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Hello! I would like to get connected with a VoIP provider for home. At some point, I'm sure I will be connecting to it via an Asterisk box, but for now, I will be using whatever hardware they provide. What recomendations do you in the Asterisk community have for a reliable VoIP service that will hopefully interoperate with Asterisk? A company that is actually willing to work with an

Before I "reinvent-the-wheel", does any one know of PHP based interface to the CDR table? If not, I will get started writing one. Thanks, Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030624/d4cb6911/attachment.htm

How do I define the default "No Caller ID" Caller ID value? When a call comes in without Caller ID, the display currently reads "From asterisk asterisk". Can I specify this to read "No Caller ID". I tried adding database put cidname "No Caller ID", but that doesn't work since it excepts a number. Thanks, Marcus -------------- next part --------------

Just a quick note to people looking for SIP firmware images for Cisco phones: To access these files from Cisco's website, you need to have a Service Contract (SmartNet) on at least on of your phones. I though a contract was several hundred dollars, but it is way cheaper! Their lowest level, 1 year Next Business Day 8x5 SmartNet program, costs around $70.00, and gives you unlimited access to

Hello, I am working on a project for a client of mine, that involves streaming an incoming phone call. I have a lot of experience with Linux, but something that I have never attempted is using sound under Linux, as all my machines have been servers. Here is the plan, is it doable? The client (a minor league baseball team) calls in to a iconnecthere.com phone number, and Asterisk answers (this

How does a user exit the directory application? Say he can't find the person that he is looking for and wants to return the main menu, how would I configure 0 to act this way? Thanks, Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031113/f5516f7d/attachment.htm

I upgraded to the latest CVS yesterday (and this morning again), and whenever I execute the reload command Asterisk seems to hang. While the current calls aren't dropped, no new calls can be made. The CLI isn't responding properly either. The only way to get going again is to exit the CLI and stop Asterisk and start again. Any comments? Thanks, Marcus