Best Configuration For SIP Sever and MTPBX

1. Brekeke Product Name and version: SIP Server & MTPBX

2. Java version: N/A

3. OS type and the version: Win2008

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

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :
Pattern5/9
6. Your problem: Hi, I am new to Brekeke and RaduisCAT. Currently we have been running a hosted PBX solution using an Ingate Siparator as my SBC/Stun Server infront of my Multitenant Asterisk network. We are using a combination of PRI's as well a SIP Trunking from a single provider. We are transitioning over to Brekeke and RaduisCAT to replace this solution. I was told that there will be no need for the ingate to stay in place as Brekeke handles most Nating issues and endpoints behind the same natted firewall will keep the media local when extension to extension dialing is done and no call recording needs to be done. My question is the company providing the setup is suggesting placing my SIP Server and MTPBX directly on the internet and using the windows firewall. I have a juniper router that I would like to use and do port forwarding. Are there any suggestions out there that could help me make our decision?
Thanks,
Chris

>> ..and endpoints behind the same natted firewall will keep the media local when extension to extension dialing is done

It is depending on settings...
If you use Brekeke PBX, all of media packets will be relayed through the PBX.


>> I have a juniper router that I would like to use and do port forwarding.

http://wiki.brekeke.com/wiki/Set-port-forwarding

We are still working on the setup of this product and looking for some direction. Here is what we have

RaduisCAT --- SIPServer --- MTPBX --- Sangoma NETBorder (PRI)

- All 4 servers are on the same LAN.
- We have a combination of SIP Trunks from a single provider and PRI's
- We are providing a hosted PBX/SIP Trunking solution.
- Hosted PBX Endpoints are various Polycom and Aastra Phones
- We provide SIP Trunks to SIP enabled PBX's

We require our customers to have access to both, Our SIP Trunks and Sangoma NETBorder with the PRI's.

My question is, what is the best setup for this?

I was thinking of having everything regiter to the SIP Server.
1) MTPBX
2) NETBorder (PRI)
3) SIP Trunks from Provider
4) Customer endpoints (IP Phones)
5) Customer SIP Trunks

The SIPServer would pass registrations back to MTPBX for the endpoints. It would pass customer SIP Trnks to the NETBorder for access to the PRI's. It would manage the SIP Trunks from our provider as well as DID's?

Any suggestions on a setup would be appreciated. Right now for testing my endpoints a re registering directly to MTPBX and a SIP Trunk is created from MTPBX to the NETBorder. This senario works, but the issue is the SIP Trunking. Since Netborder only allows 1 connection, I cannot have both MTPBX and SIPServer registered to it.

Thanks,
Chris A

What "SIPServer" is it?
Is it Asterisk? or is it Brekeke SIP Server?

It is Brekeke Sip Server.

The only Asterisk will be at some customer sites that we want to deliver SIP Trunking to.

Thanks,
Chris A