yeah, i guess there is no standard for this.
thanks for the tip!
Kelvin Chua
On Fri, Jun 19, 2015 at 6:27 PM, Daniel-Constantin Mierla wrote:
>
>
> On 19/06/15 11:52, Kelvin Chua wrote:
> > on my registrar server:
> > after calling lookup, and getting AOR with path, how do I send the
> > path
Hi Daniel,
On Fri, Jun 19, 2015 at 12:02 PM, Daniel-Constantin Mierla <
mico...@gmail.com> wrote:
> there was a bug returning 0 in case of a no match. I just pushed a fix
> in master, going to backport it.
>
the patch is working, thanks.
Best Regards,
Sebastian
_
Hello,
Suppose I have $var(a) = 1234#5678 and $var(b)=123455667
I need to get the number before '#' character if exists , meaning that from
$var(a) I must get 1234# and from $var(b) I must get empty string.
I tried $var(pref)= $(var(a){s.select,0,#}); but I'm getting the whole
string if #
Hello. I Installed kamailio on ubuntu 14.04 that runs as virtual
systemOpenVZ.
after starting kamailio I see that it runs ok with
kamailio start
or
kamctl start
at ps -ax I see working processes.
But after one minute of working kamailio fails with
kamailio: ERROR: [daemonize.c:315]: daemonize()
On Monday 22 June 2015 13:58:28 Ali Taher wrote:
> I tried $var(pref)= $(var(a){s.select,0,#}); but I'm getting the whole
> string if # doesn't exists.
>
> There is a way to check if # exists by subtracting the length of the
> string before and after removing # character and check if length cha
Hello,
my setup is a kamailio server as registrar and Blox as SBC.
Blox is a freeware GUI with Opensips as framework.
Kamailio is located in the private network and Blox operates with 2 NICs
(private and public)
Now the following problem:
If I make a call from public (phonerlite) to private (maya
On 06/22/2015 10:30 AM, kai.ohnac...@cbc.de wrote:
> Hello,
>
> my setup is a kamailio server as registrar and Blox as SBC.
>
> Blox is a freeware GUI with Opensips as framework.
>
> Kamailio is located in the private network and Blox operates with 2 NICs
> (private and public)
>
> Now the fol
In the txt file you can find the ngrep traffic.
Cheers,
Kai
-
Perhaps you have an ngrep of the sip traffic?
--fred
-Ursprüngliche Nachricht-
Von: sr-users [mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von
Fred Posner
On 06/22/2015 11:41 AM, kai.ohnac...@cbc.de wrote:
> In the txt file you can find the ngrep traffic.
>
> Cheers,
> Kai
>
> -
>
> Perhaps you have an ngrep of the sip traffic?
>
> --fred
>
Is .3 the user, .1 the kamailio, and .33 the SBC
Kamailio itself doesn't alter CSeq method, unless you do something in
the configuration file.
Looking at the traces, the broken CSeq comes like that to Kamailio --
first part here indicates that 192.168.0.1:5060 is Kamailio, and next
shows the packet sent by .0.33 to .0.1 (kamailio):
U 2015/06/22
Hello,
thanks for reporting back the results -- it will get in stable branches.
Cheers,
Daniel
On 22/06/15 11:30, Sebastian Damm wrote:
> Hi Daniel,
>
> On Fri, Jun 19, 2015 at 12:02 PM, Daniel-Constantin Mierla
> mailto:mico...@gmail.com>> wrote:
>
> there was a bug returning 0 in case of a
On 22/06/15 16:16, Daniel Tryba wrote:
> On Monday 22 June 2015 13:58:28 Ali Taher wrote:
>> I tried $var(pref)= $(var(a){s.select,0,#}); but I'm getting the whole
>> string if # doesn't exists.
>>
>> There is a way to check if # exists by subtracting the length of the
>> string before and aft
On 06/22/2015 03:05 PM, Yuriy Gorlichenko wrote:
> Hello. I Installed kamailio on ubuntu 14.04 that runs as virtual
> systemOpenVZ.
git or deb?
> after starting kamailio I see that it runs ok with
> kamailio start
> or
> kamctl start
If deb:
I would say that you should use /etc/init.d/kamai
git
OpenVZ have an issue with linux headers. When I run uname -r it give me
heades 2.6 that works at the host server. Ubuntu have 3.13.
I think it may be main reason of this issue or simphtom af another issue at
OpenVZ systems...
2015-06-22 22:33 GMT+03:00 Victor Seva :
> On 06/22/2015 03:05 PM,
Hello. I'm trying to set up this (v 4.2 stable):
peer <--> ec2 <--kamailio+rtpengine--> asterisk
scheme
I use advertised adress for SIP and WS connections.
The problem is that on SIP I get one way audio - I can receive audio from
asterisk, but I can't transmit audio there - my SIP UA tries to send
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