Hi,
I am new to Kamailio and coming a bit from the Asterisk side.
My goal is to set up a bunch of users on a Kamailio Instance on AWS
EC2 and get them able to call each other and talk to each other over
the Internet without P2P media stream.
Is rtpproxy the tool that enables that? As far my
Juha Heinanen writes:
> While testing xflags, i noticed that a regular flag that I set AFTER
> calling t_newtrans() stays set in onreply_route even when I do not
> call t_flush_flags().
I made the same test with xflags and they do require t_flush_xflags()
call if an xflag is set after
this is a big question.so, i suggest you read some guide first.
2018-03-26 15:18 GMT+08:00 星昊通技术 :
> hello all:
> I install kamailio 5.1 and siremis 5.1, I want to configure kamailio
> connection freepbx, how should i configure kamailio vai siremis GUI?
>
> Thanks.
>
>
Hi thx. Yep. NAT handling was moved to branch.
So rport called from this handler and never called fpr the 1-st invite.
On Mon, Mar 26, 2018, 16:05 Aqs Younas wrote:
> Make sure you are calling force_rport.
>
> Br, Aqs.
>
>
> On 26 March 2018 at 13:57, Yuriy Gorlichenko
Make sure you are calling force_rport.
Br, Aqs.
On 26 March 2018 at 13:57, Yuriy Gorlichenko wrote:
> Hi
> I'm using version: kamailio 5.1.2
>
> I have strange issue with auth_challenge
>
> kamialio works as NAT handler. When Register comes kamailio handles nat
> and save
Hi
I'm using version: kamailio 5.1.2
I have strange issue with auth_challenge
kamialio works as NAT handler. When Register comes kamailio handles nat and
save original ip:port at the received field in location table
I found that when local port in contact is different than external port
reply
hello all:
I install kamailio 5.1 and siremis 5.1, I want to configure kamailio connection
freepbx, how should i configure kamailio vai siremis GUI?
Thanks.
--
Best regards!
alex
Hiastar ??
Sangoma in APAC, IP??IPPBX
Asterisk cards, SBC, NetBorder VOIP
HI,
Myself Ayub from Phonology, Bangalore, India.
We are trying to set up an SIP proxy server using Kamailio with Siremis as
web interface.
Have installed and configured kamailio and siremis but need help in config
if sip domain , users and etc...
Please revert.
Regards
Ayub
Hey Ayub,
I would take it one step at a time. Kamailio is a toolkit with a bunch of
modules. I find it useful to first figure out what features you need in a
proxy. This includes understanding what are you trying to protect your
backend media servers from, problems you are currently having
Hi. New to the area of Kamailio.
i am did install in debian kamailio with rtpproxy and i have created 3
users 1000-10002 (one for each jitsi user) they talk nice between them.
I have followed this tutorial
http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour and
in less than 5
Hi,
This may be more a SIP question than a Kamailio question, but I am
trying to figure out why 407 challenges from Kamailio contain To tags.
I suppose I've not run across it before, and I can't find an RFC-based
rationale. A negative reply to an invite transaction without an
intermediate early
Am Montag, 26. März 2018, 16:55:12 CEST schrieb Alex Balashov:
> This may be more a SIP question than a Kamailio question, but I am
> trying to figure out why 407 challenges from Kamailio contain To tags.
>
> I suppose I've not run across it before, and I can't find an RFC-based
> rationale. A
Hey Atux,
Can you give a little more detail on your use case? Are you looking for
Kamailio to:
- route requests to a media server for playing announcements
- proxy requests between your endpoints and your media server(s)
- distribute calls to your carriers based on some logic
The answer may
On Mon, Mar 26, 2018 at 10:27:35PM +0200, Henning Westerholt wrote:
> I would think this is normal UAC/UAS behavior for SIP request handling:
>
> RFC 3261, sect. 8.2.6.2:
> "However, if the To
> header field in the request did not contain a tag, the URI in the To
> header field in the response
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