March 7th, 2008 Dave Posted in PBX, SIP No Comments »
August 22nd, 2007 Dave Posted in Asterisk, PBX, SIP, VOIP 2 Comments »
Asterisk’s userbase has gotten even bigger now that Genesys Telecommunications Laboratories, an Alcatel-Lucent company (PARIS: ALU) and (NYSE: ALU), has announced its decision to support the Asterisk® open source IP PBX platform. According to the president and SEO of Genesys, Wes Hayden, they decided to formally support Asterisk to be able to meet customer demands. Furthermore he stated that, “The increased importance of SIP and standards-based technology has paved the way for using open source for contact centres. We have reached a point where Asterisk and SIP are mature, reliable and proven technologies.”
August 10th, 2007 Dave Posted in Asterisk, Dialplan, IAX, SIP, Voicemail 1 Comment »
Asterisk 1.6 is starting to shape up with some features of Asterisk 1.2 and 1.4 already successfully merged and new dialplan functions already in place. Listed below is a summary of latest update in the functionality CHANGES file of Asterisk.
August 9th, 2007 Dave Posted in SIP No Comments »
SIPNET, a Russian SIP provider, now provides unlimited free calls to to Moscow and St. Petersburg. This is welcome news to people all over the world who have relatives, friends and businesses in the two cities. As of the moment though calls can only be made to certain areas with the following area codes: 495, 499, and 812. Despite the limited coverage the service offered will still affect many since the three area codes already cover a substantial part of Moscow and St Petersburg.
June 4th, 2007 Dave Posted in Asterisk, PBX, SIP, VOIP 1 Comment »
Here’s some news for all you Asterisk lovers out there. Asterisk’s parent company, Digium, just entered into a partnership with Vyatta, a company that specializes in open source routing, firewall and VPN solutions. It’s a collaboration that could have a big impact in the world of open source networking.
February 6th, 2007 Dave Posted in Asterisk, SIP No Comments »
Last time, I wrote about this free service that lets you assign a free US number (with Washington area code) to your Asterisk box via SIP. This time, here’s a service you can use to assign a UK number to your box: CallUK.
February 1st, 2007 Dave Posted in Asterisk, SIP 2 Comments »
Free inbound DID (direct international dialling) numbers can be a very cheap way to assign incoming numbers to your Asterisk PBX. For instance, your company or office can be located offshore, but you can still assign a number from the US, UK or any other country where such services exist.
October 1st, 2006 Dan Posted in Basics, SIP 1 Comment »
My last post has generated some great comments (as I hoped it would) from those who have a lot more experience using Wireshark to view the network traffic from a SIP call. We’re going over these in more detail on the * forums, but one smart commenter brought a paper to my attention that will be great research material. Read the rest of this entry »
September 16th, 2006 Dan Posted in Basics, SIP 9 Comments »
This time we will install a network protocol analyzer to watch the traffic on our LAN from initiating and connecting a SIP call.
September 15th, 2006 admin Posted in SIP 1 Comment »
Have you ever wondered why long distance calls cost so much? In part the reason is because telephone lines cost so much. When driving, you might occasionally see a telephone crew maintaining a telephone line, but what you may have never considered is that there are literally thousands of individuals working around the clock to maintain our telephone lines.
© 2007 Asterisk VOIP Tips | Powered by WordPress | Theme originally by Bob, heavily tweaked.