delay on ringing on outgoing calls

simonj · Post by **simonj** » Thu Jul 31, 2014 9:01 am

1. Brekeke Product Name and Version:

Brekeke SIP Server , Version 2.4.8.6 Advanced

2. Java version:

java 6 update 45

3. OS type and the version:

windows server 2008

4. UA (phone), gateway or other hardware/software involved:
Atran 908

5. Your problem:

OUTGOING
$request=^INVITE
To=sip:(.+)@
$addr=60.60.60.60 ( sample )

$auth=false
To=sip:%1@71.71.71.71
$session=failover %1@72.72.72.72 %1@73.73.73.73
&failover.timer.inviting=5
&failover.timer.provisional=360
$continue=false

my problim is that 60.60.60.60 is a adtran 908

its sends a call to the brekeke sip server the brekeke sip server sends it to provider 1 sip:%1@71.71.71.71

to some places it might take provider 1 send back a 180 ringing (1 or 2 sec) it wont failover to next it got a 100 trying

what i want to do , as soon as it gets back a 100 from provider 1 the brekeke sip server should send back a 180 ringing to the adtran 908

Thank You for your help
SJ

simonj · Post by **simonj** » Thu Jul 31, 2014 9:30 pm

Test 1 i tried to make a role that end point sending the calla 183 so that the end user should get back a ringback in freeswitch this opion is called ring ready

The problim is that ones the brekeke send the $action=183 the brekeke closes the call and will not continue to the next role ( i stays ringing for the user

$request=^INVITE
To=sip:(.+)@
$addr=55.55.55.55

$auth=false
$action=183
$continue=true

Test 2
i tried to put the $action=183 in the same role that is sending the call to the rovider well still same problim after the $action=183 the brekeke closes the call

$request=^INVITE
To=sip:(.+)@
$addr=55.55.55.55

$auth=false
$action=183
To=sip:%1@99.99.99.99
$session=failover %1@88.88.88.88
&failover.timer.provisional=360
&failover.timer.inviting=5

any ideas will be appreciated

Thank you

Laurie · Post by **Laurie** » Thu Jul 31, 2014 10:21 pm

There are two ways.

- Write a plugin to send "183" response and keep the session.

- Use Brekeke PBX instead of Brekeke SIP Server.
Brekeke PBX is an IP-PBX and B2B-UA. so it can send "18x" response by itself.

simonj · Post by **simonj** » Thu Jul 31, 2014 10:39 pm

Thank you for your quick reply

for a fee is it possible for your company to write the plugin ?

is there any documents on how to link the plugin to brekeke ?

Thank You

Laurie · Post by **Laurie** » Thu Jul 31, 2014 10:59 pm

Ask Brekeke to write a plugin

simonj · Post by **simonj** » Sun Aug 03, 2014 11:18 am

Hi I wrote the code bellow and it doesn't seem to do anything.

I droped the jar file in the lib directory.

I am calling it with

$session=com.test.plugin.TestAction 183

Code: Select all

package com.test.plugin;

import com.brekeke.net.sip.sv.session.EventStat;
import com.brekeke.net.sip.sv.session.plugins.AccountingBase;

public class TestAction extends AccountingBase {
    public int eventSessionStart( EventStat evstat, String[] argprm )
    {
        String codeStr = this.arg[0];

        Integer code =  Integer.parseInt(codeStr);
        String str = this.ss.rulematch.header.get("Contact");

        try {

            this.sipex.sendResponse(this.ss, this.ss.csSrc, this.ss.packetInitial, code, str, null);
        } catch (Exception e) {

        }

        return 0;
    }
}

[/code]

Laurie · Post by **Laurie** » Mon Aug 04, 2014 10:51 am

Let me try your code!