Hello Daniel, You will find the information at http://www.voip-info.org/ and http://oreilly.com/catalog/9780596510480/ (.PDF downloadable from the "Online Book" link) very useful.
The asterisk package by itself should be adequate for SIP/IAX calls. I don't think you need libpri unless you are planning on connecting asterisk to a digital connection such as ISDN or a PRI. You will need Zaptel (for Asterisk versions 1.2,1.4) or DAHDI (Asterisk versions >=1.6) if you choose to install an internal card (OpenVOX, Digium, Sangoma, etc.) I do not know if or how well this will work with a VM. I suggest testing your SIP softphone with the Echo() and/or Playback() dialplan applications before attempting to call another softphone/hardphone/etc. This will allow you to confirm that the one endpoint functions properly before adding more complexity by calling another endpoint. some things that allow you to call a conventional telephone: an ATA with an FXS port an internal card (such as OpenVOX, Digium, Sangoma) with an FXS port call a conventional phone number through the PSTN (below) To connect to the PSTN you can use any of: an ATA with an FXO port (plug an analog phone line into it) internal card with an FXO port (also to plug an analog phone line in) account with an ITSP (there is occasionally discussion on the list about advantages/issues/opinions/and flames with various ITSPs - google "site:lists.digium.com ITSP") An ATA (Analog Telephone Adapter) is basically an analog to digital converter. CISCO/Linksys does manufacture some ATAs, but this is not the only option for an ATA. There are two types of analog ports - FXO, and FXS. You plug a telephone into an FXS port. You plug a conventional phone line into an FXO port. Bad things can happen if you plug a phone line into an FXS port - do not do it. In your example of an ATA with two FXS ports, you will use two conventional telephones - one plugged into each port. The ATA will use a different user account for each of the analog ports. This requires configuring each user account on both the ATA and in Asterisk's sip.conf. The ATA functions simply as a passthrough device: "phone calls for user1 = ring port1" "phone calls for user2 = ring port2" "phone calls from port1 = use user1 account" "phone calls from port2 = user user2 account" The ATA does not decide which port is connected to which extension number. This actually happens in Asterisk's dialplan. A very basic extensions.conf to illustrate different extensions to call each port, and one to call both: [default] exten => 101,1,Dial(SIP/user1) exten => 102,1,Dial(SIP/user2) exten => 103,1,Dial(SIP/user1&SIP/user2) Hope that gets you going in the right direction. http://www.voipsupply.com/ is a good source to see what equipment is generally available to end users. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users