Hi,
a new release of Siremis is out - v4.0.0 - web management interface that
is compatible out of the box with Kamailio v4.0.x series.
More details, including link to install tutorial, are available at:
- http://siremis.asipto.com/2013/05/08/siremis-v4-0-0-released/
Regards,
Ramona
Hi,
I can configure kamailio + rtpproxy to enable calling between user behind the NAT. Thanks for the example config, it is very easy to did it.
Do we need rtprpoxy for all kind of NAT?
From what i have read, only symmetric NAT that hard to be traversal-ed. While other type of NAT can be traversaled using STUN.
But my device that behind openwrt router can't work without rtpproxy. Is it expected behaviour?
Or i'm doing something wrong in the config?
Thanks
Hi all,
Posted a similar query a few weeks ago, without much interest - any advice appreciated.
I have two sites and will send calls between them. I have Kamailio at each site which will route the calls out/in.
There are multiple distinct network routes between the sites, accessible via different IP addresses. Each Kamailio has multiple IP's, one for each route.
The purpose of the multiple routes is mainly fault tolerance. Some of the network links are unreliable, so routing must adapt when route(s) are unavailable. When all routes are available, all should handle some traffic, at differing ratios to match the bandwidth available to each route (e.g route A - 50%, route B - 30%, route C - 20%).
I know that the Dispatcher can manage the routing for the SIP traffic, with the %ge distribution, and with SIP OPTIONS 'pings' to detect route availability.
My main headache is that RTP must follow the same route as SIP for each call.
After a bit of web digging, I was thinking of a solution where each of the Kamailio servers will run multiple instances of rtpproxy (one for each ip/route). Then once the dispatcher has chosen a route for the call, to use the matching rtpproxy instance to direct the audio.
Any comments or alternate solutions/suggestions would be of interest.
Many thanks,
Mark
Hello,
I followed the step by step guide (http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb) that describe the realtime integration between Kamailio and Asterisk. I have no problem with registration but when I try a call from 101 to 102 I get the followng error:
[Mar 31 01:18:44] NOTICE[32330][C-00000006]: chan_sip.c:25195 handle_request_invite: Call from '101' (192.168.1.100:5060) to extension '103' rejected because extension not found in context 'DEFAULT NULL'.
Kamailio and Asterisk are running in the same machine.
Any idea about the cause of this problem?
Best Regards,
Theo
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Hi,
I have built a kamailio based phone system along the lines of the
tutorial below. I have updated it for version 4.0 and it is running
successfully on Debian 7.0. I have set my system up with a Gandi SSL
cert rather than a self signed one.
http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour
I want to federate/peer with another proxy running Kamailio with TLS.
What do I need to do to make this work? Thanks.
Regards,
John
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Hi,
What is the maximum number of concurrent TCP connection that Kamailio can
handle ?
I see the max_tcp_connections (which is set to 2048), is that the answer ?
How to test for this ?
Many thanks
--
Khoa Pham
HCMC University of Science
Faculty of Information Technology
Hi
My application is for mobile (MVNO) users making calls, which will generally end up on the PSTN via our carriers.
MVNO Carrier --> Our Edge Switch --> Our PSTN Switch --> Our carrier's switch
|
|---> Our internal routing switch
The issue is with our PSTN switch and the fact that it is not staying in the SIP signalling path, so when the call ultimately between our MVNO carrier and outbound Carrier is established (200 OK) the MVNO carrier and PSTN carrier begin talking to each other.
When the MVNO carrier issues a BYE to the outbound carrier, the outbound carrier does not then receive this packet as they are firewalled (and always will be).
What is the correct method of relaying calls through Kamailio but not passing on the Contact: header info? I have read that forcing a change of Contact is not the right way.
Leo
Hello,
among the topics discussed just before the last major release series
(4.0.x) was one about restructuring the source code tree. It started
mainly as a proposal to move source code belonging to core in a
dedicated folder, but there could be more variants. It's time start
discussing if we do it, and if yes, how.
Here is what I could collect so far.
a) no change, keep like it is
b) move source code of core in a dedicated folder named 'core'
c) move all source code in a folder named src, with following sub-structure
src/core - for core
src/lib - for internal libraries
src/modules - for module
Feel free to propose other variants or express your opinion regarding
the proposed options.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
Hi all,
I use ims_icscf in Kamailio to send LIR message to HSS. But, In LIR message
that I received in HSS haven't OriginatingRequest AVP.
How I can insert it into LIR message?
Thanks & Best regards,
Khue Nguyen.
