June 2013 - sr-users - kamailio.org

[Serusers] Digest Authentication with Tenor DX
by Ashutosh Kumar 29 Apr '25

29 Apr '25

Hi, Does someone have any idea/experience with Quintum DX radius server's support for digest mode authentication. The Access-Request packets sent by SER are getting an error back. Guide. Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

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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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Cannot hear voice with symmetric NAT and STUN
by Khoa Pham 08 Feb '17

08 Feb '17

Hi, I have Kamailio as SIP server and RTP server. Client is PJSIP. I read that STUN is for non-symmetric NAT, and RTP server is for symmetric NAT. Supposed A calls B. If A, B both use symmetric NAT and STUN, they cannot hear each other If A or B use non-symmetric NAT and NOT using STUN, they cannot hear each other. Why is that? I read http://tools.ietf.org/id/draft-takeda-symmetric-nat-traversal-00.txt for Prediction Failure, is that related to this problem ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

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kamailio 4.0.1 websocket configuration
by Rupayan Dutta 28 Dec '15

28 Dec '15

Hi All, I have build Kamailio 4.0.1 from source in CentOS 5.8(i386 architecture).I followed all instructions from http://www.kamailio.org/wiki/install/4.0.x/git.(Though Modules_k directory is not generated).I then edit kamailio.config file for websocket support as described in webocket.cfg file in exmples directory.But while starting kamailio it gives following error ERROR: load_module: could not open module : libunistring.so.0: cannot open shared object file: No such file or directory 0(30270) : [cfg.y:3567]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 318, column 12-57: failed to load module I checked websocket.so file in the specified directory and it is already there.Can you please help me what's wrong with it? please help. Rupayan Dutta

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Kamailio - Asterisk, problem with SIP trunk mode
by Mickael MONSIEUR 02 Dec '15

02 Dec '15

Hello, First, i have 7 years experience with Asterisk, but I started a project with Kamailio, forgive me in advance if I say silly things...! ;-) I set up a classic Asterisk / Kamailio configuration: sip phones -> kamailio -> asterisk -> sip trunks/pstn. When a call comes from the PSTN side, if I configure Asterisk as follows: [012345678] type = friend username = 012345678 secret = xxxxxxx host = dynamic fromdomain = sip.mydomain.com fromuser = 012345678 Standard mode: exten => 012345678, 1, Dial(SIP/012345678) -> The call is redirected on the phone by Kamailio ! :-) ------------------------------------------------------------------------------------------------------------------------------------------------ Trunk mode: [mytrunk] type = friend username = mytrunkUser secret = xxxxxxx host = dynamic fromdomain = sip.mydomain.com fromuser = mytrunkUser exten => 012345678, 1, Dial(SIP/mytrunk/012345678) -> The call is rejected by Kamailio.... exten => 012345679, 1, Dial(SIP/mytrunk/012345679) -> The call is rejected by Kamailio .... My question is how to allow the routing of multiple numbers (trunk mode) in a SIP account with Kamailio? Best regards, Mickael

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[Kamailio-Users] HA+DRBD setup
by Siddhardha Garige 09 Jul '15

09 Jul '15

Hello all, Just wondering if anyone know any tutorial on setting up HA+DRBD solution for kamailio. Especially creating partitions, DRBD devices and mount points. Thanks in advance, -Sid "May the light be with you." ______________________________________________ Siddhardha Garige www.luminepixels.com

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Kamailio: MI Command: 'trusted_dump' and 'trusted_reload'
by José Luis Millán 22 Apr '15

22 Apr '15

Hi, I'm using 'allow_trusted()' from Permissions module without a problem in a Kamailio 4.0 routing logic. It does just work, but the MI command 'trusted_dump' tells me that the module is not in use: ''' server:/home/jmillan# kamctl fifo trusted_dump 500 Trusted-module not in use ''' Doing a 'trusted_reload' does neither make any query to database to retrieve the data. Other MI commands of same module do work correctly. ie: 'address_reload' or 'address_dump' Note: I have tested in a Kamailio 1.5 and the result is the same. Regards. -- José Luis Millán

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Re: [Kamailio-Users] /usr/local/sbin/kamdbctl create error
by vivi 01 Nov '14

01 Nov '14

Hi all.. I have solve the problem . Cheers, vivi From: vivi [mailto:vivi.hilton@gmail.com] Sent: Saturday, June 06, 2009 11:29 PM To: 'users(a)lists.kamailio.org' Subject: /usr/local/sbin/kamdbctl create error Hi all I have specify the wanted db type (DBENGINE=MYSQL) in the /usr/local/etc/kamailio/kamctlrc then I using "/usr/local/sbin/kamdbctl create" to create MySQL database, but I got this error: ERROR: database engine not specified, please setup one in the config script root@acer:/usr/local/src/kamailio-1.5.0/sip-server# vim /usr/local/etc/kamailio/kamctlrc root@acer:/usr/local/src/kamailio-1.5.0/sip-server# /usr/local/sbin/kamdbctl create MySQL password for root: INFO: test server charset ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) Usage: grep [OPTION]... PATTERN [FILE]... Try `grep --help' for more information. /usr/local/lib/kamailio/kamctl/kamdbctl.mysql: line 112: [: =: unary operator expected INFO: creating database openser ... ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) ERROR: Creating core database and grant privileges failed! Cheers, vivi

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kamailio and app_python
by Victor Seva 11 Jun '14

11 Jun '14

Hi, is there any way to return values from a call to python_exec function orther than the default returned integer? Following code generates this error: Oct 1 09:08:23 zskio /usr/sbin/kamailio[21789]: ERROR: app_python [python_support.c:38]: python_exec2: Unhandled exception in the Python code: Oct 1 09:08:23 zskio /usr/sbin/kamailio[21789]: ERROR: app_python [python_support.c:78]: #011RuntimeError: no such function if (!python_exec("get_provider", $(var(clean_uri){s.prefixes}))) { xlog("L_ERR","Can not choose a provider for $rU\n"); sl_send_reply("503","Can not choose a provider for $rU"); exit; } def get_provider(self, msg, param): # snipped code... result = self.util.get_provider(prefixes, position, mode) if result is not None: LM_ERR("mode: %s position: %d result: %s" % (mode, position, result)) # this raises error msg.call_function('rewritehost', result['provider']['ip']) return 1 return -1 I tried to use headers as transport but this simple code does not work: Oct 1 09:45:31 zskio /usr/sbin/kamailio[22303]: ERROR: <script>: No header if (!is_present_hf("X-Test-header")) { xlog("L_ERROR", "No header\n"); } else { xlog("L_INFO", "Header added\n"); } def test_set_header(self, msg): msg.call_function('append_hf', 'X-Test-header: test\r\n') return 1 Thanks in advance.

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TLS and SIP
by Fabian Borot 23 May '14

23 May '14

Hi I am using Kamailio 4.0.1 in front of an asterisk servers farm to handle TLS with our clients and providers. The idea is to have kamailio "talking" SIP/UDP/5060 and TLS/TCP/5061 with the customers and providers and regular SIP/UDP/5060 with our internal asterisk servers. So far at least for the customers it looks like it can work. But I have a problem, when the call is established and the called person hangs up, the BYE from the called person to the calling person is ignored. Only when the calling person hangs up first the call is terminated properly. This is what I have been able to see: 1- Customer starts the TLS handshake/connection. 2- Kamailio authenticate it, then routes the call to the asterisk server using regular SIP/UDP/5060 but I see that it is inserting 2 Record Routes in the INVITE: Record-Route: <sip:192.168.1.58;r2=on;lr=on> Record-Route: <sip:192.168.1.58:5061;transport=tls;r2=on;lr=on> 3- The Contact on that INVITE to the asterisk also comes like this: Contact: <sip:94167032@172.31.196.21:53325;transport=tls> 4- The ACK sent to the asterisk once it accepts the call (200 OK) also has those 2 Record-Routes: Record-Route: <sip:192.168.1.58;r2=on;lr=on> Record-Route: <sip:192.168.1.58:5061;transport=tls;r2=on;lr=on> 5- The call is established, once the called person decides to hang up the BYE looks like this: BYE sip:94167032@172.31.196.21:53325;transport=tls SIP/2.0 Via: SIP/2.0/UDP 192.168.1.59:5060;branch=z9hG4bK40fa1c23;rport Route: <sip:192.168.1.58;r2=on;lr=on>,<sip:192.168.1.58:5061;transport=tls;r2=on;lr=on> Max-Forwards: 70 From: <sip:3030500@1.2.3.4>;tag=as37953869 To: "kamailio" <sip:kamailio@1.2.3.4>;tag=788cd7c892df40f3b1967112395e2ca4 Call-ID: f9fe65daf1074219be26cb0c224339f1 CSeq: 102 BYE User-Agent: Asterisk PBX 11.3.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 My kamailio TLS config is shown below: enable_tls=yes loadmodule "tls.so" # ----- tls params ----- modparam("tls", "config", "/usr/local/kamailio-4.1//etc/kamailio/tls.cfg") modparam("tls", "private_key", "./privkey.pem") modparam("tls", "certificate", "./kamailio1_cert.pem") modparam("tls", "ca_list", "./calist.pem") modparam("tls", "verify_certificate", 1) modparam("tls", "require_certificate", 1) The TLS client that I am using is called Blink.At this point I don't know whether kamailio is sending the BYE using TLS to the customer and waiting for the 200 OK from the customer or whether kamailio does not like something in the BYE and that is why is ignoring it. I see some encrypted packets from kamailio to the client but I don't know what is inside. Any help would be very appreciated. thank you fabian

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