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sip:providerCE 2.2 – a quick preview

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 reasons for them to do so:

It’s mostly B2B software, and users of such software are probably used to deal with bad interface design

That reason is just pure ignorance towards potential customers which are going to use your product day-by-day.

It’s not the interface, stupid! – which means that the meat of a product is not in the interface, but the back-end systems.

This is a valid reason to some extend. However, day-by-day operations with your platform is what you’re going to provide to your customer. It really doesn’t matter how great your backend *buzzword* storage engine is performing with your *buzzword* frontend implemented in *buzzword* programming language. What matters is how easily the regular tasks can be performed using your solution.

Based on our customer feedback, we’ve improved our Admin Panel quite massively, have a look at it at our Facebook Profile.

Backend System Changes:

Peer Authentication

The feature which has been requested most was to provide a way for peering authentication. This is in fact not an easy task, but we’re laying grounds in the 2.2 release in order to provide this feature. We’ll introduce a new call-flow design, putting SEMS transparently in between the signaling towards your subscribers. This enables to hard-code authentication towards specific peers. We’ll work together with the SEMS developers to make the integration of peer authentication more smooth in the future, allowing to configure authentication credentials per peer in the admin panel.

3 responses to “sip:providerCE 2.2 – a quick preview”

  1. Michael Magan says:

    Hopefully it will be more friendly with respect to making outbound calls. No registration to sip providers is possible and the way you call the ruling aspect of the system needs to be changed to a more understandable way. Callee is not something you would use it is simple outbound and inbound routes. For the life of me i can’t make and outbound call through any of my suppliers. They can be sip (asterisk) or digit sent authenticated. it is impossible. so if you cannot use the pstn what good is it for?. my 2 cents.

  2. Andreas Granig says:

    If you’ve issues with the configuration, subscribing to and asking for help there (describe what you’ve put where, what you expected and what the result was) will most likely solve your issue.

  3. Dejan says:

    The regexp routing is actually quite good thing to have once you get used to it.
    Within few hours I have been able to set up the base system. Now it is the time for adding the features that couldn’t be added if I have had to develop the whole system from scratch. Great stuff you have here Andreas!!!

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