Hi Diego,
I forgot to tell that opening large number of tcp sockets means opening large
amount of file descriptors.
echo 128000 > /proc/sys/fs/inode-max
echo 64000 > /proc/sys/fs/file-max
ulimit -n 64000
// Binan
Från: Diego Barberio
Till: Binan AL Hala
The sip_trace called only one, but opensips make two records.
Here is log:
Oct 12 00:34:47 routecall /usr/local/sbin/opensips[2191]: Trace ON.
Trace_id=8
Oct 12 00:34:47 routecall /usr/local/sbin/opensips[2191]:
DBG:core:parse_headers: flags=40
Oct 12 00:34:47 routecall /usr/local/sbin/opensips[2
Hi,
Not to get personal here but I think one should not get way too fancy with
the fonts and colors while posting on mailing lists. Atleast it made me
feel that I'm in elementary school receiving email from an excited class
mate.
Coming to the solution:
http://www.codinghorror.com/blog/20
It won't do anything if you used engage_media_proxy(), see
http://www.opensips.org/html/docs/modules/1.6.x/mediaproxy.html#id250230
2012/10/11 Binan AL Halabi :
> Hi,
>
> # BYE processing
> if (method==BYE) {
> end_media_session();
> }
>
> // Binan
>
>
> F
One of my clients is sending a buggy SIP invite which contains Privacy
headers multiple times.
Supported: replaces, timer
Privacy: id
Privacy: id
Privacy: id
Content-Type: application/sdp
Content-Length: 298
How can I strip out the last 2 Privacy headers and keep only the first
before sending i
I hope before writing this question you must have done some brain storming,
did you? or not?. Anyways considering you have basic programming skill, and
there is no OpenSIPs all you have to do is create an algorithm; fairly
simple.
An if condition, a variable to store, perform checks though
Hello All,
I have User A enregistred on my Opensips Server and i want to routing the
incomming calls for this User to an other phone number if some one call him
between 12h and 14h for example.
Have an idea to do that Please ??
--
Best Regards.
___
User
Hi,
Thank you all
But i thought to do this by hanging up the call of mobile after some time if
the mobile phone does not respond
Is opensips can do that?
thank you for your help
2012/10/9 Schneur Rosenberg
> That's true but from my experience it worked for me with a few carriers,
> of course ev
http://opensips-open-sip-server.1449251.n2.nabble.com/Python-module-on-OpenSIPS-td5018382.html#a7581444
On Thu, Oct 11, 2012 at 1:57 AM, qasimak...@gmail.com
wrote:
> Hi,
>
> Just wondering why the following link is not available in documentation.
>
> http://www.opensips.org/html/docs/modules/dev
Did you try to add logs to make sure sip_trace() is not called twice?
Regards,
Ali Pey
On Wed, Oct 10, 2012 at 10:29 AM, Dragomir Haralambiev
wrote:
> Hi,
>
> Thanks for your replay.
> The problem is not in IF operator.
>
> When use sip_trace() Opnesips make two records in sip_trace.
>
> Best re
SOLVED!!
It was because i have an error on dialog behaviour. For now I solved in this
way, hope could be interesting for someone else:
*branch_route[2] {
if (is_method("INVITE") && is_audio_on_hold()){
if ( search_body("a=sendonly")){
set_dlg_
Hi ,
i need to register the time when arrive the following messages: invite,
alerting, progress , connected and hangup.
How can I make it ? it's is possible to store these value in the extra field
in the acc module ?
Thanks
___
Users mailing list
Hi Vlad. Ok now it's a bit clear.
What i am trying to do is to solve the following issue:
PBX<--->Opensips<>Lync server
Opensips acts as UDP/TCP proxy. Call can go from PBX to LYNC and viceversa.
The problem comes when from LYNC I put on hold the call. Everytime i get the
following error, ju
Hi,
# BYE processing
if (method==BYE) {
end_media_session();
}
// Binan
Från: ㄨ冷se灬 <291490...@qq.com>
Till: users
Skickat: torsdag, 11 oktober 2012 11:14
Ämne: [OpenSIPS-Users] opensips use engage_media_proxy(), cannot
end_media_session() problem
Hello,
The validate_dialog() and fix_route_dialog() functions are not currently
allowed in branch_route.
But what are you trying to accomplish, more exactly ? The validation and
fixing functions are meant to work on sequential requests, where you
know the dialog end-points, so there is no ne
hi:
my opensips is 1.6.4-2mediaproxy is 2.4.4
now when call from PSTN to my opensips ,it works normal.
but call from my opensips to PSTN ,it has a problem:
opensips receive an invite message ,it use engage_media_proxy and the
call works normal ,my media-relay has lo
Hi Vlad, thanks for reply.
I am trying to follow your hint but I get this error when I start Opensips:
*Oct 11 10:29:45 opensips opensips: DBG:core:find_cmd_export_t:
not found
Oct 11 10:29:45 opensips opensips: DBG:core:find_cmd_export_t: found
(0) in module uri
[/usr/local/opensips_proxy/lib/op
Hi,
I think that there is a memory leak with TLS.
I have experienced it in my calls stress test, but it is possible to reproduce
it easily with sipp with a simple scenario:
you have only to send REGISTERs in multi socket mode (-t ln).
With this test I can observe the internal memory that sometimes
Hello,
Yes, if you want to use multiple flags you can just pass them
concatenated in the second do_routing() param.
Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com
On 10/09/2012 11:33 AM, JoeAkiki wrote:
Hello,
I just want to ask about the flags in do_routing: If I
Hello,
Just as the logs says, the remote Contact is not valid according to SIP
dialog point of view. At dialog establishment the Contact URI was
'sip:5100@172.16.52.51;transport=UDP;user=phone ' while the sequential
SIP message contained 'sip:5100@172.16.52.51;user=phone'.
Notice the removal
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