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37568 discussions

Two presentations at Amoocon 2010
by Daniel-Constantin Mierla 30 May '10

30 May '10

Hello, next weekend (June 4-6, 2010) in Rosotck, Germany, takes place 2010 edition of Amoocon. http://www.amoocon.com I will give two talks about Kamailio - SIP Router: - Asynchronous SIP Routing - Develop Your SIP Routing in Lua More details are posted on news section of the web sites: http://www.kamailio.org/w/2010/05/kamailio-at-amoocon-2010/ If you are around and want to meet, drop me an email. 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/

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Re: [SR-Users] Dialplan modinit failed
by Daniel-Constantin Mierla 30 May '10

30 May '10

Hello, please use the sr-users(a)lists.sip-router.org to post questions about ser or kamailio - apart of first one, most private messages are discarded. There are many people that can answer, faster and better. In this particular case, seems you haven't loaded pv module (located in modules_k). Cheers, Daniel On 5/30/10 9:50 AM, Dmitry Kirillov wrote: > Hi > > i tried to start SIP-Router (ser-3.0.2_src_2010-05-29) by including > dialplan.so in the list of loaded > modules and empty dialplan modparam. when i start sip-router, i get > > 0(23147) ERROR: <core> [pvapi.c:501]: error searching pvar "ruri.user" > 0(23147) ERROR: <core> [pvapi.c:705]: wrong char [r/114] in > [$ruri.user] at [9 (0)] > 0(23147) ERROR: dialplan [dialplan.c:163]: input pv is invalid > 0(23147) ERROR: <core> [sr_module.c:874]: init_mod(): Error while > initializing module dialplan > ERROR: error while initializing modules > > With modparam("dialplan", "attrs_pvar", "$avp(s:dest)"): > > 0(23158) ERROR: <core> [pvapi.c:501]: error searching pvar "avp" > 0(23158) ERROR: <core> [pvapi.c:705]: wrong char [s/115] in > [$avp(s:dest)] at [5 (5)] > 0(23158) ERROR: dialplan [dialplan.c:149]: invalid pvar name > 0(23158) ERROR: <core> [sr_module.c:874]: init_mod(): Error while > initializing module dialplan > ERROR: error while initializing modules > > there is no dialplan function call in ser.cfg. > > Please, help > -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

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Siremis with grandchild(?)
by Anders 29 May '10

29 May '10

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

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Issue with starting SER
by Jesse B Scites 28 May '10

28 May '10

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

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new developer access: Jon Bonilla
by Daniel-Constantin Mierla 28 May '10

28 May '10

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/

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Multiple domains in different servers
by Ricardo Coelho 28 May '10

28 May '10

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

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continue processing route script if no response from gateway
by JR Richardson 28 May '10

28 May '10

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

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kamailio 3.0.2 for Thu, May 27, 2010
by Daniel-Constantin Mierla 28 May '10

28 May '10

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/

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RPID Question
by dotnetdub 28 May '10

28 May '10

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.

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new feature: embedded http 1.0 server
by Daniel-Constantin Mierla 28 May '10

28 May '10

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/

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