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[Voxilla VoIP Forum - Linksys (Sipura) VoIP Support Forum] SPA3102 incoming through PSTN outgoing with Asterisk: Voxilla has been covering VoIP and communications since 2003, and we are always asked for recommendations. Of course, there isn't a single answer that meets everyone's needs, but one of the following Voxilla Recommendations will meet yours.

[Weave's ole blog] Using the Linksys SPA-3102 as a VOIP ATA: Also, some VOIP providers *do* handle 911 calls, so whatever you subscribe to, be sure to find out ahead of time if they support 911 and if they do, ensure your address is correct with them so it can be transmitted to the 911 center if dialed. Just don’t assume they will do the right thing with 911 (or whatever the emergency number is in your own country).

[3CX VOIP, SIP & Phone System blog] 3CX VOIP, SIP & Phone System blog » Connecting your Home Phone to 3CX: As many of you know, using remote phones without VPN is all over the 3CX forums.  It’s one of the hottest topics out there.  NAT and firewalls often create complexities in making this work easily and end up requiring router and firewall modifcations.  Since this is really the same thing (with the additional complication of not using a true IP phone but rather an ATA adapter with my home phone) I figured I was out for some trouble on this.  But, hey, I’m a gluten for punishment!

[Voxilla VoIP Forum] How to configure 2 SPA3102 units for a Point to Point PSTN Link: SPA3102 is supporting a standard SLT phone on its Line1 port, the two SPA's are network connected and appropriately programmed (like a ring-down or hotline), so now the standard SLT phone behaves as an extension to the PBX. All well and good with one exception : a hook flash on the SLT phone is not translated via IP to the PSTN port of the Home SPA thence to the PBX extension port.

[KernelCrash] Asterisk PBX and the SPA 3102 | KernelCrash: Line Enable: yesSIP Port: 5061Proxy: [address of my asterisk box]Register: yesDisplay Name: pstnUser ID: pstnPassword: [your password you setup on Asterisk]Dial Plan 2: (S0<:s>)VOIP-To-PSTN Gateway Enable: yesPSTN-To-VOIP Gateway Enable: yesPSTN CID For VOIP CID: yesPSTN Caller Default DP: 2 (ie. use the dial plan 2 setup earlier)

[Voxilla VoIP Forum - Linksys (Sipura) VoIP Support Forum] Need Asterisk and Phonegmone help.: First I would like to be able to place calls from the Phonegnome box using it's FXS (locally attached phone) port, and have it use the Asterisk outbound routes and trunks I have setup (link it was simply another SIP extension). The other think I hope to do is to use the Phonegnome's FXO (the PSTN phone line dial tone) port as an outgoing trunk to my Asterisk system.

Reflected tags on Technorati: Blog, Voip Forum, VoIPBlogging