On 1/14/13 4:02 PM, Klaus Darilion wrote:
Seems like there is a problem again with spammers.
There was a wiki user created by the spammer with a hotmail email
address. I reverted the content and deleted the user from the system,
let's see if they return.
Cheers,
Daniel
On 11.01.2013 23:25
14 jan 2013 kl. 18:23 skrev Daniel Pocock :
> 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 pr
On 14.01.2013 18:23, Daniel Pocock wrote:
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 w
On 01/14/2013 04:05 PM, Klaus Darilion wrote:
> First, you should test TLS with RTP (first make sure that TLS works, then
> enable SRTP).
I was able to partially fix the TLS problem, now I can do at least
openssl s_client -connect kamailio_ip:5061 -tls1
and get the corresponding answer.
I had
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 an
I believe what you are looking for is $null
if($avp(s:test) == $null){
}
On Mon, Jan 14, 2013 at 3:59 AM, Mino Haluz wrote:
> Hi,
>
> how should I check if the value is set?
>
> if ($avp(s:test) == "") {
>
> or is there any null keyword ? If so, does it work for $avp, $sht, $var
> and $shv ?
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 whe
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, repo
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
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
$dbr(ra=>rows) to get a count, e.g.
if($dbr(ra
and how to check null value taken from database?
$dbr(ra=>[0,0]) == ?
On Mon, Jan 14, 2013 at 1:44 PM, Alex Balashov wrote:
> if(defined $var(x))
>
> Or, if checking for empty value:
>
> if(strempty($var(x))
>
> Mino Haluz wrote:
>
> >Hi,
> >
> >how should I check if the value is set?
> >
> >i
if(defined $var(x))
Or, if checking for empty value:
if(strempty($var(x))
Mino Haluz 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
>
>
>---
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
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users
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 t
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