Ahhh, cool. Well the db is mysql and sits on the same box so probably fine.
Just don't really know.
I'll keep this in mind and apply if needed.
Thanks for all your help, amazing!
Keith
On Mon, Oct 14, 2013 at 3:46 PM, Klaus Darilion <
klaus.mailingli...@pernau.at> wrote:
>
>
> On 14.10.2013 14:
On 14.10.2013 14:57, Keith wrote:
Hi,
Klaus, thank you for pointing me in the right direction with SIP trunks,
got it working so thanks! Basically I did exactly what you said:
- Dialled number
- Match that number to a registered user (had to create a new table for
that)
- Lookup user
- Replac
Hi,
Klaus, thank you for pointing me in the right direction with SIP trunks,
got it working so thanks! Basically I did exactly what you said:
- Dialled number
- Match that number to a registered user (had to create a new table for
that)
- Lookup user
- Replace dialled s@ with original number dial
Am 09.10.2013 17:56, schrieb Keith:
Hi,
Can anyone point me in the right direction for setting up SIP trunks?
Whenever I send a call to a registered user on a trunk it just sends
to destination s@x.x.x.x. Is there anyway to say "these extensions are
location at this destination IP and port".
F
Hi,
Can anyone point me in the right direction for setting up SIP trunks?
Whenever I send a call to a registered user on a trunk it just sends to
destination s@x.x.x.x. Is there anyway to say "these extensions are
location at this destination IP and port".
Cheers,
Keith
___
Surely there must be a better way.
For example, if you are a good client for your provider, you can ask him to
forward all calls to your IP. This way, Kamailio will accept all calls and
forward them to multiple Asterisk boxes for processing.
Also - you can put one Asterisk to do only one thing - ac
Stoyan thanks for your reply, i've been doing some research before
replying (which has taken a while) and there's something I don't
understand. I apologize in advance if I'm asking something that makes no
sense.
my provider does in fact requires registration, and they provide a
single sip tha
It depends.
I can imagine next scenarios:
1. Under SIP trunks you mean calls from your provider to you
A) In case your provider can send calls to you - then you can use Kamailio,
accepting all calls from your provider - based on IP.
B) In case your provider expects registration from your system - t
Hi
I'm planning to set kamailio in front of an farm of pbx servers
(haven't decided on freeswitch or asterisk) there's a million tutorials
on how to do this, what I haven't found is what part of my setup
actually handles the sip trunks my phone company provides me with.
What's the best pract
Keith Hubner writes:
> Authentication for users is working fine for me, I add a user to the
> db and the phone can register. I am looking to connect a SIP trunk but
> there doesn't seem to be any authentication? What I want is to allow
> users to connect from any IP but only allow SIP trunks to co
Hi,
Authentication for users is working fine for me, I add a user to the db and the
phone can register. I am looking to connect a SIP trunk but there doesn't seem
to be any authentication? What I want is to allow users to connect from any IP
but only allow SIP trunks to connect from certain IPs
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