sr-users February 2014

sr-users@lists.kamailio.org

114 participants
158 discussions

One more question 'bout db_mode
by Victor V. Kustov 18 Feb '14

18 Feb '14

Hi all! my usrloc database use postgress. In config I had: modparam("usrloc", "db_mode", 1) after day of work I see strange thing: # kamctl ul show --brief | grep -c AOR 26 psql -U kamailio -c "select * from location;" ... (1 row) 26 registered users in inner kamailio's memory and only 1 in database. See in doc: 1 - Write-Through scheme. All changes to usrloc are immediately reflected in database too. This is very slow, but very reliable. Use this scheme if speed is not your priority but need to make sure that no registered contacts will be lost during crash or reboot. I think saved locations in inner memory database must be identical. Am I wrong? -- WBR, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.12.9-calculate GNU/Linux

1 0

Change table type...
by arun Jayaprakash 18 Feb '14

18 Feb '14

Hello, I am planning to change the table type in the Kamailio Mysql db from MyISAM to Innodb. Could anyone let me know if this will be an issue? Thank you. Arun

2 2

modify To header, prefix + ip, using dispacher module.
by Helena Garcia-Nieto 17 Feb '14

17 Feb '14

Hey, Thank you in advanced for the time taken on my question! I have an environment phone---sbc---kamailio---gw1, gw2 I am using kamailio 4.2.0, quite basic config, using dispatcher module to route to the gw and to monitor them with options. I have a file dispatcher.list with the two gw ips: # $Id$ # dispatcher destination sets # # line format # setit(int) destination(sip uri) flags(int,opt) priority(int,opt) attributes(str,opt) # gateways 1 sip:10.X.X.38:5060 IP 1 sip:10.X.X.39:5060 AP I have to modify the TO header, it must be prefixed and gw ip must be placed there (gw requirements) I use on the routing function ds_select_domain("1", "8") t_on_failure("RTF_DISPATCH"); and just before the ROUTE(RELAY) I have: # Add profix to gw: 0199 subst_uri('/^sip:(.*)/sip:0199\1/i'); # add prefix to RURI subst_hf("To", "/(.*):(.*)@(.*)$/\1:0199\2@$rd>/", "a"); #add prefix to TO, modify ip on TO use ruri ip It works fine if the first gw is available, but if the first option (which is the last gw on the list) is not accessible, it waits for the 30s and then tries the second one, but the ip on the TO header is not correct anymore, since it is the first gw and not the second. Question: why does it wait until the timeout if it is monitoring it with option and seeing it is not reachable? I tried modifying again the TO header with: subst_hf("To", "/(.*)@(.*)$/\1@$rd>/", "a") before the route(relay) on the failure routine, but it does not work properly, I am getting strange TO header like: To: <sip:0199XXXX@10.X.X.39> <sip:XXXXXX@10.X.X.38> It looks like it is not parsing the to header it has before but adding a parsed version of the original header at the end of the other one.... I tried with subst as well but the result is more or less the same. Probably there is a better way to set the prefix and the correct IP on the to header... I'll appreciate any help with this issue. Thanks again for your time Helena Helena Garcia-Nieto de Mir Morodo Group VoIP Department E-mail: helena.gnieto(a)morodo.co.uk Morodo Limited voip(a)morodo.co.uk / www.mo-call.com Place of Registration: Companies House, England & Wales / Registered Office: 7 Breasy Place, 9 Burroughs Gardens Hendon, London NW4 4AU

4 12

deb.kamailio.org down?
by Paul Belanger 17 Feb '14

17 Feb '14

Anybody know why deb.kamailio.org is down? Is there a mirror setup some place else? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

3 2

Re: [SR-Users] Msilo message body type
by sipatse＠in.gr 16 Feb '14

16 Feb '14

2 1

Kamailio call forward help
by Sajan Thomas 16 Feb '14

16 Feb '14

Hi sir, I have 2 server running with kamailio.I have tested in both with creating two users and chat using jitsi client. Please help me with 1.How can i implement kamailio call forward? suppose if make achat or call to kamailio server on user shoud can access the same from kamailio server 2. 2.Is there any possible modules are available for kamailio Please help me i am really got stuck here. Expecting your kind reply.

1 1

Error with ACK from SIPML5
by Jayesh Nambiar 14 Feb '14

14 Feb '14

Hello All, I have a setup which is SIPML5 -> Kamailio(Websocket) -> Freeswitch. All I am trying to do is initiate a call from SIPML5 and Play a Music file from freeswitch. This works, but the call hangs up after 30 seconds due to ACK timeout from from the SIPML5. What I can see is the SIPML5 from Chrome does send an ACK on websocket, but I see the following error in kamailio syslog: *via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found* *ERROR: <core> [msg_translator.c:1725]: build_req_buf_from_sip_req(): could not create Via header* *ERROR: <core> [forward.c:607]: forward_request(): ERROR: forward_request: building failed* *ERROR: sl [sl_funcs.c:371]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL)* Can someone help me understand what I am missing here and why kamailio is not able to create a VIA header to forward this request towards freeswitch. I have built a simple config file by following https://gist.github.com/jesusprubio/4066845 as reference. Any help in understanding this is really appreciated. Thanks, --- Jayesh

4 11

websocket.so module not generated
by Alvaro Urionabarrenetxea Gomez 14 Feb '14

14 Feb '14

Hi. I have a problem when I'm trying to start with my Kamailio server. I have an error about not existing "websocket.so" module. Here is the error: root@kamailio:# /etc/init.d/kamailio start ** Not starting Kamailio SIP server: invalid configuration file!* *** ** 0 (8327) ERROR: <core> [sr_module.c:587]: load_module(): ERROR: load_module: could not find module <websocket> in </usr/local/lib/kamailio/modules/>* *0(8327) : <core> [cfg.y:3408]: yyerror_at(): parse_error in config file /usr/local/etc/kamailio/kamailio.cfg, line 318, column 12-25: failed to load module* *ERROR: bad config file (1 errors)* *** And here is the line 318 in kamailio.cfg: *loadmodule "websocket.so"* I looked for that module in the correspondent folder but isn't there. Can anyone upload the websocket.so file to another mirror, MEGA for example? I tried to generate another time, download another kamailio versions but the error stills there. Much apreciated. Thanks!

1 0

FW: Client site NAT negotiation with no STUN.
by Rob Moore 14 Feb '14

14 Feb '14

Hi there. So I'm fighting what must be a bit of a common problem here but the usual googling isn't really helping me out. A bit of background: Classic Nat scenario of a sip handset behind a NAT firewall over the internet to VoIP SBC setup using Kamailio 3.3 and RTPProxy and Freeswitch as the registrar / calls server behind kamailio. (please excuse the beautiful ASCI art) -------------- ---- ----| RTPproxy |------- --------------- ------------ | -------------- | --------------- |SIP Phone|--------|Firewall |---------internet-------------| kamailio |-----------|Freeswitch| --------------- ------------ -------------- --------------- SIP Phones can register fine (over TCP or TLS, Kam offloads TLS if used) however when establishing a call the SIPPhone sends its local network IP address within the SDP instead of the firewalls external address. Causing media to me miss directed by the RTP Proxy. I've put a phone in with STUN to check that this is the only issue and that phone works fine .Usually in this situation we would just use STUN or TURN / ICE to get around the issue however the handsets that we are working with (and have to work with for other reasons) do not support any of these technologies. So I've been forced to try and find another solution. I was hoping to find some method within kamailio / RTPproxy to resolve this. I could just rewrite the SDP in certain situations but that feels a little bit brutal if I'm honest. I've heard of RTPproxy being able to do some sort of rtp latching where it can correct its destination ip address after it receives its first rtp packets from the SIP Phone after being forwarded by the onsite firewall but other than this 'word of mouth' suggestion I've not managed to find any detail of this functionality in the usual online resources. Can anyone suggest any other potential methods for getting around this problem? Many Thanks Rob

2 3

SSL connection issue
by Owais ul Haq 14 Feb '14

14 Feb '14

Hello, I have installed kamailio server. Now I am trying to use SSL for my server - client connections. I have made certificates succesfully. Now when my client connects, I get the following error. [tcp_read.c:882] : ERROR : TLS accept : error : 1408F10B : SSL routines:SSL3_GET_RECORD:wrong version number I have tried finding solution online. I am asked to change the version number. Can some one please tell me where I have to change the version number ?? And change it to what ?? Thanks. Cheers, Owais

2 2

← Newer
1
...
6
7
8
9
10
11
12
...
16
Older →

Jump to page:

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 February 2014