Hey all,
I'm looking to put Kamailio behind a TCP load balancer that is SIP-unaware.
My application is deployed in AWS and I'm tying to place Kamailio behind an
ELB.
For the most part, everything is fine. For my specific implementation I'm
disabling UDP as a signaling transport and using only TLS. This enables me
to not have to worry about a SIP-aware LB at the edge because replies to an
incoming request will be sent over the existing established TCP socket
(avoiding any crazy routing requirements).
However - this poses an issue with source addresses. Does Kamailio support
anything like the proxy protocol (
http://www.haproxy.org/download/1.5/doc/proxy-protocol.txt) for getting TCP
stream information from a load balancer? Do I need to go back to exposing
it directly to the world so that I can get source addresses?
Thanks,
Colin
Hello,
Kamailio SIP Server v5.1.3 stable release is out.
This is a maintenance release of the latest stable branch, 5.1, that
includes fixes since the release of v5.1.2. There is no change to
database schema or configuration language structure that you have to do
on previous installations of v5.1.x. Deployments running previous v5.1.x
versions are strongly recommended to be upgraded to v5.1.3.
For more details about version 5.1.3 (including links and guidelines to
download the tarball or from GIT repository), visit:
* https://www.kamailio.org/w/2018/04/kamailio-v5-1-3-released/
RPM, Debian/Ubuntu packages will be available soon as well.
Many thanks to all contributing and using Kamailio!
Cheers,
Daniel
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Kamailio Advanced Training - www.asipto.com
Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com
Hello all,
I've been trying to figure out a cleaner way to determine the next hop for
a SIP message, mainly for use within the NATMANAGE route for multi-homed
kamailio instances (with three or more network interfaces on the kamailio
host).
So far I have achieved this with a series of nested ifs, depending on
whether the message is a request or a response and by calculating the next
hop based on the various headers (R-URI, Route) and variables ($T_req($Ri),
$dd, ) involved in SIP routing.
A simpler way to do it, of course, would be to use the onsend_route, but
that would most likely introduce an unnecessary overhead for all routed
messages.
I recently noticed there a pseudovariable called $nh(key), and I believe I
can use $nh(d) to the same effect. I understand however, that this works
for requests only. Also, the description of this PV in the documentation
reads as follows:
Return attributes of next hop for the SIP request. Address is taken from
> dst_uri, if set, if not from new r-uri or original r-uri.
>
What is not clear to me is if this covers in-dialog requests with the Route
header set as well. Does the inclusion of a route header set the dst_uri
PV? And if yes, is it safe to rely on dst_uri during request processing or
is it set only after completion of script processing?
Lastly, is there an analogue for SIP responses?
If not, is it safe to rely on first Via header to determine next hop for
responses, or are there any other corner cases I need to heed?
Thanks in advance,
BR,
George
Hello
I have a question to those of You who uses push via Google or Apple to
initiate calls.
How long time does it usually take, before the phone gets the push and
wakes up?
Our experience is, that it can take from 2 to 10 seconds, and that not
impressing when we are talking about telephony.
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Hi Team,
I have setup IM server with kamailio using MSILO module.
Its working fine now.
I have a scenario where lot of offline messages are being stored in mysql
database for a user B. As soon as destination (B) REGISTER with kamailio,
these messages will deliver to him. Now, kamailio will deliver all of these
messages that will cause user (B) device (softphone on mobile) stuck or
freezs for the moment messages are being delivered like bombardment from
kamailio.
Is there any way, where I can define batch size of messages to be delivered
after provided interval?
like batch_size=10 messages and batch_interval= 5sec, etc.
I want to avoid applying static limit to mysql query in module source code.
Please advise.
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Hi all,
Just wanted to know what your opinions were on using DMQ modules over
database for things like dialog replication, registrations, etc...
Is DMQ the "new way to go"? I know that there lots of ways of doing things
with each having pros/cons... But I was wondering...
What does the community think on this topic?
Are you guys taking advantage of the DMQ modules or are you still relying
on database as much as possible? Maybe a combination of both?
Cheers,
Joel.
Phone sends Subscribe
Kamailio Responds 202 OK
Kamailio Responds with NOTIFY
Via: SIP/2.0/TCP 66.66.66.66:5051;branch=z9hG4bKfe3e.e4fedb84000000000000000000000000.0
To: <sip:30041@66.66.66.66 <mailto:30041@66.66.66.66>>;tag=48ae253ab7
From: <sip:47701@66.66.66.66 <mailto:47701@66.66.66.66>>;tag=00c39aabf419ddeab4250cd65af8ef90-6fb2
CSeq: 2 NOTIFY
Call-ID: 1493ef52c41b6c1a
Content-Length: 0
User-Agent: kamailio (5.1.2 (x86_64/linux))
Max-Forwards: 70
Event: dialog
Contact: <sip:192.168.1.30;transport=udp>
Subscription-State: terminated;reason=timeout
In that packet I see Subscription-state: terminated;reason=timeout
Im not sure what I am missing or misconfigured. Any help would be great. I’ve attached my config.
Thanks.
Hi guys,
I'm trying to setup a Call Forward No Reply with Kamailio and I've run into
this problem (that was expected):
When A INVITEs B, B replies with 183 and some SDP.
B doesn't answer so Kamailio triggers the timeout and sends the INVITE TO C.
C replies with 200 OK and some SDP, but A side drops the call as the media
session doesn't match with the expected one.
To solve it, I'm trying to send to A an UPDATE before relaying the 200 OK
from C. I tried to use UAC module (uac_req_send + $uac_req) but I don't see
how I can UPDATE the existing dialog. uac_req_send seems to try to setup a
new dialog always.
Is there a way to indicate to uac_req_send that I wan't to use the existing
dialog? Or maybe there is another mechanism I can use to send the UPDATE
inside the dialog?
Regards,
Alfonso.
Hi All,
I am configuring Kamailio version 4.4 with jsonrpc-s and xhttp module. My
aim is to use Kamailio commands using RPC from a remote server. I need help
to solve error. After send rpc command, I got execution error in response.
I am attaching code in kamailio.cfg file for jsonrp.
event_route[xhttp:request] {
if(src_ip!=127.0.0.1){
xhttp_reply("403", "Forbidden", "text/html",
"<html><body>Not allowed from $si</body></html>"
);
exit;
}
if ($hu =~ "^/RPC"){
xlog("json rpc dispatch");
# xlog($jsonrpl(method));
jsonrpc_exec('{"jsonrpc": "2.0", "method":
"dispatcher.reload", "id": 1}');
jsonrpc_dispatch();
# xlog($jsonrpl(code));
# xlog($jsonrpl(text));
# xlog($jsonrpl(body));
xhttp_reply("200", "OK", "text/html", "<html><body>
$jsonrpl(text), $jsonrpl(body)</body></html>");
} else {
xhttp_reply("200", "OK", "text/html",
"<html><body>Wrong URL $hu</body></html>");
}
return;
}
Also, I am attaching logs. which could help you.
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Hello All,
Unfortunately we found two more bugs in the nathelper module in my favorite Kamailio! Oh my god!
1. nathelper doesn't send pings when db_mode=3 and db is mongodb. I will provide fix soon. The root cause is incorrect date format in get_all_db_ucontacts function in usrloc.
We need to use datatime type instead string.
2. nathelper doesn't send pings if ping_nated_only=0 (default value). It looks like incorrect condition in nh_timer in nathelper.
Definitely lack of testing. It is interesting how testing is organized in the Kamailio community?
Best regards.
Konstantin
ping_nated_only
