Hello,
maybe is better to use sqlops module, it more suitable for queries with
many records in result.
http://kamailio.org/docs/modules/stable/modules_k/sqlops.html
Cheers,
Daniel
On 1/19/11 2:58 PM, Klaus Darilion wrote:
looks fine. try to increase debug level - then you should see the
On 1/19/11 7:50 AM, Klaus Darilion wrote:
Am 18.01.2011 21:26, schrieb Brandon Armstead:
Hello,
Is there anything special that needs to be done for float
comparison?
For example:
if([5.5 = 4.3])
^^^ this format is no longer supported starting with 3.0, just skip the
square
On 1/16/11 7:32 PM, David J. wrote:
I am trying to add support for call transfer in the Asterisk realtime
tutorial on Asipto;
I am not sure what I would have to do to get this feature working;
Perhaps I have to handle refer messages; but I am not sure how I
send that to Asterisk;
Any
Am 19.01.2011 19:14, schrieb Sébastien Cramatte:
Efectively it was an issue with FW. I've setup my kamailio to listen on
5060 and 5062.
Now I can call media server and my extensions can receive / make local
calls respectively.
The issue now is that I've got my SIP PSTN gateway that try to
private mails are simply ignored after first advise in this regard,
please CC the mailing list always.
If you read the config from the tutorial, you see how the invite is
relayed to asterisk. Refer should go to asterisk in the same way if it
is an out of dialog request, or follow record
Klaus,
Thanks for your continued help.
How? Does it change the Contact header?
Yes, the proxy changes the Contact header.
So the SIP server is the registrar?
Correct.
The contact usually is the IP address of the client. So, if you the SIP
server routes based on the contact header, it
The contact usually is the IP address of the client. So, if you the SIP
server routes based on the contact header, it should send the INVITE
directly to the client not to the proxy. Somehow this all does not fit
together.
Clarification : Both clients A and B have the proxy as an outbound
Klaus,
To me it seems that the SIP server also does some kind of NAT traversal:
it puts the Contact IP in to the RURI but it sends the request to the
IP:port from which the REGISTER was received (that's called NAT traversal).
So, either fix the SIP server (make sure it adds the port as in the
On 13 January 2011 08:43, marius zbihlei marius.zbih...@1and1.ro wrote:
On 01/13/2011 02:51 AM, dotnetdub wrote:
On 29 November 2010 09:33, marius zbihlei marius.zbih...@1and1.ro wrote:
On 11/26/2010 12:38 AM, dotne
Hello Brian,
Module: sip-router
Branch: 3.1
Commit:
