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I need to pick all the Asterisk and Cisco People a little. My company has a Cisco Call Manager 3.3, configured via h323 gateways. I have remote users that I want to place a SIP Server on the external WAN and be able to connect their phones to the system and be able to get calls and call people in the office going through the Cisco Call Manager and the h323 router. My only problem is that Cisco

Can anyone tell me if there is a Calling Card Platform in which I can use in conjuction with Asterisk that can give me Authentication via the caller id of the user. I don't want a PIN based Calling Card system, but the software to be able to recognize the caller ID information and authenticate the users and allow them to call through the asterisk box. Thanks.

I have a question on Asterisk and whether it will work with the following design. Install ASTERISK on the external side of the Network. Purchase an AudioCodes 4/8 port Analog Fx0 gateway. So far everything seems straight forward. Here is the twist. The company currently has Cisco Call Manager 3.3 which does not support SIP Trunking. But it does have a VG248. I would like to place 4 lines

Has anyone tested a grandstream 488 FXO gateway on an Asterisk machine? I read that the 488 has a FXO port on it, can I use the grandstream 488 to pass traffic to the pstn from Asterisk. I would use this at home to pass traffic into a foreign country's PSTN over the internet. Thanks.

I am thinking of building an Asterisk PBX, and had a question on a piece of hardware support. I want to include a 4 port PCI 10/100 Switch router card. For those not familiar it's a PCI card that acts as a switch. My question is would I be able to configure those 4 ports to support sip phones plugged in directly to the asterisk box instead of a switch. Thanks in advance

Cool New Website For everyone to see! I think they are using a specially programmed version of Asterisk to do this. www.dialaway4free.com

We are in the process of building out www.dialaway4free.com, a free world wide calling service. I am writing RFQ's for hardware, since we are going to use asterisk as our call processor. I was wondering what is the best server platform to use that will support digium cards and handle sip termination for both clients and service providers. Also should I go with the open source of asterisk as