sr-users December 2010

sr-users@lists.kamailio.org

63 participants
115 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

[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

11 24

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

2 1

[Kamailio-Users] Error in Running kamailio !!
by Dinesh gautam 25 Jul '13

25 Jul '13

Dear Sir, I am using kamailio 1.5.0, having installed and configured as given on the kamailio site <http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn> on my PC having Centos operating system. I have configured the config files kamailio.cfg, kamctlrc accordingly as given on site. when I start the kamailio using kamctl start command. it gives the error message like *[root@localhost ~]# kamctl start database engine 'MYSQL' loaded Control engine 'FIFO' loaded INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed *please* *help me out to solve this problem and suggest me where I am wrong. * *I have configured kamalio so many times it gives me the same kind of error message. * * - With Regards, Dinesh Kumar Gautam Noida, +919911838275

6 15

[Kamailio-Users] BYE message not relayed to caller
by Vikram Ragukumar 04 Dec '12

04 Dec '12

Hello, My setup includes a cellphone that uses Wi-Fi to hookup to the internet with IP 192.168.1.101. Cellphone app, tries to register with VOIPSwitch through an outbound proxy server(Kamailio + rtpproxy), which rewrites SDP to force rtp through an rtpproxy. I can make a call from the cellphone to a PSTN number. When the PSTN endpoint hangs up, VOIPSwitch relays a BYE message to the proxy that looks like : No. Time Source Destination Protocol Info 27 18.383929 IPaddr_VoipSwitch IPaddr_proxy SIP Request: BYE sip:1234@IPaddr_VoipSwitch Frame 27 (420 bytes on wire, 420 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: IPaddr_VoipSwitch , Dst: IPaddr_proxy User Datagram Protocol, Src Port: sip (5060), Dst Port: 7160 (7160) Session Initiation Protocol Request-Line: BYE sip:1234@IPaddr_VoipSwitch SIP/2.0 Method: BYE [Resent Packet: False] Message Header Route: <sip:IPaddr_proxy:7160;lr=on;nat=yes> CSeq: 1 BYE Via: SIP/2.0/UDP IPaddr_VoipSwitch:5060; branch=z9hG4bk180242100434182912333954 From: sip:5678@IPaddr_VoipSwitch;tag=18024210041929123187506289 Call-ID: LUOyIDm-ecy36NcUfj5G3v1lM6g5snkM To: "1234" <sip:1234@IPaddr_VoipSwitch>;tag=9XHJmiuf58vTrF7.vwflEK-G63JVPQSi Content-Length: 0 The request URI in the above message is the IP address of VOIPSwitch itself, which causes the proxy to forward the BYE message back to VOIPSwitch. The cellphone never receives the BYE message and consequently never hangs up. Any suggestions on how i can get around this problem ? Thanks and Regards, Vikram.

3 9

[Kamailio-Users] AudioCodes + Kamailio : Problem in SIP Message Headers
by Samuel Muller 26 Sep '12

26 Sep '12

Hello all, I recently add a classical Audiocodes Mediant 2000 with 2x 8E1, the purpose is to have several interconnections with PSTN. I configured it like this : Audiocodes registers as a gateway to the Kamailio, using a dedicated port (5062). Registration seems to be OK, and the pstn gw uses OPTIONS method to ping the proxy. I can attack the Audiocodes with a SIP phone behind Kamailio, no pbm. But the audiocodes returns some errors about SIP headers sent by Kamailio : ( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time: 12:30:26] ( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol '0' in scheme. ALPHA expected Here you have the example of an INVITE from a SIP phone to the PSTN : ** audiocodes debug ** 4d:12h:30m:26s ( lgr_flow)(44730 ) ---- Incoming SIP Message from 77.246.81.132:5060 ---- INVITE sip:0323719001@77.246.81.136:5062;transport=udp SIP/2.0 Record-Route: <sip:77.246.81.132;lr=on;ftag=71078b346a20fb3eo0;nat=yes> Via: SIP/2.0/UDP 77.246.81.132;branch=z9hG4bKdace.1ab1d59.0 Via: SIP/2.0/UDP 192.168.0.113:5060;rport=15170;received=77.246.81.162;branch=z9hG4bK-b432f96 From: "Sam" <sip:0123451010@sip.720.fr <sip%3A0123451010(a)sip.720.fr>>;tag=71078b346a20fb3eo0 To: <sip:0323719001@sip.720.fr <sip%3A0323719001(a)sip.720.fr>> Call-ID: 944d8aec-27503ee6(a)192.168.0.113 CSeq: 102 INVITE Max-Forwards: 49 Contact: "Sam" <sip:0123451010@77.246.81.162:15170> Expires: 240 User-Agent: Linksys/SPA941-5.1.8 Content-Length: 281 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER Supported: 100rel, replaces Content-Type: application/sdp P-Asserted-Identity: <0123451010> Remote-Party-ID: <0123451010>;party=caller;privacy=none;screen=yes v=0 o=- 26933860 26933860 IN IP4 192.168.0.113 s=- c=IN IP4 77.246.81.133 t=0 0 m=audio 35038 RTP/AVP 18 0 8 101 a=rtpmap:18 G729a/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=nortpproxy:yes ( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time: 12:30:26] ( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol '0' in scheme. ALPHA expected ( sip_stack)(44734 ) !! [ERROR] Message type: INVITE [Time: 12:30:26] ( sip_stack)(44735 ) !! [ERROR] Source header: [Time: 12:30:26] ( sip_stack)(44736 ) !! [ERROR] Line: 17. Column: 23 [Time: 12:30:26] The outgoing INVITE from Kamailio is exactly the same received by the AudioCodes. When I searched over Google, I just found 2 answers about Asterisk / Audiocodes unsolved problem, but no more informations. I supposed that the problem is as indicated : " s=- " where source is empty in place of "NULL" / "0" or something like this ... Someone can confirm or already met the problem ? Many thanks all :) .Sam.

5 5

[Kamailio-Users] Having problems using RTPProxy to bridge internal/external networks
by Joe Hart 15 Sep '12

15 Sep '12

Hi all, For a project on which I'm currently working, I am having some problems figuring out how to correctly configure Kamailio to communicate with RTP Proxy in order to send media into and out of a network with private IP address ranges. I have a proxy set up to send the SIP traffic, and all of this is working fine. However, I'm having some trouble getting the RTP Proxy set up. Currently, when the call is connected, the offer/answer is made and RTP Proxy seems to be taking over, but I'm having trouble getting my audio to flow in both directions. Examination of the traffic coming into and out of this machine seems to indicate that the IP addresses aren't being mangled correctly. Specifically, it appears the internal IP address isn't being changed to reflect the IP address of the machine on which RTP Proxy is running, so that when the caller tries to send audio back, the IP it's given to reply to is 10.10.x.x, which obviously won't work. I have tried experimenting with specifically setting IP addresses in the rtpproxy_offer() and _answer() methods to no avail, as well as setting various flags in those methods. However, I must admit that I'm not entirely sure what's happening under the hood with these methods, or what rtpproxy is doing with that information when it gets it. Rather than continue to hack at this by trial and error, I'm hoping someone here can point me in the right direction. Any advice, example code or pep talks would be greatly appreciated. Thanks in advance, -- Joe Hart Voice Systems Integrator Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0673

8 37

[Kamailio-Users] Converting rtpproxy recorded .rtp file to .wav file
by Vikram Ragukumar 18 Nov '11

18 Nov '11

Hello, I used Kamailio+rtpproxy to record a session and rtpproxy outputs the following files long_file_name.a.rtp, long_file_name.a.rtcp, long_file_name.o.rtp, long_file_name.o.rtcp http://www.rtpproxy.org/wiki/RTPproxy/FAQ From the Rtpproxy FAQ above, i tried to extract the audio using rtpbreak and sox. rtpbreak -W -r long_file_name.a.rtp rtpbreak -W -r long_file_name.o.rtp The above commands generate rtp.0.0.raw, rtp.1.0.raw. Then when i run sox using sox --combine merge -r 8k -A rtp.0.0.raw -r 8k -A rtp.1.0.raw -t wavpcm -s out.wav i get the following errors : sox: invalid option -- - sox: -c must be given a number Is there a switch/anything else that i am missing ? Thanks in advance, Regards, Vikram.

3 5

Bug in LDAP module?
by Pan B. Christensen 26 Apr '11

26 Apr '11

Hello, We are using kamailio 1.5 and we have the following problem with ldap_search: A call comes in. If it is to a VOIP user, it will be routed to kamailio, and eventually go to the ldap lookup route: route[10] to look up the user. This is always successful (LDAP can connect). It will then look up user preferences (call forwarding etc.) from SQL in route[12] and route based on this result in route[13]. If the result is "CFU (call forward unconditional) = on", the URI is changed and route[10] (LDAP lookup) is called again. This is always successful (LDAP can connect). If the result is "CFB (call forward busy) = on" and the user is busy, the URI is changed and route[10] (LDAP lookup) is called again. This is also always successful (LDAP can connect). If the result is "CFNR (call forward no reply) = on", the timeout is changed and the call is delivered to the phone through route[14] (location) and route[15] (relay). If the user does not answer, the URI is changed in failure_route[1] and route[10] is called. For some unknown reason, ldap_search always fails in this scenario with the messages: "ldap_session not found", "couldn't get ldap session" and then "no LDAP entry found". Why? And why does this result in the -1 return code and not the -2 return code for ldap_search? The only differences I can think of is that the unsuccessful scenario comes from FAILURE_ROUTE instead of REQUEST_ROUTE, and that longer time has passed since the original invite. Here's a sample log: NOTE the three ERROR lines, the INFO line and the ROUTE 10 line. Dec 27 13:31:55 [3373]: INVITE received from sip:<CALLER>@<SBC_IP> <source IP: <SBC_IP>> to sip:<CALLEE>@<KAM_IP>: Dec 27 13:31:55 [3373]: ERROR:ldap:ldap_reconnect: [ldapprod]: LDAP reconnect successful Dec 27 13:31:55 [3373]: ROUTE 10: LDAP returned: UID = <UID>, voipSipPrimaryUri = <PRIMARY_URI>, voipExtensionUri = sip:<CALLEE>@<DOMAIN> Dec 27 13:31:55 [3373]: ROUTE 12: Effective user properties for UID <UID>: CFU: off <FORWARDTO>, CFB: off <FORWARDTO>, CFNR: on:25 <FORWARDTO>, CFNC: off <FORWARDTO>, VM: <CALLEE>, busy_level: 1, DBrows: 1 Dec 27 13:31:55 [3373]: -------------------- Calling sip:<CALLEE>@<DOMAIN> for 25 seconds. -------------------- Dec 27 13:31:55 [3373]: ROUTE 15: Sending 105 INVITE from sip:<CALLER>@<SBC_IP> to sip:<CALLEE>@<CALLEE_IP> (via <<null>>) Dec 27 13:31:55 [3383]: Response 100 Trying to Cseq 102 INVITE received from <PROTO>:<CALLEE_IP>:<PORT> Dec 27 13:31:55 [3385]: Response 180 Ringing to Cseq 102 INVITE received from <PROTO>:<CALLEE_IP>:<PORT> Dec 27 13:32:20 [3388]: FAILURE_ROUTE 1: Timeout for: 104 INVITE from <CALLER> -> <CALLEE> Dec 27 13:32:20 [3388]: -------------------- Forwarding on no reply to <FORWARDTO> -------------------- Dec 27 13:32:20 [3388]: ERROR:ldap:get_connected_ldap_session: [ldapprod]: ldap_session not found Dec 27 13:32:20 [3388]: ERROR:ldap:lds_search: [ldapprod]: couldn't get ldap session Dec 27 13:32:20 [3388]: INFO:ldap:ldap_search_impl: no LDAP entry found Dec 27 13:32:20 [3388]: ROUTE 11: Number lookup results for <FORWARDTO>: Node: VOIP. Dec 27 13:32:20 [3388]: CONFIG ERROR: VOIP number not found in LDAP. Hanging up. Dec 27 13:32:20 [3373]: Response 487 Request Cancelled to Cseq 102 INVITE received from <PROTO>:<CALLEE_IP>:<PORT> Here's the ldap config file: [ldapprod] ldap_server_url = "ldap://ldap.<DOMAIN>" ldap_network_timeout = 500 ldap_client_bind_timeout = 500 kamailio config snippets (in case you need them): # LDAP lookup for UID and primary URI route[10] { ... ldap_search("ldap://ldapprod/<DN>?uid,voipSipPrimaryUri,voipExtensionUri?one?(&(objectClass=voipAddress)(|(voipSipUri=sip:$rU@uio.no)(|(voipSipUri=$var(uri))(voipSipUri=$var(e164uri)))))"); if ($retcode < 1) { switch ($retcode) { case -1: # no LDAP entry found route(11); return(1); case -2: # internal error sl_send_reply("500", "Internal server error"); exit; } } ldap_result("uid/$avp(s:uid)"); $avp(s:exten) = null; if (!ldap_result("voipExtensionUri/$avp(s:exten)")) { xlog("L_INFO", "voipExtensionUri not found.\n"); } if (!ldap_result("voipSipPrimaryUri/$avp(s:p-uri)")) { xlog("L_WARN", "voipSipPrimaryUri not found.\n"); #exit here? } if (isflagset(1)) { xlog("L_INFO", "ROUTE 10: LDAP returned: UID = $avp(s:uid), voipSipPrimaryUri = $avp(s:p-uri), voipExtensionUri = $avp(s:exten)\n"); } if $avp(s:exten) != null { $ru=$avp(s:exten); } else { $ru=$avp(s:p-uri); } route(12); } } # Number lookup. Non-VoIP URI's go here. route[11] { ... } # Database lookup for call forwarding. route[12] { ... } # Call forwarding logic route[13] { if (is_method("INVITE")) { $avp(s:uio_timer_avp) = "300"; if ($avp(s:cfu_status) == "on") { # Forward immediately xlog("L_INFO", "-------------------- $rU forwarded unconditionally to $avp(s:cfu_number). --------------------\n"); $rU = $avp(s:cfu_number); route(10); } ... if ( $var(dlg_busy) >= $avp(s:busy_level) ) { if ($avp(s:cfb_status) == "on") { xlog("L_INFO", "-------------------- $rU has $var(dlg_busy) active calls. Treshold $avp(s:busy_level). Forwarding on busy to $avp(s:cfb_number) --------------------\n"); $rU = $avp(s:cfb_number); route(10); } ... else if ($avp(s:cfnr_status) == "on") { # Forward to phone with timeout then to $avp(s:cfnr_number). # t_set_fr($avp(s:cfnr_timeout) + "000"); # Recommended method in 3.1. Function does not exist in 1.5. Using old method (avp). $avp(s:uio_timer_avp) = $avp(s:cfnr_timeout); xlog("L_INFO", "-------------------- Calling $ru for $avp(s:uio_timer_avp) seconds. --------------------\n"); route(14); } ... } failure_route[1] { ... if $avp(s:cfnr_status) == "on" { if (isflagset(1)) { xlog("L_INFO", "FAILURE_ROUTE 1: Timeout for: $mi $rm from $fU -> $rU\n"); } revert_uri(); $rU = $avp(s:cfnr_number); xlog("L_INFO", "-------------------- Forwarding on no reply to $rU --------------------\n"); route(10); } ... } With kind regards, Pan

3 5

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users December 2010