Using Tropo, it’s incredibly easy to add voice applications to your sip:provider CE (SPCE) or PRO system. Let’s assume that we have an SPCE installation serving a subscriber sip:test@sipwise.com, which is mapped to the phone number +4312345. By either calling +4312345 from your mobile or fixed line or calling sip:test@sipwise.com from a SIP client, you’d [...]
You will mainly extend our provisioning and monitoring systems and make the changes visible via the administrative and customer self-care panels, as well as the SOAP and XMLRPC interfaces. The job requires very strong experience in Perl, MySQL and HTML/CSS/Javascript, a strong knowledge of SOAP, XMLRPC and REST and you should be used to Debian GNU/Linux and SVN/GIT. [ Read More → ]
You will implement efficient and secure call routing features in large-scale systems over geographically distributed components. The job requires very deep knowledge of protocols like SIP, RTP and T.38, a strong experience in writing OpenSER/Kamailio configurations and scripting languages and you should be used to Debian GNU/Linux and SVN/GIT. [ Read More → ]
In the current era of a rapidly changing telecommunication world, Asipto and Sipwise are pleased to announce the merge of their IP telephony system offerings in order to strengthen the position in the market and to consolidate the development of their unified communication solutions. With Asipto’s know-how and role in the SIP routing development process, [...]
Since the initial release of the sip:provider CE, version 2.1, we’ve mostly worked on two things regarding improvements towards version 2.2: Usability of the Admin Panel, and how NOT to do it: It seems to be a general rule for soft-switch vendors to keep the administrative panel as ugly as possible. There could be different [...]
SIP is famous for being the protocol to be used for VoIP signaling nowadays. With the (not so simple) SIMPLE extension, it also offers PUB/SUB and messaging capabilities. The plan was obviously to lay grounds for a full-blown communication framework for telephony, instant messaging and presence, as we know it from XMPP. Due to various [...]
Back in December 2010 we were launching the sip:providerCE as an open-source offspring of the sip:provider PRO to bring the Sipwise NGCP technology to a much broader audience. During this month, we got a lot of feedback, and it’s time now to look at the impact for Sipwise and the market. How did it go? [...]
The Erlang-B calculation is used in telephony systems to describe the probability of call loss on a group of circuits without call buffering. It is however not limited to telephone networks, since it describes a probability in a queuing system. Based on this concept, Bill Hammack from the University of Illinois explains why the other [...]
Due to the great success of the VirtualBox image, we’ve created a VMware image of sip:providerCE for quick testing of our free product. Please check out the documentation regarding how to use sip:providerCE virtualization images.
The demand for testing the sip:providerCE on Virtualbox in Windows is actually bigger than expected, and there came up a couple of issues we’re currently working on: The default version when downloading Virtualbox is 3.2.12, which does not support the OVA format (which in fact is just a TAR of the OVF file together with [...]
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