello Everyone, I am trying adjust Record-Route, but can't match headers value And I am call route(ROUTE_HDR_ORG); right after route(WITHINDLG); route[ROUTE_HDR_ORG] { # Getting record routes if(is_present_hf("Record-Route")) { if(is_method("SUBSCRIBE")) { xlog( "$sel((a)hf_value.Record_Route[1])" ); xlog("L_INFO", "[$rm], Route for removal: '$sel((a)hf_value.Record_Route[1])'\n"); remove_hf_value("Record-Route[1]"); } } return; } Any help Thank you, Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

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Presence data to users behind NAT
by Olli Attila 22 Mar '15

22 Mar '15

Hello, Im running kamailio 4.2.3 and I'm trying to build a system where I could have the following features: SIP proxy, RTP proxy, Presence & XCAP service for IM messaging. I have a problem with the Presence service with users that are located behind nat. When Kamailio sends presence info to a watcher with sip NOTIFY message, the destaination ip for the pacage seems to be an un-routable (=private) ip, and so the watcher never receives the NOTIFY message making the watchers sip client unaware of the other users presence. Any suggestions how to make Kamailio handle the watchers behind nat so that the NOTIFY messages will get delivered succesfully? I tried the scenario with a client with public ip and the presence works fine. my kamailio.cfg: http://tny.cz/7d85773c -- Olli

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Kamailio with websocket as transport
by Austin Einter 22 Mar '15

22 Mar '15

Dear All I have a sip user agent. Today it works fine with Kamailio using tcp as tranasport. Now I am planning to support websocket in my sip ua. My question is does Kamailio support websocket transport. If so how can I configure it or while building kamailio do I need to make some flags on. I am using latest Kamailio 4.2.3 (version: kamailio 4.2.3 (x86_64/linux)). Thanks Austin

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Hardware config to run 1000 Kamailio TLS users
by elektau＠yandex.ru 20 Mar '15

20 Mar '15

Hi All, 1. What are resource requirements for running Kamailio server with TLS module active for 1000 registered users (concurrent calls) when running on VPS? In other words, what amount of RAM and processor power required? 2. What is the maximum number of Kamailio TLS users/concurrent calls can be made for the below default values? # Amount of shared and private memory to allocate # for the running Kamailio server (in Mb) SHM_MEMORY=64 PKG_MEMORY=8 Thanks.

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Kamailio hardware requirements
by Mickael Marrache 20 Mar '15

20 Mar '15

Hi, We are currently deploying an entire architecture composed of load balancers, proxies and media relays. All the components except the media relays are Kamailio instances. The media relays are RTP proxy instances. We are trying to determine the hardware requirements for the different servers. We will start with an architecture composed of: . 2 load balancers (one kept as backup to ensure availability) . 2 proxies (load is balanced over the 2 using the DISPATCHER module) . 2 media relays We are trying to determine the recommended hardware for the different components taking into account we will need to serve around 400000 users. How many registrations per second can Kamailio support? How many concurrent calls can Kamailio support? What is the recommended number of children processes? What is the recommended size for PKG memory? What is the recommended size for SHM memory? Thanks for your help, Mickael

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auth_diameter module
by Grant Bagdasarian 19 Mar '15

19 Mar '15

Hello, Does Kamailio still support authentication using the auth_diameter module? There is a note on the modules page stating the module is now obsolete. "NOTE: diameter support was developed for DISC (DIameter Server Client project at http://developer.berlios.de/projects/disc/). This project seems to be no longer maintained and DIAMETER specifications were updated in the meantime. Thus, the module is obsolete and needs rework to be usable with opendiameter or other DIAMETER servers." Is the page referring to this module or the DISC? Are there perhaps any other ways to do authentication with Kamailio using diameter?

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RTPProxy issue?
by Igor Potjevlesch 19 Mar '15

19 Mar '15

Hello, I discovered an issue related to the handling of "timestamp" and/or "Marker bit" with rtpproxy (I use the latest Extension 20081224). The call-flow is the following: one UA places a call to A and put this call on hold. Then, the same UA call another number B. Individual streams are ok. When the UA tries to transfer A with B, the RTPProxy receive a RTP packet with a huge timestamp and the Marker bit set to "True". Just after this RTP packet, RTPProxy stop forward the RTP packets from A to B. B to C is still working. Anyone have an idea? Regards, Igor.

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Sending PUSH notification asynchronously
by Mickael Marrache 19 Mar '15

19 Mar '15

Hi, I'm trying to add PUSH support to my system using Kamailio. When a call is received and the callee has no active registration, a PUSH is sent to the callee endpoint. In order to send this PUSH, my proxy sends an HTTP requests to another server that will send the PUSH. So, my route block is as follows: route[PUSH] { #Send PUSH notification using a custom module I wrote #Suspend transaction execution. t_on_failure("MANAGE_FAILURE"); if(!t_suspend()) { send_reply("503", "Service Unavailable"); exit; } $sht(push=>join::$rU) = "" + $T(id_index) + ":" + $T(id_label); exit; } I would like to execute this route block asynchronously (i.e. in a separate process). I tried using the async_task_route command, and I can see the route block is execute in a separate process, however some stuffs are broken (e.g. transaction timeout, failure route). So, I thought it may be related to the fact asynchronous execution is implemented using t_suspend/t_continue and in this case, t_suspend/t_continue would be called twice. The important point is to make the interaction with the PUSH server asynchronous. Any idea? Thanks, Mickael

2 6

Timeout after t_suspend and failure route
by Mickael Marrache 19 Mar '15

19 Mar '15

Hi, When receiving an INVITE, I arm a failure route and set the fr_timer to 30 seconds. After the fr_timer expires, I need to do some operations however I see the failure route is not executed. I guess it is related to the fact the INVITE has not been forwarded? If yes, how can I execute code when a transaction times out after calling t_suspend? Thanks, Mickael

2 7

missing BYE when 2 redundant kamailio servers share the same database
by Ding Ma 19 Mar '15

19 Mar '15

This is the SIP log when using location routing for BYE from peer server.

1 0

missing BYE when 2 redundant kamailio servers share the same database
by Ding Ma 19 Mar '15

19 Mar '15

This is the SIP log with the 500 error for BYE.

1 0

SQLOPS bug
by Mihaly Zachar 18 Mar '15

18 Mar '15

Hi Gents, Please consider applying this patch: --- ./modules/sqlops/sql_api.c.orig 2015-03-18 16:33:56.994320505 +0100 +++ ./modules/sqlops/sql_api.c 2015-03-18 16:34:11.254419341 +0100 @@ -743,7 +743,7 @@ LM_ERR("row index out of bounds [%d/%d]\n", i, res->nrows); goto error; } - if(i>=res->ncols) + if(j>=res->ncols) { LM_ERR("column index out of bounds [%d/%d]\n", j, res->ncols); goto error; Regards, Misi

2 3

missing BYE when 2 redundant kamailio servers share the same database
by Ding Ma 18 Mar '15

18 Mar '15

Hi, all I'm trying to set up 2 kamailio servers for active-active redundancy. The two kamailio severs share the the same database with db_mode=3, and no registration replication. Use pjsua2 as SIP client for testing. The test setup is as follows: kamailio server 1(k1): 10.0.1.30:5061 kamailio server 2(k2): 10.0.1.32:5061 sip client c1: sip:16317@10.0.1.30, client ip: 10.0.1.254 sip client c2: sip:72316@10.0.1.30, client ip: 10.0.1.254 c1 is registered with k1. c2 is registered with k2. When c1 calls c2, the call flow is as follows: INVITE: c1------>k1------>k2------>c2 200 OK: c1<------k1<------k2<------c2 So far, the INVITE works. c1 and c2 can establish a call. However, when c2 hangs up and sends BYE, the BYE is forwarded all the way to k1, and then k1 gives a 500 server error. The log on k1 seems to indicate k1 was attempting to create a TLS connection to itself instead of using the existing TLS connection to c1 to forward the BYE. BYE: c1 X k1<------k2<------c2 | |------->Connection refused/500 Here is the kamailio log on k1: Mar 16 10:34:21 kamailio.sip.com /usr/sbin/kamailio[13396]: ERROR: <core> [tcp_main.c:2740]: tcpconn_1st_send(): connect 10.0.1.30:55999 failed (RST) Connection refused Mar 16 10:34:21 kamailio.sip.com /usr/sbin/kamailio[13396]: ERROR: <core> [tcp_main.c:2750]: tcpconn_1st_send(): 10.0.1.30:55999: connect & send for 0x7fc96a68a1a0 failed: Connection refused (111) This is the routing logic for BYE in kamailio.cfg: if (loose_route()) { route(DLGURI); if (is_method("BYE")) { xlog("L_DBG", "=====BYE $ru from $fu $si:$sp to $du=====\n"); dlg_manage(); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } else {...... If I add location based routing for BYE from peer kamailio, then the BYE seems to be forwarded correctly: if (loose_route()) { route(DLGURI); if (is_method("BYE")) { xlog("L_DBG", "=====BYE $ru from $fu $si:$sp to $du=====\n"); dlg_manage(); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails # If BYE coming from Kamailio peer, route BYE by location $var(peerlist)=$sel(cfg_get.trusted.peers); $var(i) = 0; while($var(i)<$(var(peerlist){param.count})) { xlog("L_DBG", "=====$(var(peerlist){param.count})=====$(var(peerlist){param.valueat,$var(i)})=====\n"); if(src_ip==$(var(peerlist){param.valueat,$var(i)})) { lookup("location"); xlog("L_DBG", "=====BYE from $fu $si:$sp to $du=====\n"); break; } $var(i) = $var(i) + 1; } } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } else {...... My first question is why k1 loose_route sends the BYE to itself instead of the client. Is this a bug? My next question is whether the above location routing for BYE from peer kamailio a good/safe approach. The SIP traces are enclosed. Thanks

1 0

Kamailio and Busy Lamp Fields (BLF)
by Daniel-Constantin Mierla 18 Mar '15

18 Mar '15

Hello, a bit of time ago I played with Kamailio and BLF, therefore I published kind of getting started configuration file for this feature. It is available at: - http://kb.asipto.com/kamailio:presence:k43-blf The plan was to add more to the tutorial, some guidelines for configuring the phones, but the time didn't allow it so far. Keep an eye on it for some new content. Contributions are also welcome, send them to me or take the content and publish it with new additions on another place. Hopefully is going to be useful for some people out there. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

1 0

siremis login issue
by Boris Ivanov 18 Mar '15

18 Mar '15

Hi Hope this is right place to post my question. I have installed siremis under FreeBSD 10 It went through well. DB created with /usr/local/sbin/kamdbctl create. Problem happen when I trying to login as admin. [{"target":"ERROR","content":" \n[2015-03-11 17:31:37 (GMT)] An exception occurred while executing this script: \nError message: #8192, Non-static method BizSystem::getConfiguration() should not be called statically, assuming $this from incompatible context<\/font> \nScript name and line number of error:\/usr\/local\/www\/apache24\/data\/openbiz\/bin\/data\/BizDataObj_Abstract.php:268<\/font> \n function:<\/b> errorHandler ( 8192, \"Non-static method BizSystem::getConfiguration() should not be ca...\", \"\/usr\/local\/www\/apache24\/data\/openbiz\/bin\/data\/BizDataObj_Abstrac...\", 268, Array(0) ) @ \/usr\/local\/www\/apache24\/data\/openbiz\/bin\/sysheader.inc 117\n function:<\/b> userErrorHandler ( 8192, \"Non-static method BizSystem::getConfiguration() should not be ca...\", \"\/usr\/local\/www\/apache24\/data\/openbiz\/bin\/data\/BizDataObj_Abstrac...\", 268, Array(0) ) @ \/usr\/local\/www\/apache24\/data\/openbiz\/bin\/data\/BizDataObj_Abstract.php 268\n function:<\/b> setQuoteIdentifiers ( ) @ \/usr\/local\/www\/apache24\/data\/openbiz\/bin\/data\/BizDataObj_Abstract.php 177\n function:<\/b> __construct ( Array(1) ) @ \/usr\/local\/www\/apache24\/data\/openbiz\/bin\/data\/BizDataObj.php 37\n function:<\/b> __construct ( Array(1) ) @ \/usr\/local\/www\/apache24\/data\/openbiz\/bin\/ObjectFactory.php 166\n function:<\/b> constructObject ( \"system.do.UserDO\" ) @ \/usr\/local\/www\/apache24\/data\/openbiz\/bin\/ObjectFactory.php 56\n function:<\/b> getObject ( \"system.do.UserDO\", 0 ) @ \/usr\/local\/www\/apache24\/data\/openbiz\/bin\/BizSystem.php 252\n function:<\/b> getObject ( \"system.do.UserDO\" ) @ \/usr\/local\/www\/apache24\/data\/siremis\/modules\/service\/authService.php 92\n function:<\/b> authDBUser ( \"admin\", \"admin\" ) @ \/usr\/local\/www\/apache24\/data\/siremis\/modules\/service\/authService.php 66\n function:<\/b> authenticateUser ( \"admin\", \"admin\" ) @ \/usr\/local\/www\/apache24\/data\/siremis\/modules\/user\/form\/LoginForm.php 92\n function:<\/b> Login ( ) @ \/usr\/local\/www\/apache24\/data\/openbiz\/bin\/BizController.php 310\n function:<\/b> invoke ( ) @ \/usr\/local\/www\/apache24\/data\/openbiz\/bin\/BizController.php 110\n function:<\/b> dispatchRequest ( ) @ \/usr\/local\/www\/apache24\/data\/openbiz\/bin\/BizController.php 32\n function:<\/b> include_once ( \"\/usr\/local\/www\/apache24\/data\/openbiz\/bin\/BizController.php\" ) @ \/usr\/local\/www\/apache24\/data\/siremis\/bin\/controller.php 6<\/div>\n _____ Please ask system administrator for help...<\/div>\n"}] Anyone knows how to fix? Boris ____________________________________________________________ This message and any attachments contain confidential information and must not be disclosed to a third party without the prior written consent of setcom. If you have received it erroneously, please discard it and contact us by return. ____________________________________________________________

2 5

Redirect server for load balancing and HA
by Markus 18 Mar '15

18 Mar '15

Hi list, I'm new here, also to Kamailio. My goal is to achieve the "standard solution": Kamailio for load balancing and high availability of several Asterisk servers. For starters, I've set it up with Kamailio+dispatcher+permissions acting as a proxy and it was working fine, but I'd like to try to keep it much more simple. For these reasons: 1) I'd like to maintain different peer settings in Asterisk (DTMF, codecs). 2) I don't want/need a database for Kamailio. I think it is mandatory for permissions module + IP auth. So, let's do Asterisk the authentication, it's already there. 3) Accounting. I'm using a software where it's really useful for accounting purposes if every peer/user has its own entry in Asterisk's sip.conf / in realtime table. Long story short, my config now looks like this and it seems to be working fine for me: request_route { route(DISPATCHER); } route[DISPATCHER] { if ( method=="REGISTER") { ds_select_domain("1","4"); sl_send_reply("300","Redirect"); exit; } if ( method=="INVITE" ) { ds_select_domain("1","4"); sl_send_reply("300","Redirect"); exit; } if ( method=="SUBSCRIBE" ) { ds_select_domain("1","4"); sl_send_reply("300","Redirect"); exit; } } My questions: 1) There are other approaches to the DTMF+codecs issue. Like this thread from 2013: http://lists.kamailio.org/pipermail/sr-users/2013-January/076379.html - but I have dozens of changing PSTN peers, so authentication/accounting based on port would be cumbersome. Are there new/alternative ways? 2) I searched the archive and the Web but did not find many questions about a "I-don't-want-Kamailio-to-show-as-the-only-peer-to-Asterisk" solution. Therefore I'm wondering: Is everyone a) running such homogeneous setups that they can survive with the same codec + DTMF settings in Asterisk for all their peers + users? Or b) doing some rewriting magic within Kamailio to deal with it? Or c) I'm overlooking a simple solution. Which one is it? :) 3) And the biggest question: Everywhere I read basically "Redirect server for LB + HA is not recommended, better use proxy". Is this still true, and if so, what are the reasons? Because I should be worried that not all clients/gateways out there respect a SIP 300 Redirect? Other reasons? Thanks so much! :) Markus

4 8

msilo clarification
by mayamatakeshi 18 Mar '15

18 Mar '15

Hello, maybe this phrase in the msilo doc is misleading: *Every time when a user registers with Kamailio, the module is looking in database for offline messages intended for that user. All of them will be sent to contact address provided in REGISTER request.* http://kamailio.org/docs/modules/4.2.x/modules/msilo.html The above seems to imply that msilo would send the call directly to the contact address. But this doesn't seem to be the case based on what is mentioned in several threads in the mailing list and in what I saw in the msilo.c code. If so, is there a reason to be a such? Would it be a problem to allow msilo m_dump to get the values of contact, received and path and use them when calling t_request? Regards, Takeshi.

2 3

Re: [SR-Users] remove_hf_value
by Slava Bendersky 18 Mar '15

18 Mar '15

Hello Daniel, I want do clean up on routes from first SUBSCRIBE, that when traffic goes to public side of kamailio will be only public route and when traffic is internal the Record-Route will stay internal only. In SDP I see like this. I tired double routes set 0 in rr, but didn't clean enough. <--- Transmitting (NAT) to 10.18.130.46:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.18.130.46;branch=z9hG4bKf852.e0223f39c2bbad8366fdf1b7cb22b336.0;i=8;received=10.18.130.46;rport=5060 Via: SIP/2.0/TCP 192.168.88.252:5062;received=client_pub_ipbranch=z9hG4bK0bbe1f7d27257bba9;rport=5062 Record-Route: <sip:10.18.130.46;r2=on;lr=on;ftag=a185d974ec;nat=yes> On public side remote this Record-Route Record-Route: <sip:kamailio_pub_ip:5084;transport=tcp;r2=on;lr=on;ftag=a185d974ec;nat=yes> On private side remote this Record-Route From: "Slava Bendersky" <sips:10101@networklab.ca>;tag=a185d974ec To: <sips:10101@networklab.ca>;tag=as00757d3e Call-ID: b08adb1ad1804a83 CSeq: 236711034 SUBSCRIBE Server: FPBX-2.11.0(11.15.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 3600 Contact: <sip:10101@10.18.130.50:5060>;expires=3600 Content-Length: 0 Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Monday, March 16, 2015 7:01:54 AM Subject: Re: [SR-Users] remove_hf_value Hello, can you give an example of incoming header and how you want to be after modification? It is not clear for me what end result you want/expect. Cheers, Daniel On 14/03/15 06:41, Slava Bendersky wrote: Hello Everyone, Here updated version. Trying remove Record-Route after the coma. My log shows Mar 14 01:25:21 canlvprx01 /usr/sbin/kamailio[751]: INFO: <script>: [SUBSCRIBE], Inbound connection from <Client Public IP> Mar 14 01:25:21 canlvprx01 /usr/sbin/kamailio[751]: ERROR: <script>: <null> Mar 14 01:25:21 canlvprx01 /usr/sbin/kamailio[751]: INFO: <script>: [SUBSCRIBE], Route for removal: '<null>' route snippet. route[ROUTE_HDR_ORG] { # Getting record routes if(is_present_hf("Route")) { xlog("L_INFO", "[$rm], Inbound connection from <$si>\n"); if(is_method("SUBSCRIBE")) { if(is_ip_rfc1918($si)) { xlog( "$sel((a)hf_value.Record_Route[-1])" ); xlog("L_INFO", "[$rm], Route for removal: '$sel((a)hf_value.Record_Route[-1])'\n"); remove_hf_value("Record-Route[-1]"); } else { xlog( "$sel((a)hf_value.Record_Route[1])" ); xlog("L_INFO", "[$rm], Route for removal: '$sel((a)hf_value.Record_Route[1])'\n"); remove_hf_value("Record-Route[1]"); } } } return; } Slava, From: "Slava Bendersky" <volga629(a)networklab.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, March 13, 2015 1:00:44 PM Subject: [SR-Users] remove_hf_value Hello Everyone, I am trying adjust Record-Route, but can't match headers value And I am call route(ROUTE_HDR_ORG); right after route(WITHINDLG); route[ROUTE_HDR_ORG] { # Getting record routes if(is_present_hf("Record-Route")) { if(is_method("SUBSCRIBE")) { xlog( "$sel((a)hf_value.Record_Route[1])" ); xlog("L_INFO", "[$rm], Route for removal: '$sel((a)hf_value.Record_Route[1])'\n"); remove_hf_value("Record-Route[1]"); } } return; } Any help Thank you, Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Dialogs not deleted on BYE
by Shane Harrison 18 Mar '15

18 Mar '15

Hi all, Have found a few threads on this before which have clarified my understanding a little but not yet presented me with a solution. I am using dialog_ng. dlg_manage() is deprecated and as per the documentation, I am simply setting a dialog flag early in the request_route. Kamctl dialog show does indeed show the dialogs are tracked, but they are not destroyed on the receipt of the BYE. Dialog ref count remains at 2 and in state=4 Log shows the following Mar 16 22:43:42 hh-rcs-sipproxy3 /usr/sbin/kamailio[15484]: DEBUG: tm [t_lookup.c:1373]: t_newtran(): DEBUG: t_newtran: msg id=16 , global msg id=15 , T on entrance=(nil) Mar 16 22:43:42 hh-rcs-sipproxy3 /usr/sbin/kamailio[15484]: DEBUG: tm [t_lookup.c:527]: t_lookup_request(): t_lookup_request: start searching: hash=5943, isACK=0 Mar 16 22:43:42 hh-rcs-sipproxy3 /usr/sbin/kamailio[15484]: DEBUG: tm [t_lookup.c:485]: matching_3261(): DEBUG: RFC3261 transaction matching failed Mar 16 22:43:42 hh-rcs-sipproxy3 /usr/sbin/kamailio[15484]: DEBUG: tm [t_lookup.c:709]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found Mar 16 22:43:42 hh-rcs-sipproxy3 /usr/sbin/kamailio[15484]: DEBUG: tm [t_hooks.c:380]: run_reqin_callbacks_internal(): DBG: trans=0x7fc8d79af798, callback type 1, id 0 entered Mar 16 22:43:42 hh-rcs-sipproxy3 /usr/sbin/kamailio[15484]: DEBUG: dialog_ng [dlg_handlers.c:855]: dlg_new_dialog(): starting dlg_new_dialog and method is [BYE] Looks to me as if fails to find a dialog match and actually creates a new dialog..... Dialog_ng only has one matching mode - DID, others have been deprecated. I have captured the outgoing INVITE and there is no dialog_id parameter added to the record_route. Is that causing my matching problem? Record-Route: <sip:64.208.160.211;transport=tcp;lr=on;ftag=5419eecb;nat=yes> Record-Route: <sip:203.171.39.49:5061;transport=tls;lr=on> Cheers Shane Shane Harrison Senior Software Engineer Imagination Technologies NZ Limited Level 2 1 Market Grove Lower Hutt, 5010 New Zealand PO Box 30-449 Lower Hutt, 5040 New Zealand Phone: +64 4 890-3681 ext 3361

2 10

PRACK not sent
by Dharm S 17 Mar '15

17 Mar '15

Hi All, I am using kamailio 3.1.5. I am trying to send the PRACK message but it sends back 404 Not Here. I am new to using this. I am adding details in the configuration file below: I guess route[WITHINDLG] ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openser:openserrw@localhost/openser" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* 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:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif ####### Modules Section ######## # set paths to location of modules #!ifdef LOCAL_TEST_RUN mpath="modules_k:modules" #!else mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- 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) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); # NAT detection route(NAT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # 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 } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # 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 to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif 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() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_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; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } }

2 1

xlog from 1.5 to 4.2
by Juan Perez 17 Mar '15

17 Mar '15

I installed 4.2 and I am using the same config file form my current version 1.5.2, but none of my custom logs (xlog) are being printed in the log file, even after rising the debug level, the only messages I see are DEBUG these are my settings: loadmodule "xlog.so" # ----- xlog parms ------ modparam("xlog", "buf_size", 4096) Sample uses:xlog("L_INFO","mylog: Main logic started.\n"); xlog("L_INFO","mylog: Switchover Activated!\n"); xlog("L_INFO","mylog: Redundancy test, last 11 digits [$var(number)].\n"); xlog("L_INFO","mylog: Too Many Hops.\n"); am I missing something ?

2 1

PKG and Shared Memory statistics
by Dirk Teurlings Signet B.V. 17 Mar '15

17 Mar '15

Hi, We'd like to be able to get statistics of the used amount of memory in Kamailio. # kamctl stats shmem The above command only tells us what the shared memory is doing. Is there any way to get the committed size of memory for each of the PKG_MEMORY pools? If not, could this be added at some point? Kind regards, -- Dirk Teurlings

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