Hi Om,
finally, after some struggle with the libmysqlclient, I hope the issue
is solve. Please give it another try and let me know the result.
Thanks and regards,
Bogdan
Om Bikram Thapa wrote:
> Hi Bogdan,
>
> Just updated to latest svn, but I still see crashes after MySQL
> restart. The syslog
Hi John,
Thanks for the info - it was a valuable one. I was fighting with some
strange behaviour between different versions of libmysqlclient in
reporting the error codes.
Anyhow, a new (finally I hope) fix is available on SVN. Please update
and let me know if it works for you or not.
Regards
Forgot to include the list before
On Feb 26, 2009, at 2:42 PM, Yehavi Bourvine wrote:
2009/2/26 Zahid Mehmood
i tested directed pickup and it worked fine in pure sip
environment.. the only issues i had were with the cisco media
gateway not working properly with REFER etc. Basically
Il giorno 27/feb/09, alle ore 15:01, Thomas Gelf ha scritto:
> You can find this explanation also in the nat_traversal documentation:
Thank you all,
I have read "with attention" the nat traversal section :-)
Now I have set the keepalive_method to "OPTIONS".
Regards,
Carlo Dimaggio
_
Sorry for the delay. I did not read your reply.
Nice solution. I need to check that IOS based pbx can read a custom header.
Regards.
Antonio.
> -Original Message-
> From: users-boun...@lists.opensips.org [mailto:users-
> boun...@lists.opensips.org] On Behalf Of IƱaki Baz Castillo
> Sent:
There is nothing wrong, most phones just don't "understand" the "keep
alive" event in your NOTIFY request - but that's fine, as you don't
need a positive answer for the "keep-alive-effect", you just need a
packet keeping the hole open.
You can find this explanation also in the nat_traversal docume
Hi Carlo,
I think the problem is with the phone as they do not understand the
"keep-alive" event sent in NOTIFY.
It is nothing you can do about this (in is not on the proxy side). Maybe
in the future we can add a module parameter for removing the Event
header from the REQUEST and make the dumm
On Friday 27 February 2009, Carlo Dimaggio wrote:
> Hi all,
>
> I'm using nat_traversal and the nat_keepalive function. For phones-
> compatibility I have to use the "NOTIFY" method but all my registered
> phones reply to opensips with "Bad Request / Bad Event" (with OPTIONS
> method some phones re
Hi all,
I'm using nat_traversal and the nat_keepalive function. For phones-
compatibility I have to use the "NOTIFY" method but all my registered
phones reply to opensips with "Bad Request / Bad Event" (with OPTIONS
method some phones reply with "200 OK"). What I am wrong?
Can you help me?
Thanks Bogdan...I'll check this and get back to you
Regards
On Fri, Feb 27, 2009 at 1:34 PM, Bogdan-Andrei Iancu wrote:
> Hi michel,
>
> Should do something like:
>
> if(!cr_route("default", "0", "$rU", "$rU", "call_id")){
> sl_send_reply("403", "Not allowed");
> } else {
> # In cas
Hi Visu,
I tend to think that there is something wrong in your configuration. Can
you please send me privately a message trace( with ngrep or wireshark)
and the corresponding log for exactly the same scenario that your log
captured - the Subscribe to the list and until the Subscribe that gets
Hi Robert,
SST is available only in trunk Asterisk.
What you can try is to use SST from OpenSIPS - see the SST module -
http://www.opensips.org/html/docs/modules/1.4.x/sst.html
Regards,
Bogdan
Robert Borz wrote:
> I want to use sip session timers to ensure client sip-sessions are really
>
Hi michel,
Should do something like:
if(!cr_route("default", "0", "$rU", "$rU", "call_id")){
sl_send_reply("403", "Not allowed");
} else {
# In cas of failure, re-route the request
t_on_failure("1");
t_on_reply("1");
force_rtp_proxy();
t_relay
I want to use sip session timers to ensure client sip-sessions are really
active for billing purposes. We use Asterisk version 1.4 for pstn connectivity
which does not support SSTs.
So I want to use SSTs only on the client-side of a dialog. Would this be
possible? Has anybody a setup like this?
Dear Bogdan,
Do you mean doing something like that?
if(!cr_route("default", "0", "$rU", "$rU", "call_id")){
sl_send_reply("403", "Not allowed");
} else {
# In cas of failure, re-route the request
t_on_failure("1");
force_rtp_proxy();
t_relay();
route(2) ;
Hi Romanov,
not sure about IMS bench, but you could try with SIPP to generate
Presence traffic.
You simulate in a first step the PUBLISh traffic and to upload presence
info, to see how the server reacts. In a second step, you can generate
SUBSCRIBE .
Regards,
Bogdan
Romanov Vladimir wrote:
>
Hi Michel,
You have to call twice force_rtp_proxy() in order to complete the RTP
session (and have RTP flowing). First for the INVITE (as you already do)
and second for the 200 OK reply.
So install a onreply_route and if the reply is 200 OK, call again
force_rtp_proxy.
Regards,
Bogdan
michel
Hi All!
How I can test performance of presence server in OpenSips? We use IMS bench to
test register, uac, message.
--
Vladimir Romanov
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