sr-users January 2012

sr-users@lists.kamailio.org

72 participants
115 discussions

app_lua coredump
by noc＠nelcom-voip.com 10 Jan '12

10 Jan '12

I've installed kamailio (3.2, 3.2-nightly, dev-nightly) for debian unstable (wheezy) using repository at http://deb.kamailio.org and constantly get a coredump with this lua code: -- SIP request routing function route_request(name) sr.dbg("routing SIP request from Lua [" .. name .. "]\n") -- initial checks of SIP request if sr.maxfwd.process_maxfwd(10) < 0 then sr.sl.send_reply(483,"Too Many Hops") return end end I've started with an example at this page http://kb.asipto.com/kamailio:usage:k32-lua-routing, and then just reduced the lua script to code above. If I do comment sr.maxfwd.process_maxfwd(10) it passes. It is impossible to call sr.modf("mf_maxfwd...") because it says about fixups. What's wrong? -- With best regards, Roman Galeev.

2 14

Prepaid billing solution for a service provider
by Carlo Dimaggio 09 Jan '12

09 Jan '12

Hi all, I'm working in a project for a service provider in which prepaid is an essential requirement. They have about 12000 subscribers. The core infrastructure will be Kamailio+RTPProxy while I have some doubts about the prepaid feature. I'm thinking about a B2BUA (SEMS or Freeswitch) that is called if the user belong to a "prepaid group" and perform authorization and accounting; in this case the SIP flow could be: Kamailio -> B2BUA(Prepaid) -> Kamailio. Is a good choice? At the moment I cannot estimate the effort needed to develop this section (base prepaid feature) of the project... Do you have some hints? What are the prepaid solution implemented by you? Do you use your own developed solution or do you think that is better to choose a commercial solution (that provide also calling-card and other features)? I have seen some commercial products (Portabilling, Jerasoft/Bilberry, ...); what do you think about? Best Regards,

6 8

Kamctl reload commands
by Uri Shacked 09 Jan '12

09 Jan '12

Hi, i am using different modules like "mtree", "carrierroute", "drouting" and so. how can i tell how long did the reload takes? how do i know if it fails? is there a way to get a return code from kamailio that confirms that the reload ended and successfuly? BR, Uri

3 4

Re: [SR-Users] Topoh Log Warning
by Daniel-Constantin Mierla 09 Jan '12

09 Jan '12

Hello, On 1/6/12 10:50 PM, Javier Vidal wrote: > Hi Daniel, problem solved!!! thanks, was a bad REGISTER, i was wrong. > > Thank you!!!! welcome, jut keep the mailing list always on cc, to help others to look for solutions of similar cases. Cheers, Daniel > > El 06/01/2012, a las 19:34, Daniel-Constantin Mierla escribió: > >> Hello, >> >> On 1/6/12 6:47 PM, Javier Vidal wrote: >>> Hi all, >>> >>> Two days ago, i had activated topoh module, and now in log file, when i receive a Options, appears this error >>> >>> /usr/sbin/kamailio[29936]: WARNING: topoh [th_msg.c:1063]: Invalid/Unparsed CSeq in message. Skipping. >>> >>> I dont' know solve it, only deactivating the topoh module!! >> does the OPTIONS have the CSeq header? Can you paste here an ngrep trace of such OPTIONS? >> >> The warning message is printed from a function that tries to detect whether is a REGISTER or PUBLISH, which are requests that don't get masked since they end to a service. >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla -- http://www.asipto.com >> http://linkedin.com/in/miconda -- http://twitter.com/miconda >> -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Topoh Log Warning
by Javier Vidal 06 Jan '12

06 Jan '12

Hi all, Two days ago, i had activated topoh module, and now in log file, when i receive a Options, appears this error /usr/sbin/kamailio[29936]: WARNING: topoh [th_msg.c:1063]: Invalid/Unparsed CSeq in message. Skipping. I dont' know solve it, only deactivating the topoh module!! Sorry for my english. ;) Regards. Javier Vidal

2 1

RTPproxy on Kamailio 3.2.1 difficulty.
by nunu abe 06 Jan '12

06 Jan '12

Dear Daniel, Greetings and Happy New year to you all! I would like to thank you very much for your detailed and well structured tutorials on Kamailio. I am resending this email as I sent the earlier one before I was a member of the mailing list and I thought it may have been sent to the spam folder. If not then, I apologize for double posting. I was trying to design a similar scenario like the one " Run your own SIP VoIP service on both IPv4 and IPv6 " as explained on Asipto web-page(http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6). However I ran into difficulties with rtpproxy. The proxy is not routing media packets as it should. I followed the instruction regarding the kamailio.cfg file and edited the necessary lines according to my settings. I think my problem comes from setting up the rtpproxy. 1. I edited the /etc/default/rtpproxy file. i.e. I uncommented the line - CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" 2. I started the rtpproxy with this command: rtpproxy -F -l ADDR_IPV4 -6 /ADDR_IPV6 . However, after this command the system returns the message : rtpproxy: setbindhost: No address associated with hostname It displays the same message even when I replace ADDR_IPV4 and ADDR_IPV6 with IP addresses. 3. I have also tried using the udp socket:- CONTROL_SOCK=udp:127.0.0.1:22222, of course I have replaced the unix socket by this udp socket in the kamilio.cfg file. So my question is, what did I do wrong with the settings that SIP messages are traversing normal from IPv4 client to IPv6 client and vice versa but media files(RTP packets) are not? The rtpproxy receives the media files from one client but fails to forward them to the other client. Thanks for your help. Maedot

3 8

Kamailio 3.2 RTP repacketization
by Arif Tuhin 06 Jan '12

06 Jan '12

Hi,I'm using the new kamailio 3.2 which has some changes in rtpproxy module. And i'm finding it hard to implement a certain configuration. Because of our network infrastructure i have to repackage rtp streams in one direction. Now in kamailio 3.1.5 i used to do this in this way:In main routeroute[RTPPROXY]{if(is_method("BYE"){ unforce_rtp_proxy();else if(is_method("INVITE")){ force_rtp_proxy();}in onreply_route force_rtp_proxy("z160"); Now in kamailio 3.2 the force_rtp_proxy() is replaced by rtpproxy_manage(). Here the script reads like:main routeroute[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage("z150"); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; onreply_routethis block is called route[NATMANAGE]. Now if i put rtpproxy_manage("z150"), it tries to repackage in both direction which i do not want. So i need to write a separate block for onreply_route instead of calling route[NATMANAGE]. So please help me to write a block for onreply_route.Thanks in advance. With Best Regards Ariful Hossain Tuhin email: 1. etothepowerpi(a)gmail.com 2. etothepowerpi(a)hotmail.com 3.etothepowerpi(a)yahoo.com skype: freeburn1986

2 1

auth module realm_prefix
by Kelvin Chua 06 Jan '12

06 Jan '12

Hi guys, just wondering, what happened to realm_prefix on auth module for 3.2.x? why was it removed?

2 3

dialog module - set_dlg_profile won't work
by Ben WIlliams 05 Jan '12

05 Jan '12

Hi, can anyone see why this code wouldn't work? In the log is shows this message after set_dlg_profile called - dialog [dlg_hash.c:523]: no dialog callid='...' found Dialogs are being saved to dialog table ok but nothing is in dialog_vars table. modparam("dialog", "dlg_flag", DLGFLAG) modparam("dialog", "hash_size", 128) modparam("dialog", "default_timeout", 14400) modparam("dialog", "dlg_match_mode", 0) modparam("dialog", "table_name", "dialog") modparam("dialog", "db_url", DBURL) modparam("dialog", "db_mode", 2) modparam("dialog", "db_update_period", 10) modparam("dialog", "profiles_with_value", "caller ; allcalls") ... if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting setflag(DLGFLAG); ## add to dialog list $var(callcnt) = 0; get_profile_size("caller", "$fu", "$var(callcnt)"); xlog("$fu has $var(callcnt) concurrent calls\n"); if ($var(callcnt) >= 2) { sl_send_reply("503", "Simultaneous calls limit reached"); exit; } set_dlg_profile("caller","$fu"); } -- Thanks Ben

2 2

No memory left in send_subscribe (PUA) from update_pua while building the tm dlg_t structure
by laura testi 04 Jan '12

04 Jan '12

Hi, we are using the PUA_XMPP and PUA modules from the master branch. When the modules are started, everything are ok, the presence events from XMPP are sent to kamailio SIP servers (PUBLISH/SUBSCRIBE) and cached in the hash. But when there are several thousands records in the hash tabel, the update_pua function called in the hashT_clean gives a lot of "No memory left" error when the hashT_clean is waked up from the time: ERROR: pua [send_subscribe.c]: No memory left ERROR: pua [pua.c]: while building tm dlg_t structure The failed call is: td = (dlg_t*)pkg_malloc(size); if(td == NULL) { LM_ERR("No memory left\n"); return NULL; } in dlg_t* pua_build_dlg_t(ua_pres_t* presentity) function in send_subscribe.c. The size is about 400 and something... It's strange..... Is it the memory leak in the PUA module? I also try to increase the pkg_memory from 4MB default to 16MB, but it doesn't help. Any Idea? Thanks in advanced Laura

2 14

2025

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 January 2012