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37519 discussions

whats happened to iptel.org
by voipspace voipspace 29 Jan '18

29 Jan '18

Anyone know whats happened to iptel.org The webiste is now a wix page iptel.org the sip uri music(a)iptel.org was a useful resource. Now the song has changed and the service is down alot Has it it changed hands or been sold?

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i can not receive BYE after websocket has been closed and reconnected during the call
by Kenan Kocaerkek 29 Jan '18

29 Jan '18

Hi, I have a WebRTC softphone based on JsSIP and kamailio 4.4.7. Generally it works well but sometimes websocket connections interrupt and reconnect immediately (especially running behind proxies). If the interruption occurs during an active call, kamailio tries to send BYE message to softphone over closed websocket. gruu_enabled flag is active on registrar module and JsSIP sends proper sip.instance value. So after ws reconnects, registered contact details do not change on kamailio except Expires value. Is it possible to send BYE message via contact's last active websocket after reconnect? Jan 29 14:35:20 registrar-kenan /usr/local/sbin/kamailio[30357]: WARNING: <script>: WebSocket connection from 195.142.112.66:49086 has closed Jan 29 14:35:24 registrar-kenan /usr/local/sbin/kamailio[30323]: WARNING: <core> [msg_translator.c:2761]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jan 29 14:35:24 registrar-kenan /usr/local/sbin/kamailio[30323]: ERROR: <core> [msg_translator.c:1979]: build_req_buf_from_sip_req(): could not create Via header Jan 29 14:35:24 registrar-kenan /usr/local/sbin/kamailio[30323]: ERROR: tm [t_fwd.c:462]: prepare_new_uac(): could not build request Jan 29 14:35:24 registrar-kenan /usr/local/sbin/kamailio[30323]: ERROR: tm [t_fwd.c:1723]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Jan 29 14:35:24 registrar-kenan /usr/local/sbin/kamailio[30323]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: No error (2/SL) Regards, Kenan

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Ann: Registration to Kamailio World 2018 is open!
by Daniel-Constantin Mierla 29 Jan '18

29 Jan '18

Hello, a quick note to the community forums to announce that the registration to next Kamailio World Conference (May 14-17, 2018, in Berlin, Germany) is now open! More details are available at: * https://www.kamailioworld.com/k06/registration/ In a matter of several days, we expect to publish the details about the first group of accepted speakers. Looking forward to meeting many of you in Berlin! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

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New module – embedded MSRP relay
by kossivi HOLONOU 28 Jan '18

28 Jan '18

Hello,I would like to develop a project that will allow me to: - voice, video, screen sharing, etc. sessions with content communication via RTP - end to end presence – this is purely SIP routing - SIMPLE-based presence (aka, presence server or presence agent model) via presence* and pua* modules — user presence, dialog states notification (aka, blinking lamps), resource lists service (including OMA/RCS extensions), user location states notification and replication, audio/video conference mixer notifications, a.s.o. - embedded XCAP server – management of user contact lists, presence policies, user agent configuration files, a.s.o. There is also an XCAP client extension - embedded HTTP server – for admin and user interaction with the service via pure HTTP or XMLRPC requests - embedded MSRP relay – for relaying and fine controlling of the message-based content of SIP sessions - IRC-style instant messaging conference via imc module - storage of instant messages for offline users and relay to them when they become again online via msilo module I need your help to start. I went through the document a little without understanding much. Thanks for your help

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Kamailio with asterisk on private lan
by Mark Boyce 28 Jan '18

28 Jan '18

Hi all I’m trying to create a relatively simple setup with Kamailio dual homed on public/private ip and asterisk on private ip only. The idea is load balance / fail over asterisk boxes. Following the real-time tutorial I have clients registering with Kamailio, Kamailio registering on clients behalf with asterisk as well as invites going through. However what I’m seeing is that when an invite occurs asterisk offers media on its private ip, as it would. However this is making its way through Kamailio all the way to the client. After a bit of searching all I can find is people trying to get it working and failing, or putting asterisk on public IP. So questions - am I doing this completely the wrong way? Should Kamailio alter the media ip of asterisk on the way through or do I need to do that by hand? Surely someone somewhere has a write up on this already :-) Thanks Mark

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SEMS with sbc application
by Mojtaba 27 Jan '18

27 Jan '18

Hello, I have a problem today, It's strange for me. Suppose we have this senario: uac1------->SEMS(mo profile)------->Kamailio-------->SEMS(mt profile)---------->uac2 In above topology, we have two interfaces(intern,extern) for SEMS, and just used as SBC (sbc application). if i used port=5060 as external port, every things is right and log file is like this: [#7fed3f9f9700/32820] [run, udp_trsp.cpp:352] DEBUG: vv M [|] u recvd msg via UDP from 89.165.117.125:42411 vv --++-- REGISTER sip:kava.shatel.ir;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 89.165.117.125:42411;branch=z9hG4bK-d8754z-021bd8b61efc7ac0-1---d8754z- Max-Forwards: 70 Contact: <sip:4000@ 89.165.117.125:42411;rinstance=79011092e56e1a09;transport=UDP> To: <sip:4000@kava.shatel.ir;transport=UDP> From: <sip:4000@kava.shatel.ir;transport=UDP>;tag=82820e1f Call-ID: Y2U4YThiYjEwNTUzMzliZTIwNWZkMDI3MTM4OTZlNWU. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Allow-Events: presence, kpml Content-Length: 0 but when i changed port=4080 for external port, The Via header and contact header are changed to my public ip, like this: [#7fed3f9f9700/32820] [run, udp_trsp.cpp:352] DEBUG: vv M [|] u recvd msg via UDP from 89.165.117.125:42411 vv --++-- REGISTER sip:kava.shatel.ir;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 172.1.1.125:42411;branch=z9hG4bK-d8754z-021bd8b61efc7ac0-1---d8754z- Max-Forwards: 70 Contact: <sip:4000@172.1.1.125:42411;rinstance=79011092e56e1a09;transport=UDP> To: <sip:4000@kava.shatel.ir;transport=UDP> From: <sip:4000@kava.shatel.ir;transport=UDP>;tag=82820e1f Call-ID: Y2U4YThiYjEwNTUzMzliZTIwNWZkMDI3MTM4OTZlNWU. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Allow-Events: presence, kpml Content-Length: 0 Why my public ip address is there here? I just changed external port=4080 in sems server. In SDP protocol i have the same problem when i changed external ip port in sems server. Thanks. --Mojtaba Esfandiari.S

3 3

Transcoding
by Wilkins, Steve 26 Jan '18

26 Jan '18

Hello All, I am currently using Kamailio and Asterisk on Centos 7 servers and trying to enable WebRTC jsSIP clients to be able to do Audio/Video calls with Provider Phones (Purple, Z, Sorenson, etc....), however, the providers do not have vp8 codecs (which is what the WebRTC clients use for Audio) so I believe I will need a media proxy server to resolve the video issues. My question is, can rtpproxy or rtpengine perform this transcoding? If so, and if rtpengine is the way to go, should I use Ubuntu for the rtpengine since it is the only one that seems to have a working installation? Thank you, -Steve

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Problem to install Kameilio from Ubuntu packages
by Alain Bonnefoy - XOOL 26 Jan '18

26 Jan '18

Hello, I tried to install Kameilio on Ubuntu 16.04 from what explained in the wiki. But when I execute the command to add your GPG key, I just get replied that the timeout occured. What to do ? Alain Provenance : Courrier pour Windows 10

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Kamailio log file empty
by Ashutosh chaubey 26 Jan '18

26 Jan '18

Hi, My kamailio.log file is empty, following is my configuration: In Kamailio.cfg- log_facility=LOG_LOCAL0 #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=3 log_stderror=no #!endif Inside rsyslog.conf # don't log messages with LOG_LOCAL0 in /var/log/syslog anymore *.*;auth,authpriv.none,local0.none -/var/log/syslog # # log messages with LOG_LOCAL0 in /var/log/kamailio.log local0.* -/var/log/kamailio.log User has all the privileges to write in log file. Regards, Ashutosh Chaubey

3 2

EOL of geoip module.
by Sergey Okhapkin 26 Jan '18

26 Jan '18

Maxmind just announced EOL of legacy geolite databases used by geoip module, no updates starting this April. See more at https://blog.maxmind.com/ 2018/01/02/discontinuation-of-the-geolite-legacy-databases/ If you use kamailio's geoip module, I suggest to switch to geoip2 module which uses Geolite2 databases and is a plug in replacement for geoip module.

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