sr-users November 2009

sr-users@lists.kamailio.org

80 participants
108 discussions

by Siddhardha Garige

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

9 years, 4 months

Re: [Kamailio-Users] /usr/local/sbin/kamdbctl create error

by vivi

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

10 years

[Kamailio-Users] AudioCodes + Kamailio : Problem in SIP Message Headers

by Samuel Muller

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.

12 years, 1 month

[Kamailio-Users] Having problems using RTPProxy to bridge internal/external networks

by Joe Hart

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

12 years, 2 months

[OpenSER-Users] uaCSTA support in openser

by Hugo Koblmueller

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?

13 years, 9 months

[Kamailio-Users] kamailio and PRACK

by BERGANZ François

Hello, Does kamailio support PRACK method ? If yes, how? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

14 years, 3 months

[Serusers] Error while running icscf.sh

by Abdul Khadar

Hi, Can you please help me in resolving this issue while setting OpenIMSCore setup. Error : [root@hss OpenIMSCore]# sh icscf.sh * 0(18935) ERROR: load_module: could not open module </opt/OpenIMSCore/ser_ims/modules/mysql/mysql.so>: /opt/OpenIMSCore/ser_ims/modules/mysql/mysql.so: undefined symbol: log * 0(18935) parse error (30,13-14): failed to load module ERROR: bad config file (1 errors) 0(18935) INFO:cdp:cdp_exit(): CDiameterPeer stoping ... icscf.sh: line 5: 18935 Segmentation fault /opt/OpenIMSCore/ser_ims/ser -f /opt/OpenIMSCore/icscf.cfg -D -D please help me to resolve this problem. -- thanks & regards, Abdul Khadar

14 years, 3 months

[Serusers] ReTransmission of CANCELs on 0.9.7

by rupert.organ＠bt.com

Hi All, Has anyone come up against an issue on 0.9.6 / 0.9.7 whereby a SIP App Server, delivers a CANCEL to the SER. The SER proxies the CANCELs to all the forked endpoints. At the same time it sends "200 Cancelling" back to the SIP App server...(this suppresses CANCEL retransmission from the SIP App server). If one of the CANCELs from the SER to the SIP UA gets lost....the phone keeps ringing. There is no CANCEL retransmission on the SER !!! There is CANCEL retransmission on the SIP App, but when it sends SER replies "no pending branches" Ideally want the SER do CANCEL retransmissions. I understand the timers were rewritten in 0.10.x but this is not hardened version like 0.9.6....any suggestions? Thanks in advance Rupert

14 years, 7 months

[Kamailio-Users] Carrierroute issue

by michel freiha

Hi all, I'm using carrierroute module and ie seems that it's not working well...Suppose I have 2 entries in carrierroute table...Entry 1 with scan prefix 00 and prob 0 and a second entry with scan prefix 00 and prob=1...As soon as i make a call, this call will be forwarded to the entry with prob 0 because i guess it appears before the one with prob 1 in the table... When starting Kamailio I got the below: Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix 1, prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: domain 0 not found, add it Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_tree: tree default has 1 trees Jul 29 12:30:11 [23143] DBG:carrierroute:add_route_tree: tree 0Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: created route tree: 0, with id 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix , prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix 011, prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix 00, prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix 111, prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix 00, prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix , prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix 011, prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix 011, prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix , prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules It seems that kamailio is not reading probability entry in carrier route table Please find below carrierroute table structure: CREATE TABLE IF NOT EXISTS `carrierroute` ( `id` int(10) unsigned NOT NULL auto_increment, `carrier` int(10) unsigned NOT NULL default '0', `domain` varchar(64) NOT NULL default '', `scan_prefix` varchar(64) NOT NULL default '', `flags` int(11) unsigned NOT NULL default '0', `mask` int(11) unsigned NOT NULL default '0', `prob` float NOT NULL default '0', `strip` int(11) unsigned NOT NULL default '0', `rewrite_host` varchar(128) NOT NULL default '', `rewrite_prefix` varchar(64) NOT NULL default '', `rewrite_suffix` varchar(64) NOT NULL default '', `description` varchar(255) default NULL, PRIMARY KEY (`id`) ) ENGINE=MyISAM DEFAULT CHARSET=latin1 AUTO_INCREMENT=23 ; Can someone help me to find this issue? Regards

14 years, 9 months

[Kamailio-Users] tcp_blocking_connect related errors and Kamailio using 100% CPU

by Pascal Maugeri

Hi Using Kamailio 1.5 we periodically get the following errors: Jul 1 09:54:41 localhost /usr/local/sbin/openser[3836]: ERROR:core:tcp_blocking_connect: poll error: flags 18 Jul 1 09:54:41 localhost /usr/local/sbin/openser[3836]: ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (110) Connection timed out Jul 1 09:54:41 localhost /usr/local/sbin/openser[3836]: ERROR:core:tcpconn_connect: tcp_blocking_connect failed Jul 1 09:54:41 localhost /usr/local/sbin/openser[3836]: ERROR:core:tcp_send: connect failed Jul 1 09:54:41 localhost /usr/local/sbin/openser[3836]: ERROR:tm:msg_send: tcp_send failed They are several emails in this mailing list explaining we should play with tcp parameters. I will try to change my configuration and see if we can solve this issue. But when this kind of error happen we observe that the kamailio processes are using 100% of CPU (top). Our client is using SIP over TCP only (no UDP). Do you have any hint on how to configure Kamailio to improve TCP support ? Is it normal to kamailio processes to overload CPU ? Regards, Pascal

14 years, 9 months

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sr-users November 2009