how many G729 license do you have?

it's codec problem, which codec do you set on the phone?

Yes, Brekeke PBX supports the voice broadcast, Try to use paging function that is supported by v2.2.
Good Luck!

Does your FXO gateway support 2 stage dialing?
if so, change the FXo gateway setting to 2 stage dialing, and dial 101. once you hear the dial tone, dial the PSTN number.

here is the link for the dial plan tutorial
http://www.brekeke-sip.com/download/bss ... lan_en.pdf

Please go to callee's ID that you create at tab [users], uncheck voicemail that is next to the forward destination(no answers) and Forward destination(busy). Retry the call: the callee doesnot pick up the call or reject the call, the call will be terminated, it will not retry it, unless you are ...

it's possible, but you need to have two PBX and on each side, you need FXO gateway.

what number did you assign for the FXO port? it's 9? if so, please try to use other number. for two stage dialing, you donot need to use ARS, ARS is only for one stage dialing.

Please to update your brekeke sip server to latest version. and try to use following dial plan: ---------------------------------------------------------------------- Matching Pattern: $request = ^INVITE Deploy Pattern: $session=plugin.radius.RadiusAcct $continue = true ...

1. Brekeke Product Name and version: Brekeke PBX 2. Java version:1.5 3. OS type and the version: Windows XP 4. UA (phone), gateway or other hardware/software involved: 5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html : 6. Your problem: Hi, I changed my ...

Hi I suggest you should use Brekeke PBX, because using ARS is more easy to configure. I am using Brekeke PBX with OCS 2007 too, I could use office communicator to make a call to PSTN or receive call from PSTN. Can your office communicator make call to UA which is registered to Brekeke SIP? by the ...

you may find it at the Brekeke PBX Administrator's Guide (Advanced) Page34,35,36 http://www.brekeke-sip.com/download/bpbx/v2_1/bpbx_admin_advanced_en.pdf For changing waiting time before the call goes to MOH/operator, Please refer to the following link: http://www.brekeke-sip.com/wiki/index.php ...

you can use Brekeke PBX to make call transfer. if your x-lite's extension is 111, you need to create user 111 at pbx admintool > user, when you make call transfer to 222, you need to dial #9 first, once you hear the music on hold, you can dial 222#, that will do it.

I read the packets which show the mediatrix gateway sends 480 temporarily unavailable to Brekeke PBX. Did you receive any support by Mediatrix?

those features(call fowarding, distribution, and Queueing) are in the User setting, First thing you need to do is Creating NEW User at Users tab, under each user, you will see those features.

you should contact sale@brekeke.com

I donot have that problem. Make sure you set the userID and Password at Brekeke SIP > User Authentication, Not Brekeke PBX >users

Can you post the packets? i would like to see who send "480".

is 9406222 the number you want to make call, if so, you need to dial 99406222 or 9(area code)9406222.

Try to update to new version:2.2.1.1

try to download the Ethereal, it's free network analyzer, is a useful tool to monitor SIP and RTP packets.

Question1:
if the call was made from PSTN to FXO gateway, you have to configure it at gateway.

Question2:
can you tell me which phone you are using and if possible, can you capture the packets and post it?

Hi Niggy,
once you change your PC ip address as the FXO subnet, you can access the gateway from web interface, you need to change the gateway ip as your PC subnet and change your PC IP back.

Try to use two dial plans with different priority. ---------------------------------------------- [Matching Patterns] $request=^INVITE To=sip:(1.+)@ [Deploy Patterns] To=sip:%1@202.163.211.23 ----------------------------------------------- [Matching Patterns] $request=^INVITE To=sip:(6.+)@ [Deploy ...

it doesnot matter which side you configure first, if you need to call from voip gateway to PSTN, you need configure it at PBX > ARS, and if you need to receive call from PSTN to VOIP gateway, you need to configure it at voip gateway. here it is other gateway's configuration document, it may help ...