Call conference using Brekeke PBX/SIP Server and X-Lite

[kaustav.alcatel]

1. Brekeke Product Name and Version:Brekeke PBX, Version 3.0.7.0 , Pro Evaluation

2. Java version:

3. OS type and the version:Win XP Ver 5.1 Service Pack 3

4. UA (phone), gateway or other hardware/software involved: X-Lite Softphone 5.0.0

5. Your problem: Hi I am trying to setup a multi party (10-12) conference call using Brekeke PBX/SIP and Softphone CLient X-lite.
I have been able to establish a three party call conference, but users more than that are unable to join conference. Another problem that I am facing is the call gets disconnected within exactly 30 seconds. My call is transiting through Wireless Call server 5060 of Alcatel-Lucent which has got IPM Cards installed ,which can handle SIP calls. So I call from a mobile phone that is in the network in which the Wireless Call Server is there and my Laptop is connected to the omniswitch of the Wireless Call Server, hence can communicate with the IPM cards that are handling the SIP calls, thus the call terminates on my Laptop, which is acting as a SIP server. Below is the detailed description of things that I have tried at my end. Kindly respond after analysing.

Following things have been done till date –

.
1. Installation of X-Lite softphone dialer on laptop and registering it with Brekeke. So it shows as a registered client in Brekeke PBX
2. Configuring laptop ip as 10.134.25.104 , subnet – 255.255.255.0 and default gateway as 10.134.25.110
3. 555555555555555 number routed from Call server 1 to call srerver 2 on BICC and then from call server 2 to the SIP server (my laptop) using newly created SIP group.
4. Now call scenario tested till now is as follows - party 1 dialing 555555555555555 is landing on my laptop on X-lite and both way speech is audible. Outgoing call from laptop however is not yet confirgured and isn’t working.
5. Once call connected another party can call and it can be received on X-lite softphone and these calls can be merged into a conference call of three parties i.e myself , party 1 and party 2.
6. After this when the next party tries making a call to 555555555555555 , the line busy announcement is heard.
7. I have tried using the brekeke manual for configuring conference room (by creating extensions) but it hasn’t been successful till now.
8. Snapshot of conference room setup procedure from brekeke manual is as follows –

a.
6.10. Setting Up Conference Call (Page 19 in attached manual)
6.10.1. Creating a Conference Room
The first step to using the Conference Call feature is to set up a Conference Room. In the following example, user 2000 is set up as the conference number.
If you are using SIP phone that support “Conference Call” button, please consult the manufacturer of your SIP phone for how to set up Conference Call feature.
1) From [PBX Admin] > [Extensions] > [Conference], click [Create a new conference] button
2) Set as following:
[Extension] 2000


Leave all other settings as default


3) Save the settings
With the above settings, any user can enter in the conference room by dialing 2000.


Kindly Help.

Regards,
Kaustav Bhattacharyya

[hope]

you can dial as 2000*user1*user2*user3 to invite more users to join the conference
or set these users at conference ext "Forwarding destinations*"
when dial 2000 these users phone will ring too

there are sample at pbx doc user guide.

[kaustav.alcatel]

Hi hope,
Thanks for the reply. I have checked the document samples. So do you mean the following -

[PBX Admin] > [Extensions] > [Conference], click [Create a new conference] button
2)
Set as following:
[Extension] 2000
3)
Save the settings
With the above settings, any user can enter in the conference room by dialing 2000.


So now this call that is received by dialling 2000 is received on X-lite softphone or are you using some other client along with brekeke PBX?

[hope]

do you have any problem to make conference call?
can xlite and other parties dial to conference extension 2000?

[kaustav.alcatel]

[hope]

when making above call and from pbx/sip server side open related session detail and look for the rule name used for the call

if call is routed to pbx side then go to pbx / call status and check ARS rule used for the call
in ARS rule pattern-IN, route the incoming calls to conference extension