It’s been a while since I paid any attention to OnSIP. Since they are my primary ITSP this very fact is probably “a good thing.” It means that we haven’t had any issues. However, they have been making a few changes in their service. These changes bear closer examination.
™ Martha Stewart Omnimedia.
1. New OnSIP User Portal Looks Great!
A few weeks back Junction Networks launched the long awaited user portal for their OnSIP hosted IP-PBX service. You can find it at My.OnSIP.Com. A long time OnSIP user I was in the beta program, but since I could not roll-out the portal to my co-workers while it was in beta I didn’t have much opportunity to evaluate it in depth during the beta program. Now that the portal is broadly launched I’m introducing it to any of our staff who have OnSIP connected phones. That’s about 16 users mostly in the US, but a few in the UK, France & the Middle East.
The portal offers some nice capabilities. It’s essentially a busy-lamp-field for a distributed company. At a glance I can tell if any of our staff are on the phone. I can initiate a call, or even transfer an ongoing call to another extension from the web interface. There’s also a simple IM mechanism.
I’d like to see the status display extended to support actions involving extensions that are resources external to our OnSIP account. We have a large number of extensions programmed to our OnSIP account, where each is actually a DID occurring on our PBX at HQ. It’d be great to have all these speed dials available via the web GUI.
With a little poking around I was able to determine that Junction Networks is using Sprout Builder as the underlying tool set for creating this new user portal. Sprout Builder is a hosted development environment for Flash based applications. It includes a hosted design environment as well as hosting for the completed Flash files. I was involved in their beta program last year.
The very fact that the portal is implemented in Flash is kinda curious. That does explain the one annoyance I have about the logon process. The logon page does not remember your SIP registration name and password. This is a problem common with Flash GUI implementations. OnSIP is very rigorous about issuing complex passwords to ensure a secure system. You’re probably not going to remember that password, so keep the email reminder handy until they work out how to let your browser remember the logon.
2. Provisioning Services
I was pleasantly surprised to find that OnSIP has also added a basic ability to provision Polycom phones. This shows up as a new tool “phone” tool under the “resources” menu of the admin portal.
Diving into the menu further it’s yields a fairly straightforward tool for assigning phone parameters based upon the phones MAC address, typical of the Polycom provisioning scheme. The menu provided support for various Polycom models, a series of speed-dials, common company directory, NAT keep-alive, variable time zones and multiple call appearances per line key. The company directory looks interesting. It must build that dynamically based upon the users in the OnSIP PBX.
I’ve yet to try this myself as I already have an FTP server setup to handle provisioning all the phones that I oversee. I’ll be giving it a try eventually. It could simplify my life, which is always a noble goal.
I wonder how they will deal with significant transitions between software versions?
3. Policy Changes Re: Free Conference Services
A few days back I received an email from Junction Networks outlining a new policy toward free calling conference services. Here’s a snippet:
Starting Friday November 13, 2009, calls to seemingly free conference services and other reverse billing services will be charged at $0.50 per minute.
As of today, the affected rate centers are:
(712) 432-xxxx and (712) 338-xxxx
Calls to these rate centers are 20 times more expensive than a ‘normal’ call. Junction Networks cannot afford to subsidize these services and maintain our competitive pricing. We have only two options available to us – block calls to those numbers or charge the true market rate. We have chosen the latter.
I completely understand this position. The change in policy doesn’t effect us at all since we elected to pay the $20/mo that they charge to add a private conference bridge to our OnSIP service. The private bridge is available by SIP URI or via a DID. Calls via the SIP URI are free but access via the PSTN is billed at their nominal rate, which is exactly the same rate that we were long ago paying to access one of those pseudo-free conference services.
Perhaps this matter of policy will place a little more emphasis on their conference bridge. I’d like to see something of a GUI offered, not unlike that which we enjoy during the VUC calls hosted on ZIPDX. We are not so concerned about scheduling conferences, as managing ongoing calls. We’d like to have a host console capable of muting noisy callers, reaching out to late participants, dropping callers, etc.
Oh yeah, we’d like to have G.722-based wideband conferencing, too!
It’s great to see that the company isn’t resting on its laurels.