Hi Wesley,
Hope this is what you seeking.
modparam("tm", "fr_inv_timer_avp", "$avp(s:timeout)")
and you can load different timeout to avp using like below, this is per username
avp_db_load("$ru/username","$avp(s:timeout)");
and you can trigger failure route as follow,
t_on_failure("1");
yo
Hey Antonio,
It seems to work great in my test environment!
Thank you very much!
Cheers
On 12 August 2010 17:09, Antonio Anderson Souza <
anto...@voicetechnology.com.br> wrote:
> Wesley,
>
> Yes, it's possible to control the timeout per route, this could be made by
> the fr_timer_avp and fr_inv
Wesley,
Yes, it's possible to control the timeout per route, this could be made by
the fr_timer_avp and fr_inv_timer_avp, you need just set a valeu in seconds
in those avps to control the timeout per branch.
Have a look in the documentation of TM module [1].
[1] -
http://www.opensips.org/html/do
Hello All,
Is there any way to configure a timeout per route?
What I mean is: If my first route doesn't send a reply(100 Trying or any
other) in some seconds, the opensips sends the the call to failure route,
and then I use failure route to try another route. How can I do that?
Regards,
--
Wesl
Hi Erik,
do you develop your own module? or where do you use the recvfrom ?
Regards,
Bogdan
erik.buel...@telenet.be wrote:
> Hello,
>
> In my setup recvfrom does not return with data while I can see the
> data arriving on the specified IP:port with tcpdump.
>
> Any idea what could be verified?
Brett Nemeroff wrote:
>
> On Tue, Aug 10, 2010 at 7:57 AM, Bogdan-Andrei Iancu
> mailto:bog...@voice-system.ro>> wrote:
>
> Hi Alex,
>
> Of course, if you use the dialog support - anyhow, we have a ready
> patch
> for acc module to do accounting based on dialog support (directly
>
Matt,
I was looking at the code and the $DLG_lifetime return the difference
between the current time and the "start_time" (which the moment when the
dialog was confirmed with 200 OK ).
Regards,
bogdan
Matt lehner wrote:
> I will have to look at this again. This was not the result I saw when
>
Hi Mark
Thanks for the feedback.
Appreciate.
Will investigate into the mentioned systems.
Kind Regards
Deon
On Aug 12, 2010, at 2:46 PM, Mark Sayer wrote:
> The simple answer is yes, but not with OpenSIPS alone. You'll need
> something like Asterisk or FreeSWITCH to handle the PBX functions.
The simple answer is yes, but not with OpenSIPS alone. You'll need
something like Asterisk or FreeSWITCH to handle the PBX functions.
Mark
On Thu, Aug 12, 2010 at 8:48 PM, Deon Vermeulen
wrote:
> Good Day List
> Hope you well.
> I would like to find out if someone could assist me.
> I am in the
Good Day List
Hope you well.
I would like to find out if someone could assist me.
I am in the process of setting up a SIP Server hosting a few companies PBX
Functionality, but require some assistance/guidance.
What I am looking for is:
1) Support for Multiple Domains - Each Company with their o
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