OpenSIPS is now randomizing correctly. I guess I must have made a
configuration error or something.
Sorry for the noise.
Alex
On Thu, Jun 7, 2012 at 4:13 PM, Alejandro Recarey wrote:
> Hi all,
>
> I am using opensips 1.6.2 and am using drouting module like this:
>
> do
Hi all,
I am using opensips 1.6.2 and am using drouting module like this:
do_routing("1","1");
According to the manual this means "group 1" and "randomize members in
each group". My group has the following gateways:
13,14
When I run opensips, the calls only go to gateway 13. I thought that
Hi all,
I've been trying to configure the drouting module, but I must not
understand how to use it. When I use my drouting opensips.cfg file,
the opensips server stops replying to INVITE requests. The file has no
syntactical errors, and there is nothing in the log.
My dr_rules table contains some
Hi all,
I recently implemented OpenSIPS to load balance calls to an asterisk
cluster, using the dispatcher module and taking advantage of the
active probing for failed destinations feature.
Everything works like a charm, except when one of the destinations
goes to probing or inactive state. When
l
> INVITE message by using the Record-Route mechanism. So basically you will
> load-balance just the initial requests ( INVITE ), and let the sequential
> requests within the dialog follow the same path.
>
> Regards,
>
> Vlad Paiu
> OpenSIPS Developer
>
>
> On 09/
Hello all,
I was reading the documentation on the dispatcher module and it says:
"Is dispatcher dialog stateful?
No. Dispatcher is stateless, although some distribution algorithms are
designed to select same destination for subsequent requests of the
same dialog (e.g., hashing the call-id)."
I
I reinstalled Opensips to version 1.6.4 compiled from source, fresh install.
I am still having these strange messages in my opensips log, and I get
about 40 lines of
error messages per second!
I am very worried. A packet capture on port 5060 found normal sip
packets and some stun
packets, strange
Hello all,
I am currently running 2 opensips, one for "inbound" calls and one
for "outbound". My inbound OpenSIPS handles registrations and load
balances the calls over asterisk gateways. The "outbound" opensips
just proxies outbound calls for the VoIP operators who wish to receive
all calls from
Solved, sorry for the bother. It's related to errors in the comunication
between syslog and opensips because of a faulty configuration.
Thanks anyways!
Alex
On Wed, May 25, 2011 at 7:43 PM, Alejandro Recarey
wrote:
> I am getting the following errors at opensips startup:
>
>
>
I am getting the following errors at opensips startup:
May 25 19:41:40 c1opsip1 opensips[5417]: INFO:core:sig_usr: signal 15 received
May 25 19:41:40 c1opsip1 opensips[5411]: INFO:core:sig_usr: signal 15 received
May 25 19:41:40 c1opsip1 opensips[5409]: INFO:core:sig_usr: signal 15 received
May 2
Hi all,
I have been checking the SIP security of my configuration and am shocked
to find out that my configuration is currently not working correctly.
I am using OpenSIPS 1.6.2 and the check_source_address function to only
allow calls from my own domain but it seems that no matter what I write
to
()
> functions :
> http://www.opensips.org/html/docs/modules/1.6.x/tm.html#id272062
>
> And make the condition like:
>
>
> if (t_check_status("(408)") && t_local_replied("all")) {
>
>
> Regards,
> Bogdan
>
> Alejandro R
Hi all,
I am currently trying to enable GW disabling from inside the routing
script with the load balancer.
I managed to get it working reading the module documentation, but now
I have a problem. I want the gateway to be disabled only if it is
unreachable, that is, if it does not answer at all to
Hi all, and thanks for taking the time to read my mail.
I am currently studying OpenSIPS to replace Asterisk in a network that
I administer. I am doing this because Asterisk call quality quickly
starts degrading once you hit 100 simultaneous calls.
Although NAT issues are not terribly important (
> asterisk should not be confused at all - according to SIP, proxies on
> the way may change the RURI without breaking anything. The idea is to
> change only the RURI (for original INVITE), so that the SIP dialog will
> be consistent for its whole duration.
Hello Bogdan, thank you for the insight
I'm currently using OpenSIPS as an outbound proxy for my asterisk
boxes, presenting only one IP to my providers. However, sometimes,
their service is less than stellar and it goes down, which is why I
want to be able to do OPTIONS probing and disable failed gateways, and
re-enable them when they go
I am using OpenSIPS to load balance incoming calls between my asterisk
boxes, with great results! The LB module is perfect for this.
I am currently configuring OpenSIPS to also act as an outbound proxy
to my asterisk boxes. This is because some VoIP providers want you to
send them all calls from a
Great help, thank you both!
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Hi all,
I'm trying to use the keepalive / ping option in OpenSIPS Loadbalancer
module, but asterisk always responds with a 404 not found. Has anybody
managed to get this working with Asterisk 1.6.1.X + OpenSIPS 1.6.2 ?
Thanks!
Alex
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where
> opensipsctl finds the FIFO.
>
> Flavio E. Goncalves
>
> -Mensagem original-
> De: users-boun...@lists.opensips.org
> [mailto:users-boun...@lists.opensips.org] Em nome de Alejandro Recarey
> Enviada em: Monday, March 01, 2010 4:43 PM
> Para: users@lists.opensips.o
Hello Brian,
I'm running OpenSIPS as opensips:opensips.
Although I checked that /tmp/ has chmod 777, I created the opensips
user as a system user so it might be more restricted.
Maybe that is the problem? I don't mind running OpenSIPS as root, as
it will be the only service running on my box (it
Hello everybody,
I have just made a fresh install of OpenSIPS 1.6.1 using the
http://debian.leurent.eu/ mirror.
I am using debian lenny 5.0. I have read all of the tutorials and I
believe I have a working configuration. OpenSIPS is running and my cfg
file is loaded without problems.
However, whe
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