Zahid,
Would you mind to share us *secret* option for cisco please?
On 06/01/16 18:56, Zahid Mehmood wrote:
Hi,
INFO is generated in early stage.
The information I provided in my original post was not entirely
accurate. My setup was as follows:
Carrier PRI - Siemens PBX (4000)
When OpenSIPS handles the SIP messages carrying SDP (typically INVITE
request and 200 OK reply), OpenSIPS will communicate with rtpengine and
update the SDP with the new IP and port. Never tested, by AFAIK
rtpengine may know to handle SRTP...not sure, you may check with the
project.
Regards,
http://kb.smartvox.co.uk/opensips/clustering-opensips-part-1/
refer it
Best Regard
On Wed, Dec 30, 2015 at 12:19 PM, Jerry Kendall <
jerry.kend...@bishophosting.com> wrote:
> Did you get it working? Overplayed with opensips but could not get it to
> work. Would you be willing to share your
Hi Stas,
I checked with couple of SIP UACs and I found none using the "DOCTYPE"
line the published presence XML. So, I guess you should simply drop such
a line in your testing.
The "tuple" node is replacing your "atom" node (at least this is what I
noticed while trying other UACs). Here is
Hi Surya,
Even for checking if all the memory was properly release, you still have
to use the Memory debugger - have you managed to get the memory dump on
SIGUSR1 (as per example in the page I pointed)? if so, such a dump
simply contains a list of all allocated memory chunks - size and where
Hi,
I do not contest the correctness of your cfg, but I'm simply asking if
you are 100% sure that your opensips is using the correct opensips.cfg
file ( be sure by explicitly pointing the file via "-f" startup option).
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
Ok. Incase if the media needs to go via rtpengine, then how the signaling
happens, and how the rtpengine is forwarding the media to the other end?
In case of DTLS, only the DTLS handlshake needs to be taken care by the
RtpEngine, and the SRTP is handled by some other media server, and only the
Hi;
Has anyone successfully interfaced OpenSIPS event_xmlrpc with a Tombat xmlrpc
server servlet? If you yes can you share any tips, because I've been trying it
for a while without any success.
Thanks for any help!!
Juls
-Original Message-
From: users-boun...@lists.opensips.org
Hello everyone!
Jan 12-13, the YUM repository (yum.opensips.org) will be down due to
maintenance.
--
Nick
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Serge,
The option is not really a secret, it was just not working as expected
due to out setup. In Cisco documentation it is refferred to as " Buffered
Calling Name Completion
Hi Bogdan,
I didn't do the SIGUSER1 test, instead I checked at time of shutdown. At
the end huge log is generated and I don't know if all of that is unfreed
memory.
You may wish to look at the log http://pastebin.com/M9tmXk5F. This is not
complete log as complete one was very huge and I couldn't
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