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I've just gotten off a skype conference call and it pisses me off that the quality of skype is higher than my asterisk calls. Is there such a thing as a super high bandwidth codec? In a situation that you have the bandwidth to share is there something that I can use for important calls when the situation warrants it? TIA, Dean -------------- next part -------------- An

Hi, I'd be interested in comments from any users of the vonage service in the UK? http://www.vonage.co.uk is the website. Where are the servers located, traceroute would be useful. What is the general reliability like? Thanks Mike

Hi all, I am building an Asterisk PBX with voicemail and music on hold functions. An ISDN BRI line will also be available and G.729 IP-phones will be used. Are there patents rights applicable to France? Which licence could I use and how many ones are required (only one per phone or also for voicemail and MOH)? Regards Amaury -------------- next part -------------- An HTML

Skype is very succesfsfull and get more and more users, ... we can ignore them, accept them or do something,... My suggestion is that we try to do something, ... If we would peer to each other, than we get soon also a great amount of users together, and than our service becomes more valuable, ... Let's discuss advantages and disadvantages! bye Ronald -- Ronald Wiplinger (CEO of

I was wondering if anyone has put together a comprehensive list (that is reasonably maintained) that lists country codes, landline numbers, mobile numbers, etc. The particular requirement is for a dialplan to know what is going to be charged to whom. For example, mobile and landline rates are the same in the US the US has a unified numbering plan of 1NXXNXXXXXX, while the UK has: 441xxx

Hello everybody, for all of you that have searched for a real fax solution, look no further. We now have T38 faxing. Please contact me for more information. Thanks Michael D. Schelin ShellTel 626-814-2354

Hello, I have the following situation: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number

Good news for service providers in my opinion, Ebay will likely start alienating Skype users like they did with Paypal users. "SkypeSucks.com" domain already taken, shucks.... http://www.kesq.com/Global/story.asp?S=3837895 -- Cory J Andrews Partner / Purchasing +++++++++++++++ VOIPSupply.com - Everything you need for VOIP 454 Sonwil Drive Buffalo, NY 14225 +++++++++++++++ tf voice