Really need help for a basic PSTN setup

[hacktek]

1. Brekeke Product Name and version: Brekeke PBX/Sip Server 2.4.3.0

2. Java version: 1.6

3. OS type and the version: Windows 2003 Server R2

4. UA (phone), gateway or other hardware/software involved: 3CX Softphone, Handytone HT-503 ATA as gateway (1 FXO, 1 FXS)

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html : Pattern 2

6. Your problem:

I have 0 experience in voip and really need help setting a basic infrastructure with Brekeke. Basically i need softphone to softphone calling (i have been able to do this by creating the users in the Sip server) and softphone to PSTN calling. Also, inbound calling should work.

Dial rules seem very confusing to me, can someone give me a simple newbie scenario? This is what i have:

HT-503 Gateway @ 192.168.230.180:5060
Brekeke Server @ 192.168.230.170:5060

As additional info, when i pick up the phone connected to the gateway i need to dial 9 to reach out Analog PBX here at work and then 9 again to reach the PSTN.

Thank you.

[taitan]

I want to know what you want to do...

Can you make a call between softphones now?

>> As additional info, when i pick up the phone connected to the gateway i need to dial 9 to reach out Analog PBX here at work and then 9 again to reach the PSTN.

Is it gateway's requirement?
Do you want to use '9' as the prefix of phone numbers to reach the PSTN?

[hacktek]

Thank you for your quick response.

1.

What i want to do is call from the soft phone to the outside world.

I am able to talk to make calls between soft phones just fine (and from a soft phone to the phone plugged into the FXS port of the gateway).

I cannot make calls from a soft phone to the PSTN, which i think would be the next logical step.

2.

The gateway by default works in VoIP mode, which means that when i pick up the phone connected to it i get the Gateway's dial tone and i can reach extensions via the sip server. This works fine.

If i press 9 i get to the second dial tone, which is our analog PBX here at work. From here we can reach lines that still aren't on the VoIP network.

Pressing 9 again makes me reach the PSTN so i can make regular phone calls.

What i think i need are Dial rules to reach each of those communication tiers (if that makes any sense). In short:

Tier 1: Pick up phone = VoIP, sip extensions
Tier 2: Press 9 when in Tier 1, reach analog extensions
Tier 3: Press 9 when in Tier 2, reach any normal phone number

Make sense?

Thanks

[hacktek]

Actually for some reason it isn't working anymore, i cannot call between soft phones

I get "Destination is busy", a 486 error in the BSS log.

The 2 soft phones and 1 analog phone i'm using (connected to the gateway) show up as registered on BSS.

Let's start at the beginning: do i need to set up a dial plan rule to call between soft phones? Right now i have a default installation.

Thanks

[voipwell.com]

No, you do not need to set up any special dial plan to call between registered softphones or hardphones.

go to the sip server sip configuration tab and look under authentication and find the field realm/domain. Is it blank? Put in there the domain name/ host name of your server by looking at an ipconfig /all to get it. If you don't see one put in localhost there. Restart the sip server and see if that allows you to call each other.

[hope]

are you using sip server only or brekeke pbx?
if it is brekeke pbx, you need to create pbx user for each registered phones with the same number shown at sip server/registered clients

[hacktek]

I am using brekeke pbx. I will try that and see how it goes.

Thanks