[OpenSIPS-Users] Timer based Failover to SIP Provider

duane.larson at gmail.com duane.larson at gmail.com
Wed May 25 18:49:39 CEST 2011


I just took the OpenSIPSB2BUA server out of the equation. So now the  
OpenSIPSProxy sends the calls directly to the SIP provider. With this test  
the failover does work. So the only difference here is that OpenSIPSB2BUA  
is no longer in the middle of the proxy and the SIP provider and also  
instead of setting

$du = "sip:" + $rU + "@173.XX88:5060"; # IP Address of my OpenSIPS B2BUA  
server
$ru = "sip:" + $rU + "@4.2.2.2:5060";

I have to set $rd to the SIP provider.

So I am not sure why I would be seeing the issue when the OpenSIPSB2BUA is  
in the middle of the proxy and the SIP provider.

On May 23, 2011 5:04pm, Duane Larson <duane.larson at gmail.com> wrote:
> I do have it set. Forgot to include my modparams








> modparam("tm", "fr_inv_timer", 20) #When ringing callee and receive a 180  
> RINGER reply after x seconds failover
> modparam("tm", "fr_inv_timer_avp", "$avp(s:fr_inv_timer)")


> modparam("tm", "fr_timer_avp", "$avp(s:fr_timer)")
> modparam("tm", "onreply_avp_mode", 1)


> modparam("tm", "restart_fr_on_each_reply", 1)


> modparam("tm", "pass_provisional_replies", 1) #This is needed for B2BUA










> On Mon, May 23, 2011 at 5:00 PM, Brett Nemeroff brett at nemeroff.com> wrote:








> On Mon, May 23, 2011 at 4:41 PM, Duane Larson duane.larson at gmail.com>  
> wrote:




> I am trying to set up some logic that will allow me to failover to a  
> secondary SIP Provider's IP address and I found this subject on how you  
> can implement it





> http://opensips-open-sip-server.1449251.n2.nabble.com/Timer-Based-Failover-Question-td5758903.html










> In order to test I send my first call to 4.2.2.X (which is not my SIP  
> providers Gateway). So this call fails as planned. Then a second call is  
> sent to the correct backup SIP Providers IP address. This call starts to  
> go through but then my Timer times out and the call is canceled. I am not  
> sure why the call is canceled because I get a 183 from the SIP provider  
> and in my OnReply_Route I set the timeout to be 300 seconds. It seems  
> like when I set fr_inv_timer_avp it doesn't take affect. Not sure what I  
> am doing wrong. Here is what I do












> Have you set:


> modparam("tm", "onreply_avp_mode", 1)






> -Brett






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> --
> --
> *--*--*--*--*--*
> Duane
> *--*--*--*--*--*
> --



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