Hello guys,
I've got a kamailio on a docker and kamailio on another docker on different
boxes.
The issue i'm having is that when kamailio tries to enable rtpengine it
tries pinging it.
The ping arrives at the actual container where rtpengine is running, but
for some reason i haven't figured out,
Hi,
There are two funny things, which I do not understand:
1. The first (orig) S-CSCF should already replace the TEL URI by a SIP URI
and do an ENUM query:
Lines 1140 – 1158 of Kamailio_scscf.cfg:
route[PSTN_handling]
{
# First, we translate "tel:"-URI's to SIP-URI's:
# $ru:
Id love to help, but I didn’t understand the problem:
- you had configured your kamailio with IP 1.2.3.4 and ALIAS sip.fqdn.com
- DNS on sip.fqdn.com returned 1.2.3.4
All good.
Then the DNS pointer changed 1.2.3.4 to 4.3.2.1.
Is that the question?
If so, I assume the IP where kamailio is
Interesting. Thanks for the tidbit. I will play around with
msg_apply_changes and find the right place for it.
On Fri, Jun 26, 2020 at 9:01 AM Richard Fuchs wrote:
> Rtpengine takes and replaces the entire SDP body, so if there's another
> module also manipulating the SDP, one module won't
Thanks Richard.
I do have the nathelper module running but not rtpproxy. How does
nathelper cause this issue?
On Fri, Jun 26, 2020 at 8:27 AM Richard Fuchs wrote:
> Looks like you're using both rtpengine and some other SDP-modifying module
> together, such as nathelper or rtpproxy, without
in kamctlrc i defined an DOMAIN but later that ip was updated..
so my question: it make sense inclusivelly if already defined more
allias in kamailiorc.cfg file?
in case of more complicated response,, anybody can explain me about that why?
--
Lenz McKAY Gerardo (PICCORO)
Hello,
Unless I'm mistaken, there is no "out-of-the-box" solution.
A way to do it would be an active/sdandby with keepalived + virtual IP + HA
Redis setup.
With Kubernetes, it could be possible, but I'm not sure it's well suited for a
wide port range (usually 1 UDP ports), but I'm
Hi List,
I'm having trouble adding the parameter user=phone in the RURI. I have
tried to use the function add_uri_param from siputils module but it doesn't
work (see below). Here is the code I use to add user=phone parameter to
FROM, TO and RURI
uac_replace_from("",
RTPEngine has Redis-based RTP flow state sharing you may wish to look into.
—
Sent from mobile, with due apologies for brevity and errors.
> On Jun 26, 2020, at 9:39 AM, Evgeniy wrote:
>
> Is that possible to implement RTP voice fail-over?
>
> In case RTP node goes down - the call should not
Hello,
Just wanted to update what fixed our issue.
The "contact_user" which is "asterisk" that we were using for the Kamailio
context for outgoing SIP trunk/channel in the pjsip.conf file of the Asterisk
16 server was not a valid user in the IMS HSS database (FHoSS implementation).
However not
Hi Mojtaba,
not looked in the code – but it could be related to be able to check with this
function for request and replies. Replies will have the method name in their
CSEQ.
If you have a requirement that is not available right now in the module, it can
be of course extended (e.g. by pull
I would agree about the marginal utility.
—
Sent from mobile, with due apologies for brevity and errors.
> On Jun 26, 2020, at 11:00 AM, Sergiu Pojoga wrote:
>
>
> The most thorough presentation about the topic that I know of was 1&1's where
> BGP is used.
> The task of detecting such a
Hi Evgeniy,
you can for example save rtpengine sessions informations in redis and do an
active / passive setup for media nodes.
Rtpengine is capable to fetch up the session on startup from redis if you
point it to the same redis dB
Just an idea.
Cheers
Karsten
Evgeniy schrieb am Fr., 26.
The most thorough presentation about the topic that I know of was 1&1's
where BGP is used.
The task of detecting such a failure within a couple of seconds so that
user's won't hang up intuitively, avoid the failse-positive failure
detection, makes the entire task pretty daunting and marginally
Is that possible to implement RTP voice fail-over?
In case RTP node goes down - the call should not be interrupted.
This is kinda strange - but client want to have super redundant system.
For example use 2 RTP nodes and simultaneously send 2 RTP stream over 2
nodes (yes - network traffic will
Rtpengine takes and replaces the entire SDP body, so if there's another
module also manipulating the SDP, one module won't see the changes made
by the other one unless you call msg_apply_changes in between, leaving
you with bits of string in the wrong places.
Cheers
On 26/06/2020 08.50,
Looks like you're using both rtpengine and some other SDP-modifying
module together, such as nathelper or rtpproxy, without calling
msg_apply_changes() in between.
Cheers
On 25/06/2020 16.46, Andrew Chen wrote:
Hi forum,
I'm starting my rtpengine project and I'm facing a strange problem
Any idea?
On Wed, 24 Jun 2020, 14:26 Mojtaba, wrote:
> Hello,
> Some confusing things while developing in Kamailio:
> In is_method_f function in textops module, the code checks the HDR_CSEQ_F
> in msg also, and if the method name appears in CSEQ_F, The result is
> returned TRUE.
> These
18 matches
Mail list logo
