It’s been 3 weeks since sip:providerCE v2.2-rc1 came out, and today we’re proud to release the final version of sip:providerCE v2.2! What’s new? For the list of fundamental changes since v2.1, please read the v2.2-rc1 announcement linked above. Since v2.2-rc1, we only fixed bugs and improved the general handling of the SPCE: Sponsored development of…
Musings of a telco vendor about the cloud thing Over the last days, I was working on deploying and testing our open-source VoIP soft-switch SPCE v2.2-rc1 on an Amazon EC2 instance in order to provide a ready-to-run AMI for our community. Coincidently, I came across this tweet from @martingeddes: When you hear a vendor selling…
After over 4 months of development, we proudly present you the official Release Candidate 1 of sip:providerCE v2.2! Based on the feedback from our awesome Community and Customers, and aligned to our long-term Road Map, we’ve implemented some fundamental changes into v2.2 compared to v2.1, and we bring you some desired new features. What’s new…
You want to meet us at Linuxtag 2011, Europe’s most important convention for Open Source Software, taking place from May 11 to May 13 2011 in Berlin? Andreas Granig, CTO of Sipwise, will present the sip:providerCE v2.2 in Hall 7.2b at the Kamailio booth 112 (near to the Mozilla project) on Thursday and Friday. You…
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…