sr-users May 2011

sr-users@lists.kamailio.org

78 participants
129 discussions

by hb

Dear：Bucur Marius： Thank you for your help. As you said the presence module does not work if I disable it's module. This time I used wireshark to get the packages. I found when I disable the presence module, one point send the "Notify" packages directly to the other, and there were no changes in the watchers table. And when I enable the module, I found it can create some lists in watchers. But at this time , I found there is nothing contents in packages like "Subscribe" and "Notify".What's more when one point change the state, it dose not send any packages to the server. You said active_watchers will have something when the subscribe is authorised. What dose it mean? My server have not authorised the subscribe? But it can have a call with other subscribe. Look foreword your help. haibo huang

13 years, 6 months

INSTALL file

by Claudio Furrer

Hello, Is this still applicable? See INSTALL file (line 669) 3.1) create MySQL tables for SER flavour - if you have a previously installed SER on your system, use /usr/sbin/ser_mysql.sh reinstall to convert your SER database into new structures - otherwise, if this is your very first installation, use /usr/sbin/ser_mysql.sh create to create SER database structures (you will be prompted for password of MySql "root" user) Because I did not found the *reinstall* option on sources.. Thanks, Claudio

13 years, 6 months

： about presence

by hb

Dear everyone: I use kamalio 3.1. I want to add the xcap_sever and presence modules into my kamailio. I followed the lists "http://cache.baidu.com/c?m=9d78d513d99a12f00bb6ca255001d7175b14d7226b8b8142…" step by step. The only difference is that i use the database postgresql. When i start the server, it worked well, but the softphone was not able to show the presence of the contacts. I tried many times and found that when i disable the sentence "#!define WITH_PRECENCE" in "kamailio.cfg", it can show the state of the contacts. However, when i check the database, there was nothing in "Active Watchers" and "Presentity" tables. I think there must have something wrong with the configure of presence. Any response will be appreciated. ps: the configuration of the presence in kamailio.cfg as follows(same as theTutorial): #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) modparam("presence", "fallback2db", 1) modparam("presence", "db_update_period", 20) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 0) modparam("presence_xml", "integrated_xcap_server", 1) #!endif haibo huang

13 years, 6 months

Error compiling Kamailio with mi_xmlrpc

by Lucas Alvarez

Hi, I'm trying to compile kamailio with mi_xmlrpc, I've followed the instructions, first installing xmlrpc-c-0.9.1.0 and xmlrpc-c-dev-0.9.1.0 but the folder xmlrpc-c in /usr/include is not created, so kamailio couldn't be compiled. Then I've installed the latest version of xmlrpc-c-dev, in this case the folder xmlrpc-c with the headers was created but the module didn;t compile, this are the errors: Makefile:46: You are using an unsupported libxmlrpc-c3 version (1.16.34), compile at your own risk! CC (gcc) [M mi_xmlrpc.so] abyss_data.o abyss_data.c:218: error: expected ')' before '*' token abyss_data.c:235: error: expected ')' before '*' token abyss_data.c:248: error: expected ')' before '*' token abyss_data.c:286: error: expected ')' before '*' token abyss_data.c:298: error: expected ')' before '*' token abyss_data.c:311: error: expected ')' before '*' token abyss_data.c:325: error: expected ')' before '*' token abyss_data.c:331: error: expected ')' before '*' token make: *** [abyss_data.o] Error 1 I would appreciate if someone can point me through the right path. Thanks in advance. Lucas Alvarez

13 years, 6 months

i want to setup a SIP proxy and i dont know where to start.

by Eliezer Croitoru

i have a main office and a branch office. the main office and the branch office connected via vpn connection the branch office is natted to the main office. so i want to put on the gateway machine a SIP proxy server to nat the sip connection and also to proxy all the connections to SIP uses of the office. means i have one asterisk machine on the main site and other SIP providers that in a case i want to use i can use them also without connecting to the main office. i dont need asterisk on the branch office. the gateway server is ubuntu 10.04 amd64 what i should use Kamailio or the openser on the ubuntu repos? for now the Kamailio is installed.(not working). another feature i want to add is to be able to make calls between SIP phones using this server such as berni(a)mydomain.com from outside the office and vise verse... to dial people at bob(a)otherdomain.com hope for little help Eliezer

13 years, 6 months

Kamailio performance testing

by Martin Pohančeník

Hi I am trying to test performance of kamailio server. I have made some tests on old hardware, that was ok. I found that it can handle around 450 calls per second with authorization. CPU utilisation was up to 100%. That testing was OK, because server's CPU utilization was increasing. Now I want to do some test with new hardware and do some comparisons of those two hardware configurations. I have made some test but i found that it can handle up to 540 calls per second with authorization. But CPU utilisation was only around 8%. When I increased call rate server responded with 404 Not Here and 500 I am terribly busy messages. I think that it is because of that it is not capable of finding a user location that fast. Can i do something to increase the performacne of this server? I am using SIPp testing tool for generating calls. I have registered users manualy with kamctl ul add sipp sip:sipp@192.168.1.1:5060 command. I was using this tutorial: http://www.troubleshootingwiki.org/OpenSER#Stress_Test_The_SIP_Signaling. And also i made my own scenarios. Any suggestion would be usefull. Thanks.

13 years, 6 months

New module ipops

by Iñaki Baz Castillo

Hi, I've commited a new module named 'ipops' which provides operations with IPv4 and IPv6: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=5940394… The README describes the module functions: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=modules/… The module has no dependecies on other siprouter modules or external libraries, so if it's ok for you it could be enabled by default. Cheers. -- Iñaki Baz Castillo <ibc(a)aliax.net>

13 years, 6 months

SIP outbound

by Klaus Darilion

Hi! IIRC we recently had a discussion about SIP outbound [1] and the benefits of an implementation in Kamailio. Just playing around with a SIP client on an iphone I discovered a usecase. Whenever the iphone gets a new IP address (quite often as it often changes between WiFi and 3G) the client reREGISTERs with the new IP address (establishing a new TCP connection). This leads to lots of "broken" registrations and TCP connections [2]. By use of SIP outbound and re-registration using the same "reg-id" the proxy would be able to replace the old registration with the new registration. Further, if Kamailio would have a mapping from registrations to TCP connections it could also close the dead TCP connections. regards Klaus [1] http://tools.ietf.org/html/rfc5626 [2] I do not know if TCP keepalives (socket option) could detect broken TCP connections without making too much noise on the receiver side (activating the background application and draining battery)

13 years, 6 months

Kamailio v3.1.3 in Action

by Andreas Granig

Hi all, If you want to see a full-blown Kamailio v3.1.3 based open-source VoIP system in action, check our latest sip:provider CE v2.2-rc1, which has been released today: http://www.sipwise.com/news/announcements/spce-v2_2rc1-release/ It leverages the capabilities of Kamailio to build a quite complex SIP routing engine using a SIP load-balancer and a SIP proxy/registrar, as well as SEMS acting as b2bua. On top of it, it provides provisioning and billing modules and admin and user web interfaces to form an out-of-the-box VoIP soft-switch. Feedback is very much appreciated. Andreas

13 years, 6 months

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sr-users May 2011