sr-users January 2012

sr-users@lists.kamailio.org

72 participants
114 discussions

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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File kamailio_fifo is removed...
by Bruno Bresciani 12 Mar '13

12 Mar '13

Hi, Kamailio is running fine, but when I try run again by command line without killing the previous, the kamailio_fifo is removed and functions fifo in kamctl doen't work anymore. Why when I try running kamailio when already exist one, it remove de file kamailio_fifo? I can't find the part of source code that file kamailio_fifo is removed... Someone knows? Best Regards

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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 1.5 and Asterisk 1.8 issue (Protocol Error, Unspecified)
by Rowie 24 Feb '13

24 Feb '13

Hi, We are having an issue where a phone (snom in particular) cannot make a call through Asterisk. It just hangup and does not allow the call to go through. I am including a a sip trace on this thread to show what is happening within the call. Please see below: Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:41:476 (490 bytes): SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-wowp1kmdy4rl;rport=5060 From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f Call-ID: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 -------------------------------------------------------------------------------- Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:41:481 (387 bytes): ACK sip:vm9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wowp1kmdy4rl;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f i: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 -------------------------------------------------------------------------------- Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:130 (868 bytes): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-5evtiw6dm0po;rport=5060 Record-Route: <sip:10.1.10.80;lr=on> From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:9513261429@10.1.10.83:5060> Content-Type: application/sdp Content-Length: 256 v=0 o=root 1355451627 1355451627 IN IP4 10.1.10.83 s=Asterisk PBX 1.8.7.1 c=IN IP4 10.1.10.83 t=0 0 m=audio 16094 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv -------------------------------------------------------------------------------- Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:42:132 (385 bytes): ACK sip:9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wszafb7cbzpw;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 i: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 -------------------------------------------------------------------------------- Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:232 (503 bytes): BYE sip:91421@10.30.0.64:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.10.80;branch=z9hG4bKe723.bf70c1f4.0 Via: SIP/2.0/UDP 10.1.10.83:5060;branch=z9hG4bK69f53cf1 Max-Forwards: 69 From: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 To: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.7.1 X-Asterisk-HangupCause: Protocol error, unspecified X-Asterisk-HangupCauseCode: 111 Content-Length: 0 -- View this message in context: http://old.nabble.com/Kamailio-1.5-and-Asterisk-1.8-issue-%28Protocol-Error… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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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

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