sr-users April 2014

sr-users@lists.kamailio.org

100 participants
171 discussions

[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

2 1

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.

3 3

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

8 18

Using the auth_ephemeral module
by Hemanshu Vadehra 17 Jan '17

17 Jan '17

I'm involved in setting up a Kamailio instance and was hoping to make use of the auth_ephemeral module for authentication. But the module documentation doesn't quite make clear how exactly the module is to be employed or the web service set up. Does anyone have a working example? Regards, Hemanshu Vadehra hemanshu.v(a)directi.com

5 8

Record-Route header
by jignesh gandhi 01 Apr '16

01 Apr '16

Hello All, I have a need to save the Record-Route header from the INVITE and use it as a route header in requests like BYE. PROXY --> Kamailio(as PROXY) --> Asterisk I am using rr module to add a record_route when passing the INVITE to Asterisk. Thanks, Jignesh Gandhi

2 4

Kamailio TLS
by Kethzer Docteur 12 Feb '16

12 Feb '16

Im trying to enable Kamailio TLS but the configuration is not running at all when I cahnge it liike this as follow. Kamailio 3.3 ubuntu 12.04 LTS #!define WITH_TLS #!ifdef WITH_TLS enable_tls=yes #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif Please help -- Kethzer Docteur

4 3

Append to reply in default action of failure_route
by Efelin Novak 28 Dec '15

28 Dec '15

Hi, I would like to append a header to a 'winning' negative reply in failure_route and let the Kamailio do the default action (state fully forward the winning reply). When I use append_to_reply("Foo: bar\r\n"); and then call exit; in failure_route nothing is appended. When I use same append_to_reply then t_relay("505","Error"); and exit; the header is appended. When I use append and t_reply with dialog modul turned on I got a bug I'm solving here '[SR-Users] t_reply in failure route with dialog module'. So my question is how to put a header into a reply when I don't want to alter its code or text? I'm using Kamilio 4.0.4 on Debian 7.1 Thanks for an answer Efelin

3 3

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

4 3

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

3 6

Kamailio IMS - VM
by Bruno Cruz 17 Nov '15

17 Nov '15

Hello everyone, My name is Bruno, I'm a student from Portugal studying IMS at the moment. I've been following tutorials online, trying to get Kamailio to work as an IMS core - I'm pretty much a newbie in this field. Unfortunantly I've been getting several errors and I can't seem to properly configure everything. I was wondering if there is a All-in-One VM that can be downloaded, with Kamailio ready to be used as an IMS core? Thanks in advance. Best regards, Bruno C.

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sr-users April 2014