Hi.
It is normal in this case. I have same scenario and it work normal
with this warning.
I think it because after conversion first content type (in SIP
headers) is "Content-Type: multipart/mixed;" and "Content-Type:
application/sdp" is inside multipart body.
--
Best regards,
Sergey Basov
I'm not sure why you would immediately jump to the conclusion that you
need to rewrite the To header.
This is just a normal 407 challenge. The first and most straightforward
interpretation is that the provider expects you to provide digest
authentication credentials, or that they don't have your
Hi,
after Kamailio did a correct (INVITE sip:089321608@5.9.87.18 SIP/2.0 looks
good to me), to our Sip-Provider we get:
"SIP/2.0 407 Proxy Authentication Required."
Could it be that i have to rewrite also the following line:
To: .
To: .
and how can i do that?
Until now i use just ds_select_doma
Santosh,
Looks like you didn't include the mohdir information (see
http://kamailio.org/docs/modules/4.4.x/modules/mohqueue.html#dir.parms) or
there are no files in that directory.
BTW, I rarely use this email address so please address your questions to
sr-users@lists.kamailio.org.
Hope this help
Hi list,
I'm converting body message to multipart. Calls establishes ok and Kamailio
it's not logging errors but I'm seeing some info and warnings that I don't
know it they are correct.
Maybe you could tell me if it's normal or if I'm doing something wrong in
my script.
kamailio[13428]: INFO:
Hello Daniel,
Thank you for reply.
I haven't used event-route in this scenario.
When i fork parallel (without suspend/continue), memory leak does not
happen.
On Mon, Apr 24, 2017 at 5:44 PM, Daniel-Constantin Mierla wrote:
> Hello,
>
> there seem to be quite a lot of avps -- are you using an ev
Hello,
there seem to be quite a lot of avps -- are you using an event_route
block with avp variables inside?
Cheers,
Daniel
On 24.04.17 14:34, Julia Boudniatsky wrote:
> Hello all,
>
> I need to fork an incoming call to the second destination, after
> receiving 180 reply from the first destinat
Nice work Guillaume!
On 24 April 2017 at 15:21, Siwek, Cezary wrote:
> Nice work Guillaume!
>
> On 21 April 2017 at 09:55, Daniel-Constantin Mierla
> wrote:
>
>> Hello,
>>
>> I would like to announce that Guillaume Bour (https://github.com/gbour)
>> has now developer privileges on Kamailio's gi
Hi all,
I'm trying to transmit a CANCEL sip message from one server to multiple other
servers (using ds_select_domain and ds_next_domain). Ideally, I would simply
use forward() multiple times. The problem is however, that this would apply the
changes of forward() multiple times too, for example
On 04/23/2017 11:40 PM, Alberto Llamas wrote:
Hi guys,
Any of you have experience using the '*start_recording()*' function of
the RTPENGINE in a WebRTC scenario?.
At this point, I am able to record the stream but I am not able to
play them. The output is a .pcap file in a raw format. I have
Alex,
So far they have: IP header and UDP headernot RTP header.
Thanks !
On Mon, Apr 24, 2017 at 12:06 AM, Alex Balashov
wrote:
> Alberto,
>
> But does it have IP, UDP or RTP headers?
>
> On April 23, 2017 11:57:00 PM EDT, Alberto Llamas <
> albertollam...@gmail.com> wrote:
> >Hi Alex,
> >
Hello all,
I need to fork an incoming call to the second destination, after receiving
180 reply from the first destination.
I use kamailio version 4.4.5
It working fine, but with shared memory leak.
The script part:
$avp(00x)=$_s(sip:$var(00x)@$si:$sp;user=phone);
if(!is_known_dlg()
Hi, Daniel.
I got it worked for kamailio 4.4.5, there are only changes in path for
includes. All other is work for me.
--
Best regards,
Sergey Basov e-mail: sergey.v.ba...@gmail.com
2017-04-24 14:37 GMT+03:00 Daniel-Constantin Mierla :
> Hello,
>
> thanks for troubleshooting,
Hello,
thanks for troubleshooting, testing and reporting the results, the
feedback helped a lot to solve these issues.
I will look at the patch and see if should be pushed to 4.4 -- iirc,
there were quite some changes in topos internals that may be too heavy
for backporting.
Cheers,
Daniel
On
Hello,
there are a few bits that are not going to work, respectively the MI
commands (in the control panel and iirc, some reload commands), but most
of them are just going work ok.
Hopefully there will be some spare time to upgrade it to use jsonrpc
instead of the MI commands.
Cheers,
Daniel
O
Not getting invite on other client
04-24 03:53:46.984 4191-18519/com.gsma.rcs
I/[RCS][MultimediaSessionServiceImpl]: Add an event listener
04-24 03:53:46.984 4191-32546/com.gsma.rcs
I/[RCS][MultimediaSessionServiceImpl]: Initiate a multimedia messaging
session with +19253248960
04-24 03:53:46.984 4
Hello,
I have installed kamailio 5.1.0-dev and would like to know if it is
compatible with SIREMIS v4.3.0?
Has anyone tried it?
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sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr
Hi, Daniel.
Thank you for your time and patches!
I just finished load test and after 25000 calls there is no memory leak now.
I have done backport of topos module to kamailio 4.4.5, test it and
result is success.
I does not very good in using github, so I attached source folder of
topos module f
Hello,
it looks like you have two t_relay() executions for the same request.
Can you double check the configuration file?
Cheers,
Daniel
On 21.04.17 17:19, Slava Bendersky wrote:
> Hello Everyone,
> Based on trace message generated by t_relay();
>
> Apr 21 00:57:37 cavprx00 kamailio: 2(4104) DE
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