Hi All,
I would setup a configuration where Kamailio authenticate asterisk SIP trunk
using TLS and SRTP.
At moment I was able to configure everything, including RTTProxy since most of
the asterisks v1.8.19.1
are behind NAT. So far so good it works pretty good using standard
authentication and
if(defined $var(x))
Or, if checking for empty value:
if(strempty($var(x))
Mino Haluz mino.ha...@gmail.com wrote:
Hi,
how should I check if the value is set?
if ($avp(s:test) == ) {
or is there any null keyword ? If so, does it work for $avp, $sht, $var
and
$shv ?
Thanks,
Mino
and how to check null value taken from database?
$dbr(ra=[0,0]) == ?
On Mon, Jan 14, 2013 at 1:44 PM, Alex Balashov abalas...@evaristesys.comwrote:
if(defined $var(x))
Or, if checking for empty value:
if(strempty($var(x))
Mino Haluz mino.ha...@gmail.com wrote:
Hi,
how should I
On 01/14/2013 09:16 AM, Mino Haluz wrote:
and how to check null value taken from database?
$dbr(ra=[0,0]) == ?
That depends on what you truly mean by null value. :-)
If you want to check for no rows returned, you can check the value of
$dbr(ra=rows) to get a count, e.g.
The caller should use the NATPR and thus should use TLS. The SIPS+D2T
does not requires the URI to be a SIPS URI.
See also the thread
NAPTR, SRV and sips vs. transport=tls from 1.Dec.2012
regards
Klaus
On 11.01.2013 18:45, Daniel Pocock wrote:
I'm just wondering if anyone can comment on
Seems like there is a problem again with spammers.
On 11.01.2013 23:25, Daniel-Constantin Mierla wrote:
Hello,
fyi, I upgraded dokuwiki to latest version, switching to the new default
template (which is not bad at all, btw) as the old one is not yet fully
compatible. If you find any issue,
First, you should test TLS with RTP (first make sure that TLS works,
then enable SRTP).
Seconds, it seems like an Asterisk problem, thus may get better answers
on the Asterisk mailing lists.
regards
Klaus
On 14.01.2013 11:23, Roberto Fichera wrote:
Hi All,
I would setup a configuration
On 14/01/13 15:59, Klaus Darilion wrote:
The caller should use the NATPR and thus should use TLS. The SIPS+D2T
does not requires the URI to be a SIPS URI.
That was my understanding too - do you feel it is always working this
way in practice though with the major SIP proxies/PBXes? Or are any
