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Just curious: has anyone seen or heard about a USB-based T1/E1 interface device? I've seen some serious T1/E1 testing equipment that is USB-based, but I was wondering if there was something more generic, like a Zaptel-ish T1/E1 that used USB instead of PCI/PCIx. Thanks! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL:

I setup sip realtime. Is it possible to use a type of friend? User and Peer seem to work fine. -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com

Greetings list, Wondering if anyone's come across this before. I've configured a couple of our servers with a "privatedundi" context to allow calls to still flow between extensions even if they're registered to different servers . The DUNDi lookups seem to work fine, evidenced by the following on the originating server: -- Called

With all of the recent talk on the list about DUNDi, I have a question. From the outset it appears that SER is often used for high availability solutions and as a tool for almost clustering Asterisk boxes behind it. It appears to me that DUNDi is providing a lot of this as well. Now I know DUNDi is not an application by itself to proxy SIP requests but can I hear any information out there that

I am new at this. I have read "Asterisk: The Future of Telephony" and have installed AsteriskNOW (beta 4, due to the dual processor problem in beta 5). The GUI interface does not seem to provide the capability that I need, although I have modified the *.conf files to successfully create what I need. Given this, I would like to install Asterisk on a distro. I am most familiar with

Hello people. I would like to know if someone knows about any applet to include in a web page to start calls. What I am looking for is something that doesn't allow users to change numbers, or any other option, so I can include it in my web page and force them to call to me and no one else. I have tried JIAXClient, but it allows people to call anywhere, and what I want is just a configurable

I just recompiled my version from this morning's CVS Head. My systems voice files (voicemail, time etc) were playing nicely. Until that is I added an extension and now the files won't play. Worse than that, * thinks the files have played and goes to the next step in the dial plan. What gives? -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

I'm having some troubles with my * machine, when i place a call on hold the callee doesn't hear any MOH and the call is dropped because of lack of RTP. I also don't see * starting MOH on the SIP channel the callee is on (moh class is defined, there are MP3 files and mpg123 is active). I'm using * 1.0.6 right now with Cisco 7940's, i can see * recieving a SIP invite with