Hello.
I feel like a total idiot.
I was inspecting the various GitHub branches for Kamailio using a GitHub
visual client, and I think I managed to accidentally merge a branch back
into master.
The branch I was inspecting was 'lazedo/presence_improvements'.
I was inspecting diffs and branches, and was amazed that I was able to
screw up so badly with one single click, but I guess that's the danger of a
visual git client.
Anyway, before trying to do anything more, I figured I'd asked for advice,
because I don't want to screw up further.
Hopefully there's a simple way to revert the accidental merge, but I first
wanted to confirm that I actually did make that mistake, and that it's
visible to others.
Thanks.
Hi list
I have problem with invite packet in some wifi networks because the invite
packet is less than 1000bytes(frame size) and the access point not redirect
to softphone, this scenario happen with incoming calls to my DID, the
packets come from the provider to kamailio with a size more than 1000 and
when kamailio redirect to softphone sometimes is less than 1000.
how i could solve this issue?.
thanks for any help.
Hello !
I'm using Kamailio for 2 months, and I have a problem with the routing logic.
My aim is to use 2 Kamailio servers (in a cluster, so I have a primary and a backup server with a Virtual IP that can switch between the 2 servers) working with 2 medias servers (same as above, a primary and a backup). The Kamailio servers are here to handle failover, so all the requests go through them, and then are sent to the active media server.
My question is : Which module(s) should I load to transmit all the requests (including registers) to my media server, through the Kamailio server ? Any particular parameters ? Then, what will be in the "request_route" ?
I'm using Kamailio 4.3.4.
Best regards,
Alexandre
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Thank you.
Hello there,
Can anyone tell me what is the exactly memory size that kamailio use for
one sip register saved on memory?
How much memory i need allocate to kamailio in order to support 50.000
subscribers.
Best Regards
Hello,
only few days left, so we need to nail the location down for Kamailio
dinner at Fosdem.
One suggestion in walking distance is:
- http://restaurant-italien-picotin-bruxelles.be/en
I understood that it was used in the past also for Fosdem Speaker's
Dinner, so it might be busy and we can't get the seats. So new
suggestions are more than welcome in order to have some backups to try!
If there is no strong preference for something else by end of today,
tomorrow we should attempt to do the reservation at this place.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
Hello everyone. I'm a student and I'm working on project which about SIP communication and PBXs with IP phones.
So, my first test results were OK with JITSI. But I think I need Freeswitch for working with IP phones. In a word, I need configure an integration Kamailio and Freeswitch.
First, I setup Debian 8.3 Jessie and installed Kamailio on this my local server, after that I installed another Debian 8.3 Jessie (from same image) as my second server and it was going to run Freeswitch on itself.
Then I tried to send my communication requests to my Freeswitch server. I made some tests, my Jitsi accounts which are set in Kamailio server were done. I've read Kamailio 3.2 and Freeswitch integration from the Kamailio wiki and I applied exactly the same thing but I did not work.
When I tried to reach another IP phone from my IP phone (I registered them) phone always say "NOT FOUND". I'm sure I did my configuration properly.
So, how can I figure it out these things? How can I understand, does Kamailio routes to Freeswitch to my phones?
Best regards.
Thanks.
Hi,
I am not sure, if its a kamailio issue or something else, but our Kamailio 4.2 server crashes everyday. Throwing out these logs: http://pastebin.com/z1MWdk7q. We have to restart the server to get this fixed. Please help us understand what is causing the issue.
Thanks and Regards,
Sharon Nathaniel
The information in this email is confidential and may be legally privileged. It is intended solely for the addressee and access to it by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken based on it, is strictly prohibited and may be unlawful.
Hello everyone. I'm a student and I'm working on project which about SIP communication and PBXs with IP phones.
So, my first test results were OK with JITSI. But I think I need Freeswitch for working with IP phones. In a word, I need configure an integration Kamailio and Freeswitch.
First, I setup Debian 8.3 Jessie and installed Kamailio on this my local server, after that I installed another Debian 8.3 Jessie (from same image) as my second server and it was going to run Freeswitch on itself.
Then I tried to send my communication requests to my Freeswitch server. I made some tests, my Jitsi accounts which are set in Kamailio server were done. I've read Kamailio 3.2 and Freeswitch integration from the Kamailio wiki and I applied exactly the same thing but I did not work.
When I tried to reach another IP phone from my IP phone (I registered them) phone always say "NOT FOUND". I'm sure I did my configuration properly.
So, how can I figure it out these things? How can I understand, does Kamailio routes to Freeswitch to my phones?
Best regards.
Thanks.
