i suggest you use tls common names to identify servers behind your
trunks,
-- juha
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Hi, all
I'm trying to set up 2 kamailio servers for active-active redundancy. The
two kamailio severs share the the same database with db_mode=3, and no
registration replication. Use pjsua2 as SIP client for testing. The test
setup is as follows:
kamailio server 1(k1): 10.0.1.30:5061
Have you tried forwarding after the timeout of suspended transaction
occurred? What kind of operations you had in the failure route. Trying
to see what scenarios were covered...
Cheers,
Daniel
On 18/03/15 21:13, Mickael Marrache wrote:
Thanks Daniel!
Your patch seems to have fixed it…
I already looked at your presentation, the PUSH is send synchronously.
What I'm trying to achieve is sending the PUSH asynchronously. So, when I
receive an incoming call and the callee has no registration, I would like to
suspend the transaction and then delegate PUSH sending to another
You can define as many rtimer processes as you want and tell them to
execute same route block, consuming from the same queue.
Cheers,
Daniel
On 19/03/15 09:39, Mickael Marrache wrote:
Thanks, it helps.
However, if I create a timer and specify mode to 1, I will only have
one extra
OK, if you do more advanced scenarios, with forwarding, let me know the
results. If no issues, the fix can be considered for backporting.
Cheers,
Daniel
On 19/03/15 09:06, Mickael Marrache wrote:
No, I didn't try forwarding after transaction timeout because no
voicemail server is configured.
Hello,
look at the sip traffic on server to see if it is forwarded or not, and
if yes, where. You can use ngrep, like:
ngrep -d any -qt -W byline sip port 5060
Also, check your syslog file to see if any error message is printed there.
It is not possible to guess what a fix would be if the real
Hello,
don't do explicit t_suspend() if you are calling the async_task_route()
because it is done internally.
I presented a way for async push notifications during the Kamailio World
Conference 2014, see:
-
Thanks, it helps.
However, if I create a timer and specify mode to 1, I will only have one
extra process to send all PUSH notifications. Therefore, at some point, this
extra process will be continuously busy handling messages in the queue, and
the queue will grow over and over.
Is it
You can delegate to send of the push to a rtimer process using mqueue --
see same presentation, the section about tweeting.
Cheers,
Daniel
On 19/03/15 09:14, Mickael Marrache wrote:
I already looked at your presentation, the PUSH is send synchronously.
What I'm trying to achieve is
Hi,
I'm trying to add PUSH support to my system using Kamailio.
When a call is received and the callee has no active registration, a PUSH is
sent to the callee endpoint. In order to send this PUSH, my proxy sends an
HTTP requests to another server that will send the PUSH.
So, my route
No, I didn't try forwarding after transaction timeout because no voicemail
server is configured.
failure_route[MANAGE_FAILURE] {
route(NATMANAGE);
if (t_is_canceled()) {
exit;
}
if (t_check_status(3[0-9][0-9])) {
Hello,
We have setup kamalio at server and it is working at some DNS with
call/chat.
At some IP chat is working but when we call then it is not working. At some
DNS(Net IP) it is working .
What can be problem at server?
Following is login for SIP test user:
tester1@23.253.110.48
Thanks, it works great!
From: Daniel-Constantin Mierla [mailto:mico...@gmail.com]
Sent: Thursday, March 19, 2015 11:56 AM
To: Mickael Marrache; 'Kamailio (SER) - Users Mailing List'
Subject: Re: [SR-Users] Sending PUSH notification asynchronously
You can define as many rtimer processes as
Hello,
OPTIONS is not the request for initiating the calls, that is INVITE. You
would need to know SIP a bit in order to be able to understand and
configure Kamailio.
If you don't see any INVITE on kamailo server via ngrep when you call,
then the issue is on client side or there is a firewall
Hello,
When I am calling with other SIP user then I did not see any INVITE . that
have issue with DNS.
If we call with different DNS that is working fine then we see INVITE option
like
U 2015/03/19 12:39:01.744616 117.215.244.16:63380 - 23.253.110.48:5060
SIP/2.0 180 Ringing.
CSeq: 2
Please keep in mind that I have no control over SIP trunk providers.
The vast majority do not allow me to do any of these things as far as I
know. This is something that needs to be solved in Kamailio with
standard user/pass/realm authentication. TLS is not an option for me.
On 3/18/2015
Hello Maxim,
Just to let you know that I quick-fixed this issue by increasing the value
compared into ts_less in rtp.c. Instead of the bitwise calculation, I set the
maximum value of an unsigned integer.
I will schedule to test, qualify and replace all my instances with the new
version.
We got following response:
U 2015/03/19 13:00:00.87 23.253.110.48:5060 - 115.252.208.170:62554
SIP/2.0 180 Ringing.
CSeq: 2 INVITE.
Call-ID: 09b36cf7988131e179e345af90922a4c@0:0:0:0:0:0:0:0.
From: tester1 sip:tester1@23.253.110.48;tag=aca2e78b.
To:
canuck15 writes:
The vast majority do not allow me to do any of these things as far as I
know. This is something that needs to be solved in Kamailio with
standard user/pass/realm authentication. TLS is not an option for me.
then the vast majority don't care a bit about security, which is
It looks like auth_check() will work. It seems intelligent enough to
scan all instances of the same domain as long as the username is unique
so that should get things working.
The problem here is that there is a fundamental difference between
Asterisk and Kamailio authentication. Asterisk
These are replies to INVITE requests, but if you see them, the INVITE
passed through the server as well.
If you are not aware of a firewall, then perhaps you don't have one
unless is a default installation with it enabled or one on the network.
I suggest you do sip tracing on the client machine
Hi,
We are currently deploying an entire architecture composed of load
balancers, proxies and media relays. All the components except the media
relays are Kamailio instances. The media relays are RTP proxy instances.
We are trying to determine the hardware requirements for the different
Hello,
the module hasn't be used for long time and was developed when diameter
was a draft (like 2003 -- iirc). Not being demanded, it was no focus on
it and somehow marked as obsolete. I expect to be easy to fix/update if
there is any issue. If you can, give it a try and report the results.
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