Thanks.
On Fri, Apr 24, 2015 at 2:01 PM, Eric Tamme wrote:
> As I said, failure route is transaction based, so any transaction failure
> will trigger it. see
> http://www.opensips.org/Documentation/Script-Routes-2-2#toc3
>
>
> On 04/24/2015 11:57 AM, Podrigal, Aron wrote:
>
> The problem is th
As I said, failure route is transaction based, so any transaction
failure will trigger it. see
http://www.opensips.org/Documentation/Script-Routes-2-2#toc3
On 04/24/2015 11:57 AM, Podrigal, Aron wrote:
The problem is that there is no response coming back, media server 1
crashed.
On Fri, Apr
I guess I need to set fr_timer .
On Fri, Apr 24, 2015 at 1:57 PM, Podrigal, Aron
wrote:
> The problem is that there is no response coming back, media server 1
> crashed.
>
> On Fri, Apr 24, 2015 at 1:54 PM, Eric Tamme wrote:
>
>> failure_route is called based on a transaction failure - so eve
The problem is that there is no response coming back, media server 1
crashed.
On Fri, Apr 24, 2015 at 1:54 PM, Eric Tamme wrote:
> failure_route is called based on a transaction failure - so even if the
> request was in dialog, as long as you t_relay() it and it gets a response
> code >=300 fai
failure_route is called based on a transaction failure - so even if the
request was in dialog, as long as you t_relay() it and it gets a
response code >=300 failure route will be called on receipt of the response.
-Eric
On 04/24/2015 11:22 AM, Podrigal, Aron wrote:
Hi,
What is the right way
Hi,
What is the right way to handle failures for sequential requests. So if
Media Server #1 hangs, I wanna create a new transaction route to Media
Server #2.
--
Aron Podrigal
-
//Be happy :-)
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Doesn't this cause more problems? I thought bypass media behind NAT or in mixed
systems - NAT/no NAT - is a mess.
I would love to have a local (dumb) proxy on each site to handle the local
calls but all the intelligence (user database/ management) should remain in the
centralized master.
Remember that if the audio path is p2p the DMTF tones do not work
anymore, so you can't transfer a call, etc. You should use SIP INFO if
supported.
For the audio path give a look to the asterisk options /directmedia/=yes
and directrtpsetup=yes. You could not need to use openSIPS and rtpproxy.
As an alternative you could try bypass media.
https://freeswitch.org/confluence/display/FREESWITCH/Bypass+Media+Overview
On Fri, Apr 24, 2015 at 11:17 AM, Roman Dissauer wrote:
> Dear all,
>
> I’m running a centralized Freeswitch based PBX for use on several sites.
> All phones register against
Dear all,
I’m running a centralized Freeswitch based PBX for use on several sites. All
phones register against this Freeswitch instance.
Now I want to install SIP and RTP Proxies on every site to keep RTP traffic
locally on site. Registration should still be done by freeswitch. Can anybody
giv
Just in case somebody deal with the same issue, the problem seems to be
event_xmlrpc module. I tried with the event_datagram to notify the external
process and I got no more crashes for a couple of weeks.
Now I'm using event_rabbit module instead of datagram without problems for
a couple of days.
Hi!
How to check To/From username against digest credentials if
radius_send_auth() used instead of aaa_www_authorize/aaa_proxy_authorize.
If I call db_check_to() after successful radius_send_auth() call I get this
errors:
Apr 24 14:02:28 sip-node1 /sbin/opensips[2653]: ERROR:uri:check_username:
Just a small typo:the meeting's topic is "OpenSIPS Easy deployment"[2] :).
See you on Wednesday!
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 04/24/2015 10:49 AM, Răzvan Crainea wrote:
Hello, All!
The next public meeting is scheduled for 29th of April, 2015, 17:00
EET[1].
Hello, All!
The next public meeting is scheduled for 29th of April, 2015, 17:00
EET[1]. The topic is "OpenSIPS Easy development".
You can find more information about what we are planning to discuss the
next Wednesday on the event's page[2].
Looking forward to seeing you all next Wednesday on
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