January 2011 - sr-users

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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[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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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-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

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Wrong ACK packet sent from Cisco gw
by Nawfel Oujdi 07 Mar '13

07 Mar '13

Hello! I m facing the same strange behaviour with my AS5300 voice gateway. When the gw is connected directly to PBX everythings works well but when i put a sip proxy forwarding calls between gw and PBX all the calls hangs up after 5 sec (+or -). Looking into the trace sip i realize that gw send a wrong ACK in reply of INVITE , then sip proxy discard it and PBX hangs the call cause he never receive the ACK. ACK sip:79.125.120.12:5060;lr=on;did=ce.3716ea02 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 Route: <sip:911111500@PBX:5060> To: <sip:911111500@sip_proxy>;tag=as7f388e3f Date: Mon, 17 Jan 2011 09:26:36 GMT Call-ID: B6F61A2E-215211E0-802BD462-C4432B89@cisco_gw To work fine , the content of Route header should be in ACK header and viceversa. I tried to compare between the sip trace of a wrong call and a good one (using other cisco gw AS5350 who works well with sip proxy in the same escenario) and i realize that the only difference is the INVITE of wrong case doesn' t send branch number in the via header. INVITE sip:911111500@sip_proxy:5060 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 To: <sip:911111500@sip_proxy> i m using c5300-is-mz.123-26.bin ios version. Anybody understand what is happening in there?? is there any solution?? i ll send more information if it s requested. Thanks in advance. Nawfel Oujdi here is the result of ngrep: U 2011/01/13 15:14:43.791514 cisco_gw:51703 -> sip_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 6. Remote-Party-ID: <sip:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.791893 sip_server:5060 -> cisco_gw:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: OpenSIPS (1.6.3-notls (i386/linux)). Content-Length: 0. . U 2011/01/13 15:14:43.791957 sip_server:5060 -> asterisk_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 5. Remote-Party-ID: <sip:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.792775 asterisk_server:5060 -> sip_server:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Length: 0. . U 2011/01/13 15:14:43.793770 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2011/01/13 15:14:43.794688 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:43.856520 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:43.993417 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2011/01/13 15:14:43.993613 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:44.038774 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:44.193431 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. -- ------------------------------ Aviso legal: Este mensaje electrónico está dirigido únicamente a la(s) dirección(es) indicadas anteriormente; el carácter confidencial, personal e intransferible del mismo está protegido legalmente. Cualquier revelación, uso o reenvío no autorizado, completo o en parte, está prohibido. Si ha recibido este mensaje por equivocación, notifíquelo inmediatamente a la persona que lo ha enviado y borre el mensaje original junto con sus ficheros anexos sin leerlo ni grabarlo, total o parcialmente.

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[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.

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[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

Float Comparison
by Brandon Armstead 25 Aug '11

25 Aug '11

Hello, Is there anything special that needs to be done for float comparison? For example: if([5.5 >= 4.3]) .... or if(5.5 > 4.3) .... The conditional does not seem to be coming back as true like it should? This is simply an example... the precision in the actual case is different (longer decimal places). Thank you for all and any help in advance!

4 9

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

tcp hangup
by Juha Heinanen 05 Apr '11

05 Apr '11

i have two sr proxies talking with each other over tcp. one is running 3.0 and the other 3.1. sometimes i notice that no sip requests or any other tcp packets get through from 3.0 sr host to 3.1 sr host. netstat on 3.0 sr host shows that tcp connection to 3.1 sr host is in ESTABLISHED state whereas netstat on 3.1 sr shows that the same tcp connection is in FIN_WAIT1 state. it takes several minutes before the hanging tcp connection is replaced by a new working one. there is no ip level connectivity problem between the two sr hosts. any suggestions on what might be going on? is there some known bug in 3.0 tcp implementation that could explain this? -- juha

3 12

How to access additional xavp's
by Alex Hermann 07 Mar '11

07 Mar '11

Hello, I'm currently toying with xavp's and have some trouble accessing the values. I want to have access to the xavp that isn't the last added one. From the wiki page on http://sip-router.org/wiki/devel/xavp I got the impression that indices are supported, but that doesn't seem to work. In the following fragment i want access to the values 1A & 1B, how to do that? $xavp(test=>a) = "1A"; $xavp(test[0]=>b) = "1B"; $xavp(test=>a) = "2A"; $xavp(test[0]=>b) = "2B"; xlog("$xavp(test[0]=>a)"); xlog("$xavp(test[0]=>b)"); xlog("$xavp(test[1]=>a)"); xlog("$xavp(test[1]=>b)"); pv_xavp_print(); Which gives the following result: Dec 21 14:30:35 veyron wsproxy1[10398]: ERROR: <script>: 2A Dec 21 14:30:35 veyron wsproxy1[10398]: ERROR: <script>: 2B Dec 21 14:30:35 veyron wsproxy1[10398]: ERROR: <script>: <null> Dec 21 14:30:35 veyron wsproxy1[10398]: ERROR: <script>: <null> Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:470]: +++++ XAVP list: 0xb3a38770 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:473]: *** XAVP name: test Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:474]: XAVP id: 2063405720 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:475]: XAVP value type: 6 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:496]: XAVP value: <xavp:0xb3a387c4> Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:470]: +++++ XAVP list: 0xb3a387c4 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:473]: *** XAVP name: b Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:474]: XAVP id: 110 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:475]: XAVP value type: 2 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:484]: XAVP value: 2B Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:473]: *** XAVP name: a Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:474]: XAVP id: 109 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:475]: XAVP value type: 2 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:484]: XAVP value: 2A Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:505]: ----- XAVP list Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:473]: *** XAVP name: test Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:474]: XAVP id: 2063405720 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:475]: XAVP value type: 6 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:496]: XAVP value: <xavp:0xb3a386c8> Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:470]: +++++ XAVP list: 0xb3a386c8 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:473]: *** XAVP name: b Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:474]: XAVP id: 110 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:475]: XAVP value type: 2 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:484]: XAVP value: 1B Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:473]: *** XAVP name: a Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:474]: XAVP id: 109 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:475]: XAVP value type: 2 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:484]: XAVP value: 1A Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:505]: ----- XAVP list Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:505]: ----- XAVP list -- Greetings, Alex Hermann

2 4

Re: [SR-Users] [sr-dev] planning release of v3.1.2
by Bernhard Suttner 06 Mar '11

06 Mar '11

Hi, whats about: http://sip-router.org/tracker/index.php?do=details&task_id=108 I think that some users are trying to register to an asterisk system. Or does somebody have a solution for that? A new release would be nice!!! Thanks in advance. Best regards, Bernhard ----- Original Message ----- From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] To: kamailio [mailto:sr-users@lists.sip-router.org], sr-dev [mailto:sr-dev@lists.sip-router.org] Sent: Fri, 28 Jan 2011 20:15:50 +0100 Subject: [sr-dev] planning release of v3.1.2 > Hello, > > I think it is time to release v3.1.2, first date that comes in my mind > is next Thursday if everyone feels it is enough time to take care of > backporting any fix he/she did and it is not yet there. That will > provide us a fresh release for the FOSDEM event. If not, then maybe the > other week, Tuesday, so the participants at the Kamailio Devel training > in Barcelona can practice on it. > > Soon after we should plan also a release for previous stable, branch 3.0. > > Anyone having other options? > > Thanks, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > > _______________________________________________ > sr-dev mailing list > sr-dev(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev >

2 1

Merging nathelper modules
by Ovidiu Sas 10 Feb '11

10 Feb '11

Hello all, The nathelper module in modules_k was split in two: - nathelper (dealing with signaling); - rtpproxy (dealing with rtpproxy protocol). I would like to move the rtpproxy module from "modules_k" into "modules" and remove rtpproxy functionality from nathelper (s). This will give to (s) and (k) users: - rtpproxy: a single module for dealing with rtpproxy servers; - nathelper: two variants for dealing with NAT signaling. Next step, will be to merge the two nathelper modules into a single one. Thoughts? Regards, Ovidiu Sas

2 2

[OpenSER-Users] uaCSTA support in openser
by Hugo Koblmueller 09 Feb '11

09 Feb '11

high.all! i'm wondering if there is any support of uaCSTA in openser (planned)? i'm just working on the integration of asterisk (*) environment to OCS 2007 environment, having openSER in the middle (mainly for TCP/UDP translation and smoothing out the protocol deficienes on both sides). in this setup the * having the openSER in front is talking to the OCS (and vice versa) via the OCS mediation server, which is moreorless sending standard SIP messages, which enables normal softphone (integration to *) of the office communicator. this configuration is already working... now i'm planning to go for the CTI integration, where there is no OCS mediation server in between OCS and openSER, doing the translation of SIP/CSTA to SIP. i'm thinking about using openSER for this task, that's why i'm looking for a CSTA module or perl programm, which is capable of this functionality. afaik for the CTI communication there isn't the full complexity of CSTA needed, just a subset mainly for call setup and call clearing. anyone having experience on this topic? thx & cheers -hugo Great Ideas for Small Devices Hugo Koblmueller Senior Staff Engineer Software Development COMNEON electronic technology GmbH & Co. OHG Freistaedter Strasse 400 4040 Linz Austria hugo.koblmueller(a)comneon.com tel: fax: mobile: Skype ID: +43 (5) 1777 - 15730 <http://www.plaxo.com/click_to_call?lang=en&src=jj_signature&To=%2B43+%285%2 9+1777+%2D+15730&Email=hugo(a)koblmueller.com> +43 (5) 1777 - 15810 +43 (676) 82051280 <http://www.plaxo.com/click_to_call?lang=en&src=jj_signature&To=%2B43+%28676 %29+82051280&Email=hugo(a)koblmueller.com> drhookson Want to always have my latest info? <https://www.plaxo.com/add_me?u=21475050628&src=client_sig_212_1_banner_join &invite=1&lang=en> Want a signature like <http://www.plaxo.com/signature?src=client_sig_212_1_banner_sig&lang=en> this?

7 13

planning release of v3.1.2
by Daniel-Constantin Mierla 08 Feb '11

08 Feb '11

Hello, I think it is time to release v3.1.2, first date that comes in my mind is next Thursday if everyone feels it is enough time to take care of backporting any fix he/she did and it is not yet there. That will provide us a fresh release for the FOSDEM event. If not, then maybe the other week, Tuesday, so the participants at the Kamailio Devel training in Barcelona can practice on it. Soon after we should plan also a release for previous stable, branch 3.0. Anyone having other options? Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 1

503 Server error response for REGISTER request (Asterisk v1.6.0.5)
by Vikram Ragukumar 04 Feb '11

04 Feb '11

Hello, I had posted this question earlier on asterisk-users, but didn't receive any reply, so I'm posting on sr-users in the hope that someone can provide guidance on how to debug this problem. Our setup involves a Sip softphone registering with a fresh install of Asterisk v1.6.0.5 through an outbound proxy (Kamailio v3.1.1). We observe that about 80% of the time, immediately after startup of the softphone, when the softphone sends a REGISTER message, Asterisk replies with a "503 Server error". However, subsequent REGISTER messages are replied to appropriately (with a 401 Unauthorized). There are very few devices (under 5) registered with Asterisk, so this is not an overload issue. Please find attached a SIP trace for device registration. - What might cause this behavior in Asterisk ? - Are there any config file parameters that require to be changed ? - Would it help to see any other specific logs, if so which ones ? Thanks and Regards, Vikram.

2 3

SIP Router 3.03 topoh
by dotnetdub 04 Feb '11

04 Feb '11

Hi All, Kamailio had just started to get SIP attacked and it looked like they were sending a bad header and Kamailio died.. Nov 20 19:48:51 sipproxy sip[3383]: INFO: <script>: [R-MAIN =>] Request method OPTIONS is to our domain (212.124.123.170) Nov 20 19:48:51 sipproxy sip[3378]: INFO: <script>: [ROUTE-1] Received registration from sip:1368250498@sipproxy (212.124.123.170) Nov 20 19:48:51 sipproxy sip[3378]: INFO: <script>: [ROUTE-1 !] Credentials invalid; issuing challenge Nov 20 19:48:51 sipproxy sip[3381]: INFO: <script>: [ROUTE-1] Received registration from sip:100@sipproxy (212.124.123.170) Nov 20 19:48:51 sipproxy sip[3381]: INFO: <script>: [ROUTE-1 !] Credentials invalid; issuing challenge Nov 20 19:48:51 sipproxy sip[3374]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.401446] kamailio[3374]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3373]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.636774] kamailio[3373]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3388]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.680305] kamailio[3388]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3390]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.710444] kamailio[3390]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3383]: INFO: <script>: [ROUTE-1] Received registration from sip:2478223614@sipproxy (212.124.123.170) Nov 20 19:48:51 sipproxy sip[3383]: INFO: <script>: [ROUTE-1 !] Credentials invalid; issuing challenge Nov 20 19:48:51 sipproxy sip[3378]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.753334] kamailio[3378]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3379]: INFO: <script>: [ROUTE-1] Received registration from sip:noauth@sipproxy (212.124.123.170) Nov 20 19:48:51 sipproxy sip[3379]: INFO: <script>: [ROUTE-1 !] Credentials invalid; issuing challenge Nov 20 19:48:51 sipproxy sip[3375]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.768137] kamailio[3375]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3383]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.775628] kamailio[3383]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3386]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.790925] kamailio[3386]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3379]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.798926] kamailio[3379]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3381]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.815493] kamailio[3381]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:54 sipproxy sip[3344]: INFO: <core> [main.c:737]: INFO: terminating due to SIGCHLD Nov 20 19:48:54 sipproxy sip[3399]: INFO: <core> [main.c:788]: INFO: signal 15 received Nov 20 19:48:54 sipproxy sip[3396]: INFO: <core> [main.c:788]: INFO: signal 15 received Nov 20 19:48:54 sipproxy sip[3400]: INFO: <core> [main.c:788]: INFO: signal 15 received Nov 20 19:48:54 sipproxy sip[3392]: INFO: <core> [main.c:788]: INFO: signal 15 received Nov 20 19:48:54 sipproxy sip[3401]: INFO: <core> [main.c:788]: INFO: signal 15 received Nov 20 19:48:54 sipproxy sip[3394]: INFO: <core> [main.c:788]: INFO: signal 15 received Nov 20 19:48:54 sipproxy sip[3402]: INFO: <core> [main.c:788]: INFO: signal 15 received Thanks Brian

4 20

Redundancy between 2 Kamailio servers
by Danny Dias 02 Feb '11

02 Feb '11

Hello my friends, I wonder if someone has done this before? i would like to implement 2 Kamailio servers with redundancy, something like the following: SERVER_A is working as the primary sip proxy (virtual IP as the sip signaling), if it fails, the other server (sleeping) should UP the virtual IP and takes all the traffic....i guess there is many Linux implementations thah can do this, but i would like to know if there is someone that has done this before and hear some recomendations... I've read some difficulty in the synchronisation of registrations because Kamailio works best when it stores registrations in memory and registrations are constantly changing - they expire and are renewed, as well as new ones joining and old ones leaving. To make the failover solution function seamlessly, it is necessary to synchronise the in-memory registrations between the primary and the backup server . This can be done by forking a copy of the registration request to the backup server, but there are some practical problems in doing this, has anyone do something with this? Thanks in advance!

6 21

ACC error | failed to insert into database
by alex pappas 02 Feb '11

02 Feb '11

Hi all, Hve anyone seen before the following error? *Kamailio acc [acc.c:398]: failed to insert into database* After a Kamailio restart it is ok but it start again afetr x time. Thank you Alex

2 2

Trunk connection asterisk-kamailio
by Mino Haluz 01 Feb '11

01 Feb '11

Hi, I have a question about kamailio-asterik interconnection. I'd like to connect 1000 numbers with a trunk, but it would be painful to add 1000 trunks on asterisk. Do you have some idea how could I group those numbers into one trunk connected to kamailio? asterisk and kamailio configuration settings. Thank you very much.

2 1

kamailio and sip-router at FOSDEM 2011 - social event
by Henning Westerholt 31 Jan '11

31 Jan '11

Hello, in about two and a half week the annual open source developer conference FOSDEM (http://fosdem.org/2011/) takes place in Brussels. As in the last two years we would like to meet here for a social event, probably a dinner on Saturday evening, 6th February. This developers already confirmed their participance: - Daniel-Constantin Mierla - Marius Zbihlei - Henning Westerholt This year we'll also be present in the Open Source Telephony room on sunday with two talks, "Scaling location services in large SIP networks with Kamailio from Marius and myself, and "Unifying SIP and Web worlds with Lua (Kamailio)" from Daniel. Olivier Taylor agreed helping us again in choosing a nice place for the evening. Participation at the event is free of charge, everybody pays for its own expenses. In order to organize a restaurant and do a reservation please send me a short mail if you would like to attend. Looking forward to see you in Brussels! Best regards, Henning P.S.: Photos of the 2009 event can be found on the web site: http://www.kamailio.org/events/2009-FOSDEM/FOSDEM2009-Album/ -- Henning Westerholt - Head of IT Operations Internet Access & Communications 1&1 Internet AG, Karlsruhe, Germany

2 3

xlogl and pseudo variables
by Bernhard Suttner 29 Jan '11

29 Jan '11

Hi, I am using xlogl for debugging and logging the most important actions within the kamailio script. Does it take much resources if I use the pseudo variables with the xlogl statements? Example: xlogl("L_NOTICE", "Route $rm to destination! F=$fu T=$tu IP=$si \n"); I´m planning to do this in all the xlogl statements to be able to debug / "see whats going on" on the system. Thanks in advance. Best regard, Bernhard

2 1

Re: [SR-Users] onreply_route and failure_route
by Bernhard Suttner 29 Jan '11

29 Jan '11

Hi, that was _really_ fast. Thanks. :-) One more question: Could I somehow test with the route[NAT] for example, if the request was from the main route { ... } or from the onreply_route[] ? Best regards, Bernhard ----- Original Message ----- From: Alex Balashov [mailto:abalashov@evaristesys.com] To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] Cc: sr-users(a)lists.sip-router.org [mailto:sr-users@lists.sip-router.org] Sent: Sat, 29 Jan 2011 21:37:34 +0100 Subject: Re: [SR-Users] onreply_route and failure_route > Yes. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > On Jan 29, 2011, at 3:35 PM, "Bernhard Suttner" <bernhard.suttner(a)winet.ch > > wrote: > > > Hi, > > > > is it possible to call a route within onreply_ and failure_route? I > > mean, within the main route { ... } its possible to call the route > > [NAT]. Is this also possible for this "special" routes? If this is > > possible, is it also allowed to call a route which will normally > > called from the main route { ... } > > > > Thanks in advance. > > > > Best regards, > > Bernhard > > > > _______________________________________________ > > 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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sr-users January 2011