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toy buy my first cisco 186 but when i read this page http://www.djernes.org/~shawn/ata186.htm i cant find in my dev page some parameters just like " UseSIP " what i need to do to show this parameters Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030626...

...e clone video, IBM 15GB, IBM 60GB, IBM 80GB IDE drives and four fans) Westtel DSL modem, Netgear 10mb hub (outside interfaces), Intel Pro 8-port managed switch (inside interfaces) for 25-30 minutes and that is usually enough for the power company equipment to reset around here. Thanks --- Shawn L. Djernes, N0PEU Djernes.org shawn@djernes.org | sdjernes@telerama.com | sdjernes@earthlink.net http://www.djernes.org PH: (412)-257-3252 519 Washington Ave. #2, Bridgeville, PA, 15017, USA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/aster...

I've installed Asterisk and configured an ATA-186 as described at this link: http://www.djernes.org/~shawn/ata186.htm Unfortunately this guide abruptly ends before it explains how to deal with the sip.conf and extensions.conf files. So I left extensions.conf alone and my sip.conf looks like this: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0...

I really should be doing something better on this beautiful weekend, but I'm trying to save myself some time for later projects by documenting some things that have been particularly troublesome in the past. That being said... I've written up a configuration guide for the Cisco ATA-186, which describes some of the features that are possible to set in the ATA and specifically what

...t, I'd be happy to maintain a page. Of course it would just be another dangling link unless tied in... Useful links I've gathered over the last few weeks: http://www.digium.com/index.php?menu=faq http://www.loligo.com/asterisk http://www.automated.it/guidetoasterisk.htm http://www.djernes.org/~shawn/ata186.htm

I have an ATA-186 in a SIP configuration (following Shawn Djernes how-to), but I get the following error at the asterisk console when I try to call the phone connected to the ATA: ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device Failed to register zone 'United States / North America': No data available Everything works if I remove indication...