sr-users March 2016

sr-users@lists.kamailio.org

88 participants
128 discussions

acc_cdrs empty
by Nelson Migliaro 31 Mar '16

31 Mar '16

Hello, I am not able to set up cdr. I see acc table is updated with 2 new entries after each call (INVITE and BYE) but acc_cdrs is empty. There are errors in the log. I think that I set up everything but is not working, - load the module, set module params and set the flag? Am I missing something, should I use a store procedure or something? ------------------------------------------------------------------------------------------------------------ loadmodule "acc.so" modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 1) modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdrs_table", "acc_cdrs") modparam("acc", "cdr_start_id", "start_time") modparam("acc", "cdr_end_id", "end_time") modparam("acc", "cdr_duration_id", "duration") if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } route[WITHINDLG] { if (has_totag()) { if (loose_route()) { route(DLGURI); if (is_method("BYE")) { CREATE TABLE `acc_cdrs` ( `id` int(10) unsigned NOT NULL AUTO_INCREMENT, `start_time` varchar(32) NOT NULL DEFAULT '', `end_time` varchar(32) NOT NULL DEFAULT '', `duration` varchar(32) NOT NULL DEFAULT '', PRIMARY KEY (`id`), KEY `start_time_idx` (`start_time`) ) ENGINE=MyISAM AUTO_INCREMENT=2 DEFAULT CHARSET=latin1$$ Thank you.

1 1

UAC Register Asterisk
by Alberto Sagredo 31 Mar '16

31 Mar '16

Hi Im trying to modify example for Asterisk/Kamailio integration that miconda did, and trying to handle Forwarded register to Asterisk in an authenticated way instead removing secret in asterisk. I have made this changes route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $var(astip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $var(astport); $uac_req(furi)="sip:" + $(au{s.substr,8,0}) + "@" + $var(rip); $uac_req(turi)="sip:" + $(au{s.substr,8,0}) + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; xlog("Dentro de REGFWD"); t_on_failure("REG_AST"); uac_req_send(); } failure_route[REG_AST] { if (t_is_canceled()) { exit; } xlog("Dentro de REG AST"); if(t_check_status("401|407")) { xlog("Dentro de 401"); $avp(auser) = "test"; $avp(apass) = "test"; uac_auth(); t_relay(); exit; } } What i see on traces: U x.x.x..240:5060 -> x.x.x..56:5060 REGISTER sip:x.x.x.56:5060 SIP/2.0. Via: SIP/2.0/UDP x.x.x.240;branch=z9hG4bKfb46.a47535d4000000000000000000000000.0. To: <sip:200@x.x.x.56>. From: <sip:200@x.x.x.56>;tag=533cb9e91f4b999cf76861cbb9ed54ed-6a4f. CSeq: 10 REGISTER. Call-ID: 09dc120d65932ed7-16044(a)127.0.0.1. Max-Forwards: 70. Content-Length: 0. User-Agent: ANY. Contact: <sip:emp_001_200@x.x.x.240:5060>. Expires: 3600. . # U x.x.x.56:5060 -> x.x.x..240:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP x.x.x..240 ;branch=z9hG4bKfb46.a47535d4000000000000000000000000.0;received=x.x.x..240 ;rport=5060. From: <sip:200@x.x.x.56>;tag=533cb9e91f4b999cf76861cbb9ed54ed-6a4f. To: <sip:200@x.x.x.56>;tag=as62b47138. Call-ID: 09dc120d65932ed7-16044(a)127.0.0.1. CSeq: 10 REGISTER. Server: FPBX-2.11.0(11.17.1). Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="59e39692". Content-Length: 0. . But Kamailio is not sending authentication . Any idea what could be wrong? Or any tip to do what i try. The idea is that a sip endpoint register with authentication on kamailio and it sends register with authentication to asterisk. Thanks in advance

4 7

Kamailio v4.4.0 Released - new major version is out
by Daniel-Constantin Mierla 30 Mar '16

30 Mar '16

Kamailio v4.4.0 is out – it comes with 9 new modules and a consistent set of improvements touching more than 60 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v4-4-0-release-notes/ Many thanks to all developers and community members that made possible this release. Enjoy Kamailio v4.4.0! The development will start to focus on 5.0.0 - join the discussion on sr-users mailing list to design and decide what should Kamailio 5.0.0 look like - some notes are already collected at: * https://www.kamailio.org/wiki/devel/kamailio-5.0-design Thank you for flying Kamailio! Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 0

Releasing Kamailio v4.4.0
by Daniel-Constantin Mierla 30 Mar '16

30 Mar '16

Hello, with no major issues reported lately, I think we are on a good track for releasing stable v4.4.0. Next week a lot of people prepare for Easter, therefore I propose to do the release on Wednesday, March 30, 2016. If new things pop up, the exact date can be adjusted a bit before or after the current proposal. Any new discovered issue should be reported to bug trucker: * https://github.com/kamailio/kamailio/issues Help to complete the upgrade guide would be very appreciated: * https://www.kamailio.org/wiki/install/upgrade/stable-to-devel A draft list of what is new in 4.4 is already available at: * https://www.kamailio.org/wiki/features/new-in-4.4.x Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 2

Kamailio local and hosted
by Sébastien Brice 29 Mar '16

29 Mar '16

Hi everyone At my workplace there's plenty of hardphone directly connected to an hosted sbc in the cloud. As an added value i proposed to add locally Kamailio as a proxy server to 'federate' all the sip traffic. ipphone behind nat (A) <---> sip proxy local 192.168.x.x (B)<----> sbc registrar (mydomain.com) and sip proxy with public IP (C) when hardphone try to register, B rewrite (with rewritehost) the ip public of (C). B then uses exported function of TM to t_relay_to(C). to route INVITE from A to C i use proxy_authenticate("mydomain.com") function (on B) from auth_db to pass the calls inbetween. This works well, but i am wondering if that makes sense if i get several phones behind B. Should stateless forward works better in this kind of scenario? In a near future i plan to offer disaster recovery, and i want all users binding to C (users(a)mydomain.com)instead bind on B (in case link B---C get down)and assume some kind of telephone functionnality Is that possible ? Do you have some sort of example Kamailio do it good On-premise/localy ? How to deal with nat since all hardphones have rfc1918 address? , thx you PS: as an sbc (C) i did setup kamailio/asterisk in a real-time configuration. Sébastien BRICE VoIP, Support et Intégration

1 1

Proxy configuration
by Jignesh Gandhi 29 Mar '16

29 Mar '16

Hello, Is it possible to use kamailio to be a stateful outbound proxy ? The use case is kamailio is an outbound proxy for multiple instances of Asterisk. Thus all responses need to go back to the same instance of Asterisk that generated them. After the call is established. How can I ensure that the Re-Invites, BYEs, etc are sent to the same instance of Asterisk that generated the outbound Invite ? Any help is appreciated. Thanks, --Jignesh

2 1

Destination Timeout (ACK message)
by Al S 29 Mar '16

29 Mar '16

Hi, The destination node is down. By adding the following code Kamailo re-routing INVITE to a new destination after getting 408 timeout (as below). failure_route[MANAGE_FAILURE]{ if (t_branch_timeout()) { # change to new destination IP $rd = "10.10.10.10"; t_relay(); exit; }...} However, the ACK message coming from source is still being send to the old destination. How can I reroute other messages in the dialog to be sent to this new destination.. Thank, AS

1 0

SIP / WebRTC / GSoC proposals
by Daniel Pocock 29 Mar '16

29 Mar '16

Hi all, We had a few GSoC applications[1] from students who would like to do work relating to SIP and WebRTC this summer. The students have a range of skills. Some are interested in the server side (C, C++) code. Some are interested in Android and Java projects (e.g. the Lumicall and CSipSimple dialers) and some are web developers, interested in making practical use of WebRTC. Would anybody else from the community be interested in helping to mentor one of these projects? Prospective mentors can review the proposals privately before making any firm commitment, please contact me for details. Mentors can have some input in the direction of the students' work, just as long as all code is released publicly. Regards, Daniel 1. http://danielpocock.com/gsoc-2016-voice-video-chat

2 2

XML-RPC reply doesn't contain any data
by Arsen 29 Mar '16

29 Mar '16

Hi guys! We have an issue with XML-RPC module. >From this doc: http://kamailio.org/docs/modules/4.3.x/modules/xmlrpc.html "Success replies always contain at least one return value. In our case the simplest success replies contain single boolean with value 1" We are trying to reload domain module after adding/removing new domain, so we send the following request: XX - web server; YY - kamailio T XX.XX.XX.XX:55028 -> YY.YY.YY.YY:8080 [AP] POST /RPC2 HTTP/1.1. Host: YY.YY.YY.YY:8080. Accept: */*. Accept-Encoding: deflate, gzip. User-Agent: cURL. Accept-Charset: UTF-8. Content-Length: 120. Content-Type: application/x-www-form-urlencoded. . <?xml version="1.0" encoding="iso-8859-1"?> <methodCall> <methodName>domain.reload</methodName> <params/> </methodCall> T YY.YY.YY.YY:8080 -> XX.XX.XX.XX:55028 [AP] HTTP/1.1 200 OK. Sia: SIP/2.0/TCP XX.XX.XX.XX:55028. Server: kamailio (4.3.4 (x86_64/linux)). Content-Length: 108. . <?xml version="1.0"?> <methodResponse> <params> <param> <value></value> </param> </params> </methodResponse> As you can see in the response from the kamailio there is no any data, so our script is waiting until timeout... Please advice. Thanks, -- Regards, Arsen.

4 6

Timeout Handling (update)
by Al S 28 Mar '16

28 Mar '16

Hi, I am trying to re-route to a different $ru after timeout and right before Kamailio reply with a 408.I made the changes right before sl_reply_error(), however it does not work as I expected. Any help will be appreciated. Also, $rm represents the request message received by Kamailio. Is there a variable represents the messages leaving Kamailio?. route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { // re-route to a new $ru before sending 408 sl_reply_error(); } exit;} Thanks,AS

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sr-users March 2016