search for: shido

...ly with our analog telephone system. On the adtran router there was a network jack which I just connected directly to our hubs then pointed all of our computers at it's ip address 193.169.0.1 effectively routing our data properly and allowing internet access. Thanks to a lot of help (thank you Shido/Twisted) we now have the machine running with the T1 line plugged directly into it. The data/internet is all running properly and the driver that pulls the data out of the T1 channels (hdlc) is loaded properly and running. It looks just like an ethernet card to the Operating System. The IP address...

...if this is true, then * VoIP players like Nufone with their 80+ * server farms.....are using what E1/T1/DS3/etc * compatible hardware? are Digium and VoiceTronix the only * compatible hardware? I googled a lot and found "will work with opensource apps" but do not explicitly say Asterisk. (Shido can you shed some light on this?) Since I am an end user/Unix sysadmin/VoIP learner I think I will ask the forum to give me the task of doing a wiki or update existing wikis and voip-info.org (when we all agree of course). I bought the * book from signate. It's very good but lack some things....

Is digium gear AUSTEL approved? $25,000 fine for using equpment not approved by the Aussie gov't. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030713/d8df8091/attachment.htm

...e popular the product is, the more remote the possibility of the creator making any money from it. Take Digium. The more experts on Asterisk pop-up, the less demand is for Digium services. In fact, having tried Asterisk support from Digium and others, I think the best Asterisk people --like Jeremy, Shido and swk286-- are somewhere else. So the question is: how do we make sure that the creator of the product makes even one dollar from every copy put in use of his creation? The answer is: there is no answer. There is where Microsoft wins. Additionally, Microsoft support services do know their product...

Semi-offtopic, Anyone know of voip providers who can provide tollfree number service? E.g. route 800-xxx numbers to our * ? Even better if they are familiar with * or can speak IAX ... -Dan

I have a linejack and a phone jack in my asterisk server working well between the SIP phones and the phonejack. what I cannot get to work is the outbound linejack Phone/phone0 trunk line? how can I get a SIP or Phone/phone1 phonejack phone to dial 9 then outside number and pickup Phone/phone0 and dial it? right now it accepts a 95551212 but busy's on the last digit 2. no outside dial.

I just received my budgetone 100 and the difference between using sjphone and being at the mercy of my cheap Muse XL PCI sound card is like comparing night and day. Now that I have some real hardware I have my budgetone talking ulaw to my linux box then out to TNN (The Nufone Network) for IAX term... sound quality is great EVEN DURING massive file xfers. I can xfer by using # extensions # with