sr-users May 2010

sr-users@lists.kamailio.org

67 participants
106 discussions

by Anders

Hi, This might not be directly Siremis-related, but in any case it is a consequence of using Siremis: The stored procedure for calculating and populating the 'cdrs' table has started to act out all of a sudden - it won't do anything. In my syslog it looks like this: May 29 09:24:01 vae /USR/SBIN/CRON[25555]: (root) CMD (echo "call kamailio_cdrs();"|/usr/bin/mysql -u root -pxxx kamailio >/dev/null) May 29 09:24:23 vae /USR/SBIN/CRON[25548]: (CRON) error (grandchild #25549 failed with exit status 1) May 29 09:24:23 vae /USR/SBIN/CRON[25548]: (CRON) info (No MTA installed, discarding output) ..and that's every time the cron job runs, the only thing that changes is the "grandchild" number. I have no clue what "grandchild" means in this context - ...anyone? Thanks!! //Anders

14 years, 5 months

Issue with starting SER

by Jesse B Scites

I am not sure if this mailing list is for any type of product support, but I am at a loss as how to continue, so any type of guidance would be appreciated. I am new to using SER, however I downloaded the source baseline, then made it, etc. according to the "Getting Started" guide. The very basic configuration (minus MySQL, etc.) seemed to build fine and all binaries, the ser.cfg file, etc. were installed to the correct locations. Since I just wanted to start using a basic configuration, I chose to use the "Hello World" configuration file as outlined in the guide, so I swapped it out with the default one. As the super user, I ran >>ser -V and received the expected version feedback as listed within the "Getting Started" guide. However, when I tried to start SER normally with >>ser I received the following error: 0(6989) parse error (11,1-5): syntax error 0(6989) parse error (11,1-5): ERROR: bad config file (2 errors) So I went back and checked the "Hello World" config file and I could find no syntax errors since the exact replica can be compared to the example in the "Getting Started" guide. So I swapped back in the original ser.cfg file and I still got errors. Of course I also got additional errors with that file since I did not have MySQL, etc configured. So I guess my question would be, what may be causing SER to error out on the config file (I assume it means there is some issue with parsing out the contents of the ser.cfg file). I am using Fedora 11 on an i386 platform with gcc 4.4.1, bison 2.4.1, flex and all other needed dependencies listed within the INSTALL file. Any suggestions would be much appreciated. Regards, Jesse Scites

14 years, 5 months

new developer access: Jon Bonilla

by Daniel-Constantin Mierla

Hello, lately, most of the bugs related to Debian packaging were filled by Jon Bonilla (Manwe), so the solution was to ask him to join and help to fix/improve the spec files. Jon has been around for quite some time, working with other SIP related projects as well. I am looking forward to his contributions to packaging system, part which was not much under focus lately, but needs to be brought up-to-date to ease the installation on various OSes. His commit id is: manwe Welcome Jon! Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

14 years, 5 months

Multiple domains in different servers

by Ricardo Coelho

Hi, Right now I have 2 machines (one on domainA and the other on domainB). I want to allow a phone registered in the openser from domainA to call a phone registered in the openser from domainB (and vice-versa). I have already inserted both domains in both openser but when I try to call a "user not found" message is returned. Can anyone help me? Thanks in advanced, Ricardo Coelho

14 years, 5 months

continue processing route script if no response from gateway

by JR Richardson

Hi All, I am lab testing carrierroute modue on kamailio 1.5.4-notls (i386/linux) and have a question on how to continue processing a call if kamailio sends a call with t_relay() but does not get a response from the gateway. I read about the timers in the tm module and set them pretty low in the config file: # ----- tm params ----- modparam("tm", "fr_timer", 2) modparam("tm", "fr_inv_timer_next", 3) But nothing happens in the routing script after an invite is sent and there is no response. Debug just shows the invite to the gateway being re-transmited several times. Any ideas? Thanks. JR -- JR Richardson Engineering for the Masses

14 years, 5 months

kamailio 3.0.2 for Thu, May 27, 2010

by Daniel-Constantin Mierla

Hello, I am planning to release kamailio 3.0.2 this Thursday. There were some fixes since 3.0.1 that worth to be packaged. If you have major reports for current stable version, please write to sr-dev(a)lists.sip-router.org Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

14 years, 5 months

RPID Question

by dotnetdub

Hi All, Kamailio 3 I currently store a default RPID in our database. We allow some asterisk customers to send us the RPID. I want to store the RPID sent by the customer on the radius record. We test: if!(is_present_hf("Remote-Party-ID")) { #RPID Not Set by Asterisk - Will set now from DB append_rpid_hf(); } Do I use textops to get the RPID that the Asterisk sends and then overwrite what we read from the database? $avp(s:rpid) = ? Not really sure how to extract the RPID from the SIP message sent from the asterisk. TIA. Stephen.

14 years, 5 months

new feature: embedded http 1.0 server

by Daniel-Constantin Mierla

Hello, there is a new module named xhttp which provides an embedded http server inside the SIP server. It reuses the transport layer from the core, so you can send http requests via TCP and TLS. The port is the same as for SIP traffic. The documentation of the module is available at: http://sip-router.org/docbook/sip-router/branch/master/modules/xhttp/xhttp.… Since sip and http are very similar, you can use other modules in handling the http requests. For example, you can do www-authentication: event_route[xhttp:request] { if (!www_authorize("xcap", "subscriber")) { www_challenge("sip-router.org", "0"); exit; } xhttp_reply("200", "OK", "text/html", "<html><body>Welcome $au, you are connected from $si:$sp</body></html>"); } allowing in this way only local SIP users to do http requests. For now supports only http 1.0, meaning it cannot take chunked requests (usually happening in PUT requests with large body). Hope it is useful for some of you. Its main goal right now is to offer the transport layer for embedded xcap server (the new xcap_server module), but it can be used for service monitoring, server control, etc. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

14 years, 5 months

Kamailio v3.0.2 Released

by Daniel-Constantin Mierla

Hello, Kamailio v3.0.2 was just released. It is a patch release that includes the fixes done since release of v3.0.1. You should consider to upgrade if you run v3.0.0 or or v3.0.2, there is no change that you should do to configuration file or database, just reinstall the binaries. Source tarballs are available at: http://www.kamailio.org/pub/kamailio/3.0.2/src/ Detailed changelog: http://www.kamailio.org/pub/kamailio/3.0.2/ChangeLog Download via GIT: # git clone –depth 1 git://git.sip-router.org/sip-router kamailio # cd kamailio # git checkout -b kamailio_3.0 origin/kamailio_3.0 Binaries and packages will be uploaded at : http://www.kamailio.org/pub/kamailio/3.0.2/ Modules’ documentation: http://www.kamailio.org/docs/modules/3.0.x/ What is new in 3.0.x release series is summarized in the announcement of v3.0.0: http://www.kamailio.org/w/kamailio-openser-v3.0.0-release-notes/ If you are eager to see what's up to next major release (3.1.0), visit: http://sip-router.org/wiki/features/new-in-devel Enjoy, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

14 years, 5 months

Carrierroute failover domain not working

by JR Richardson

Hi All, I'm testing carrierroute modue in kamailio 1.5.4-notls (i386/linux). I have things working like this: sipp><kamailio carrierroute>< 2 asterisk servers A & B even distribution of calls So A & B are in carrier1 domain1, I have failureroute setup with 2 more asterisk servers C& D in carrier1 domain2. While I'm processing calls from sipp and sending to A & B, I can fail A and then all calls go to B as expected. But if I fail both A & B, the calls fail, so I'm guessing I don't have the faliure_route configured correctly. Here is my current config: route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if (!allow_trusted()) { sl_send_reply("403", "Forbidden"); exit; }; if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); route(2); } route[1] { if (!t_relay()) { sl_reply_error(); } exit; } route[2] { if(!cr_route("1", "1", "$rU", "$rU", "call_id")){ sl_send_reply("403", "Not allowed"); } else { t_on_failure("2"); route(1); } } failure_route[2] { if (t_check_status("408|5[0-9][0-9]")) { if(!cr_route("1", "2", "$rU", "$rU", "call_id")){ t_reply("403", "Not allowed"); } else { t_relay(); } } } I think the database entries are setup ok, I used the example in the module doc, but I can share that if needed. Any guidance will be appriciated. Thanks. JR -- JR Richardson Engineering for the Masses

14 years, 5 months

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users May 2010