8 okt 2012 kl. 16:33 skrev Daniel-Constantin Mierla mico...@gmail.com:
Hello,
some bits were left with the old project name OpenSER, not to disturb too
much at that time and see where everything goes.
Other voices expressed same idea in the past, now everything is stable and
peter suggested using nginx for dispatching xcap requests and that works
fine.
has someone tried using nginx to dispatch also sip over websocket
requests? is there some drawbacks?
-- juha
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
Hello,
I am considering packaging Kamailio v3.3.2 at the beginning of next week
(Monday or Tuesday). If any of the devs have backports to be done, do
them before the end of this week.
Also, if someone spotted some fixes in the master that were not
backported, write to the lists so we can
I've never tested NGiNX with WebSockets, but it should work if NGiNX is
correctly configured. The WebSocket protocol is designed to work
through HTTP proxies.
Regards,
Peter
On Tue, 2012-10-09 at 11:01 +0300, Juha Heinanen wrote:
peter suggested using nginx for dispatching xcap requests and
Hello,
I'm looking at getting richer content for the FAQ wiki page:
* http://www.kamailio.org/wiki/tutorials/faq/main
Probably we can get it good enough with a bit of mobilization for one
week. If anyone is having some questions fitting the purpose of the FAQ,
just add them to the page or
Hi Daniel,
Thanks. But, I think there is any think I have not understand.
I have my route[MCU] module with the following code:
route [MCU]{
if (uri=~^sip:[7,8][0-9]+@){
xlog(Forward to MCU);
rewritehost(10.95.94.142);
rewriteport(5070);
rtpproxy_manage();
t_relay();
Hi folks,
I was making some tests with kamailio (both 3.3.0 and 3.3.1 tgz) with
siptrace module sending homer packets to a remote central agent where the
data is stored and presented to the user.
The related config is:
# - siptrace params from HOMER -
modparam(siptrace, duplicate_uri,
Dear All,
I have set up my kamailio server successfully and it is working very
fine. i have followed the below link for setup:
http://kb.asipto.com/kamailio:presence:k31-made-simple. Now i want to
enable xmpp module. At Kamailio wiki page, i have found that there can
be two way to implement
Did you get a core dump, can you send the backtrace?
Daniel
On 10/9/12 11:00 AM, samuel wrote:
Hi folks,
I was making some tests with kamailio (both 3.3.0 and 3.3.1 tgz) with
siptrace module sending homer packets to a remote central agent
where the data is stored and presented to the user.
I can not locate where the core is (wasn'i it in /tmp?) but I see the
following statement in the log:
Oct 9 12:44:21 kamailio1 kamailio[17420]: ALERT: core [main.c:788]: core
was generated
so i guess somewhere in the disk there is a core file
Thanks a lot, Daniel,
Samuel.
On 9 October
If you provided -w parameter, then it is at that path. If not, it should
be in '/' or in the OS-configured corefiles directory.
Daniel
On 10/9/12 12:45 PM, samuel wrote:
I can not locate where the core is (wasn'i it in /tmp?) but I see the
following statement in the log:
Oct 9 12:44:21
The line matches the access to the action, can you send also the output
inside gdb for command:
p *a
Also, as I understand you can reproduce it, can you load debugger
module, set its parameter cfg_trace to 1 and run again. Then send the
lines printed by the cfg trace before the crash, so I
Hi All
Facing a strange issue in Kamailio / RTP proxy setup.
Using
1. Kamailio - 3.1.5
2. RTP proxy 1.1
I have pne sip client, lets say it is 'A'.
'A' sent INVITE to Kamailio proxy. Kamailio proxy modified the IP address
present in SDP to RTP proxy IP address, and sent to main proxy.
Main proxy
Hello @all,
want to use the remote register function of the UAC module.
We need to register 42 accounts on one openSER server.
At kamailio startup those 42 remote register requests will be startet at the
same time.
This leads to a kind of FlodAttact on the openSER server of our provider, which
Hello,
you added an extra parameter to the function as well, so there should be
other modifications. Better send the patch with your changes to see if
there is something wrong. If you got it from git, run:
git diff /path/to/patchfile
Cheers,
Daniel
On 10/9/12 2:00 PM, Rene Bormann wrote:
Thanks Daniel for you prompt answer.Find
the patch file attached.- ren
Daniel-Constantin Mierla
Dienstag, 9.
Oktober 2012 14:08
Hello,
you added an extra parameter to the function as well, so there
should be other modifications. Better send the patch
You added sleep(1000), which is sleep for 1000 seconds -- that's
quite some time to wait (more than 15min) before sending next
registration.
Also, the extra parameter is not really needed, you can test that
ticks==0, as condition for first iteration.
Hello,
the signaling shows traffic going from public IP (caller) to private IP
(kamailio) then back to public IP (callee). So it is a bit of complex
routing if you keep it like this. The best is to get it done working
with kamailio, rtpproxy and mcu on their public ip, because they have
such
Dear All,
I have set up my kamailio server successfully and it is working very
fine. i have followed the below link for setup:
http://kb.asipto.com/kamailio:presence:k31-made-simple. Now i want to
enable xmpp module. At Kamailio wiki page, i have found that there can
be two way to implement
The output of the requested command is:
(gdb) p *a
Cannot access memory at address 0x485f3832315f4d43
I'll try to load the debugger module, and send the requested information
to digg into the problem.
Thanks again!
SAmuel.
On 9 October 2012 13:25, Daniel-Constantin Mierla mico...@gmail.com
Hi,
I am doing some performance tests and I cannot force SIP INVITEs to
use mediaproxy within LAN, without any NAT. I'm using
engage_media_proxy or rtpproxy_manage but phones are still exchanging
voice data between them, not through the proxy. Is there some trick
how to forcibly tell kamailio to
These are the lines printed, as you can see there isn't yet a rtpproxy
running but it has been working in the tests without crashing. The config
is pretty standard so you'll see the usual routes. I've printed the line
number for an easy debugging.
I'm not 100% sure that it always happens in the
Hi,
If i have two keys on a HTABLE that are the same. i undestand i need to set
the key type to 1 (array).
i use the $shtcn and get 3 (size is an entry as well).
so far, so good.
But, now i am trying to get all the values from the table to an AVP oto say
it in another way - i need all values
Hello,
kamailio cores when receives a corrupted route header.
For example, this was causing the core.
Route:
sip:10.236.236.100;transport=tcp;r2=on;lr;ftag=1348218287134-Test-553188;osb-tag=NM;nat=yes;twan=yes?[=
Hi,
I'm playing around with xavp, but there are some things I can't wrap my
head around.
What basically works is this:
$xavp(a=foo) = 'foo';
$xavp(a[0]=bar) = 'bar';
Getting the value like this:
xlog(L_INFO, a-foo='$xavp(a=foo)' and a-bar='$xavp(a=bar)');
as well as
xlog(L_INFO,
So it is at different points in config, it works if you don't set flag
22, which is used for sip trace.
I see you are using tcp, can you try the latest version of git branch
3.3. There was a fix related to tcp after 3.3.1 and it is better to be
sure it is not actually that case - a guide is
Hello,
On 10/9/12 3:32 PM, Uri Shacked wrote:
Hi,
If i have two keys on a HTABLE that are the same. i undestand i need
to set the key type to 1 (array).
i use the $shtcn and get 3 (size is an entry as well).
so far, so good.
But, now i am trying to get all the values from the table to an AVP
My fault, typoo in dispatcher's IP address.
On Tue, Oct 9, 2012 at 3:25 PM, Mino Haluz mino.ha...@gmail.com wrote:
Hi,
I am doing some performance tests and I cannot force SIP INVITEs to
use mediaproxy within LAN, without any NAT. I'm using
engage_media_proxy or rtpproxy_manage but phones
Hi!
Are there any means to store the route set of a dialog (on dialog
creation) and use the stored route set when forwarding in-dialog
requests instead of using the requests' route set?
thanks
Klaus
___
SIP Express Router (SER) and Kamailio
For messages from the main proxy use the r' flag when activating rtpproxy:
http://kamailio.org/docs/modules/3.3.x/modules/rtpproxy.html#id2548160
regards
klaus
On 09.10.2012 13:58, Kamal Palei wrote:
Hi All
Facing a strange issue in Kamailio / RTP proxy setup.
Using
1. Kamailio - 3.1.5
2.
On 08.10.2012 16:40, Cenk İlker İzanlı wrote:
Hello,
I think it is better to do all changes at once.
+1
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SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
In such cases you should search on the pseudo-variables wiki page for
the relevant key word, in this case branch:
http://www.kamailio.org/wiki/cookbooks/3.3.x/pseudovariables#branch_attributes
regards
Klaus
On 05.10.2012 18:45, Stephen Dodge (Bistech) wrote:
Hello,
Could someone advise if it
Daniel,
I had debug=4 yesterday and I had to delete my logfiles this
morning because the log partition was full. IT seems like when I have
DB ONLY mode turned on for both presence and RLS there is a lot of
logging causing the disk to fill up.
I have my debug level set to 3 and have also
Thanks Klaus, I did look through the cookbook but I couldn't see anything
that would allow me to access the domain directly, I thought there might be a
module that could offer that functionality.
I will look at some find/replace scripting on the $branch(uri).
Thanks,
Stephen Dodge
Tel:
One thing I forgot to include is the handling of the put:
case PUT:
xdbg(= xhttp: put received $var(uri)\n);
xcaps_put($var(uri), $hu, $rb);
if($xcapuri(u=auid)=~pres-rules) {
On 10/9/12 5:45 PM, Stephen Dodge (Bistech) wrote:
Thanks Klaus, I did look through the cookbook but I couldn't see anything
that would allow me to access the domain directly, I thought there might be a
module that could offer that functionality.
I will look at some find/replace scripting
just to complete the thread, nginx does not currently support proxying of
requests that use websocket transport. it is in the roadmap, but no date
is set.
when nginx receives request over websocket it converts
'Upgrade: Connection: Upgrade' header to 'Connection: close' header
and removes
Daniel,
I don't know if this corresponds to the errors I was seeing before.
But on a closer look I see the following errors when a subscribe
request comes in. This is how I am handling subscribe requests:
if( is_method(SUBSCRIBE))
{
xdbg( Handling subscribe
So I fixed a few things today
1. In subscribe.c:rls_get_by_service_uri The code was doing a strlen
check as follows:
if((uri-len==strlen(val)) (strncmp(val, uri-s, uri-len)==0))
The uri coming in is: sip:8475551001@10.50.251.12 and the service
uri in the rls-services list is:
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