Hello,
My opensips server is just a registrar server and I have enabled tls with
the following settings:
listen=tls:xx.xx.xx.xx:5061
disable_tls=no
tls_certificate="/etc/opensips/pbx-bundle.crt"
tls_private_key="/etc/opensips/pbx.key"
When my sip phones try to open tls connection, they reject
Hi Guys,
Using opensips 2.1 and when I use uac_replace_from to do the following on an
initial invite;
uac_replace_from("anonymous","sip:anonymous@anonymous.invalid");
This is on anonymous calls, where I engage this only when PAID and Privacy:ID
are present.
Once the Call establishes, for some
Hi all,
I am using following setup
OpenSIPS 2.2 devel with wss support.
sipjs 0.7 as websocket client
RTPEngine 4.4
I have two web users like 1000 and 2000. Both are registered to OpenSIPS
through the WSS. Whenever i made call between the 1000 to 2000 or vice
versa, i am getting error saying "**
Ok. Thank you.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org em nome
de Johan De Clercq
Enviado: quinta-feira, 7 de abril de
Thank you so much for the link .
One more question .
If the codec format is g729 then how I will run the sox command ? Is that
will take g729 in place of u-law ?
*Thanks & Regards*
*Sasmita Panda*
*Network Testing and Software Engineer*
*3CLogic , ph:07827611765*
On Thu, Apr 7, 2016 at 3:43
Hi,
I experienced something weird: I got two servers sharing the same
location table. (usrloc module)
My problem is: both servers got nathelper ping (OPTIONS) enabled. The
Opensips instance A has it's own clients and instance B also...
Now as the location database/table is being shared by
https://voipembedded.wordpress.com/2011/11/15/extracting-audio-from-calls-recorded-with-rtpproxy/
On Apr 7, 2016 1:17 PM, "Sasmita Panda" wrote:
> Hi All ,
>
> I am trying to do recording on rtpproxy . I am using opensips-1.11
> versin and rtpproxy-1.2 version .
>
>
Hi All ,
I am trying to do recording on rtpproxy . I am using opensips-1.11
versin and rtpproxy-1.2 version .
I have added rtpproxy_start_recording() in request route and and
on_reply route in opensips sonfig file .
I am running rtp proxy with following command .
rtpproxy -l
Wild guess here, based on what I've read..
It might be your request URI is explicitly declaring which protocol to use,
and OpenSIPS has no interface configured for that protocol.. eg.
;transport=tls is added to the request URI but there's no listen = tls:
1.2.3.4:5061
I know it's tricky to look