sr-users March 2015

sr-users@lists.kamailio.org

88 participants
149 discussions

ERROR: sipcapture [hep.c:139]: hepv2_received(): ERROR: sipcapture:hep_msg_received: unknow protocol [1]
by Satish Patel 24 Mar '15

24 Mar '15

Hello, Recently i have installed Homer capture server with kamailio. It works! but in logs i am seeing many error like following, I am using Opensips 2.1 siptrace server to send all trace to Homer. ERROR: sipcapture [hep.c:139]: hepv2_received(): ERROR: sipcapture:hep_msg_received: unknow protocol [1] ERROR: sipcapture [hep.c:139]: hepv2_received(): ERROR: sipcapture:hep_msg_received: unknow protocol [1] ERROR: sipcapture [hep.c:139]: hepv2_received(): ERROR: sipcapture:hep_msg_received: unknow protocol [1]

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Compiling with db_unixodbc
by Grant Bagdasarian 24 Mar '15

24 Mar '15

Hello, I'm trying to compile Kamailio 4.2.3 on Debian 7.8 with db_unixodbc but the sql.h file can't be found. I've used the following command to compile and install unixODBC. tar -zxvf unixODBC-2.3.2.tar.gz && cd unixODBC-2.3.2 && ./configure --sysconfdir=/etc --prefix=/usr/local/unixODBC --disable-gui && make && make install #####################: locate odbc_config /usr/local/unixODBC/bin/odbc_config /usr/src/unixODBC-2.3.2/exe/odbc_config #####################: locate sql.h /usr/local/unixODBC/include/sql.h /usr/src/kamailio-4.2.3/modules/db_mysql/km_db_mysql.h /usr/src/kamailio-4.2.3/modules/db_postgres/pg_sql.h /usr/src/unixODBC-2.3.2/Drivers/Postgre7.1/isql.h /usr/src/unixODBC-2.3.2/Drivers/nn/isql.h /usr/src/unixODBC-2.3.2/Drivers/nn/nnsql.h /usr/src/unixODBC-2.3.2/exe/isql.h /usr/src/unixODBC-2.3.2/include/sql.h Kamailio Compile Error CC (gcc) [M db_unixodbc.so] connection.o In file included from connection.c:34:0: connection.h:42:17: fatal error: sql.h: No such file or directory compilation terminated. make[1]: *** [connection.o] Error 1 make: *** [modules] Error 1 Modules.lst include_modules= carrierroute db_unixodbc dialog_ng lcr tls uuid Removed these includes from the exclude_modules Any ideas?

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Inter Kamailio communication: UDP vs TCP vs SCTP
by Mickael Marrache 24 Mar '15

24 Mar '15

Hi, We are deploying a VoIP system based on Kamailio. The system is composed of load balancers, proxies, registrars, etc What is the recommended transport protocol to use for internal communication (i.e. load balancer <-> proxy/registrar)? We started with UDP but we also see that Kamailio supports TCP and SCTP. For information, at first stage, the UAs will communicate using UDP but we may need to use TCP in case messages are too large (we had such issues when using ICE). Thanks, Mickael

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

23 Mar '15

Hello Everyone, In my case I am using Path and I asterisk generate NOTIFY with route based on path, then I see on kamailio side tries reach on client on public IP side and open new connection not follow ESTABLISHED/RELATED link then it times out. Log: Mar 23 11:00:17 canlvprx01 /usr/sbin/kamailio[20193]: INFO: <script>: [NOTIFY], Inbound connection from <10.18.130.51> traffic from Asterisk Mar 23 11:00:17 canlvprx01 /usr/sbin/kamailio[20193]: INFO: <script>: [NOTIFY], Present Route: Client_public_IP. Mar 23 11:00:28 canlvprx01 /usr/sbin/kamailio[20204]: ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect Client_public_IP:5060 failed (timeout) Here trace on kamailio side http://fpaste.org/201813/15287114/ Here trace from soft phone client side. http://fpaste.org/201814/14271530/ Slava. From: "Alex Balashov" <abalashov(a)evaristesys.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Monday, March 23, 2015 6:42:59 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Anthony, The Contact presented by the subscriber in the initial subscription is: m: "Test User" <sip:172.16.4.7;line=sr-D8G7CE2.5PUeK-xuarl7NYDdNYDxNYlFUYoeUeQ8Cw.6DE2vDdyJDAa4TliwC84OC82LK-2ehwl7NYDdNYDxNYlFUAm6UYzm0gme>;+sip.ice And the subsequent NOTIFY is correctly targeted to this RURI: NOTIFY sip:172.16.4.7;line=sr-D8G7CE2.5PUeK-xuarl7NYDdNYDxNYlFUYoeUeQ8Cw.6DE2vDdyJDAa4TliwC84OC82LK-2ehwl7NYDdNYDxNYlFUAm6UYzm0gme SIP/2.0 The subscriber returns this Contact in the 200 OK response for the NOTIFY: m: "Test User" <sip:172.16.4.7;line=sr-D8G7CE2.5PUeK-xuarl7NYDdNYDxNYlFUYoeUeQ8Cw.6DE2vDdyJDAa4TliwC84O>;+sip.ice And subsequent NOTIFYs appear to use this RURI instead of the original Contact of the subscriber: NOTIFY sip:172.16.4.7;line=sr-D8G7CE2.5PUeK-xuarl7NYDdNYDxNYlFUYoeUeQ8Cw.6DE2vDdyJDAa4TliwC84O SIP/2.0 NOTIFY sip:172.16.4.7;line=sr-D8G7CE2.5PUeK-xuarl7NYDdNYDxNYlFUYoeUeQ8Cw.6DE2vDdyJDAa4TliwC84O SIP/2.0 etc. A SUBSCRIBE is a dialog-creating event, so in-dialog messages (e.g. NOTIFYs) should be targeted at a Request URI corresponding to the Contact URI of the SUBSCRIBE-er. It seems to me that this is the problem here, rather than truncation. -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.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

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Kamalio call issue
by Yogendra Gupta 23 Mar '15

23 Mar '15

Hello, We have setup kamalio at server and it is working at some DNS with call/chat. At some IP chat is working but when we call then it is not working. At some DNS(Net IP) it is working . What can be problem at server? Following is login for SIP test user: tester1(a)23.253.110.48 123456 Host IP : 23.253.110.48 I have attached mu kamalio config file. Let me know what we need to change at config file or server Thanks

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What is the best SIP trunk authentication strategy
by canuck15 23 Mar '15

23 Mar '15

Been struggling with this for awhile now. So far I am finding it rather difficult to come up with way to authenticate SIP trunks taking into account all possible scenarios. My setup is Kamailio combined with Asterisk realtime. Everything is in a MySQL database. All authentication is done by Kamailio. Kamailio handles SIP extension (user) authentication quite well but seems poorly equipped for SIP trunk (peer) authentication. The biggest problem I have run into so far is that SIP trunks typically use DNS names and Kamailio is not at all designed to used DNS names. Also SIP trunks do not use named "realms" whereas that is a key part of Kamailio authentication. For IP authentication about the only solution I have found is to DNS lookup and save all returned IP addresses in the ip_addr mysql table. Then I do: #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address() && $au == "") { # source IP allowed return; } I can run a cron job every hour to DNS lookup and update the ip_addr table as needed so I think this is a satisfactory solution for IP authentication. SIP trunk user/pass authentication is the one I am now struggling with. This standard Kamailio authentication section does not work for SIP trunks. if (is_method("REGISTER|INVITE") || from_uri==myself) { # authenticate requests #if (!auth_check("$fd", "subscriber", "1")) { if (!auth_check("$fd", "subscriber", "0")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); This is a multidomain setup and therefore: modparam("auth_db", "use_domain", 1) So the authentication section will try authenticate the realm of the trunk which will always be an IP address. However the SIP trunk realm will usually be saved in the "subscriber" database as the FQDN. There is no mechanism for auth_check() to directly or indirectly check against FQDN. The other problem is that even if I used the ip_addr table somehow it will only try use the user/pass of the first instance of a matching IP address. For SIP trunks it is possible I could have multiple subaccounts and therefore multiple instances of the same IP with different user/pass. So auth_check() is not designed for being used this way as far as I can tell. Seems to me like there should almost be a special kamailio module just for SIP trunks. I had a look a carrierroute module it's not designed for this either.

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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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sr-users March 2015