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VoIP solutions could benefit from iPhone 4 - Why bother? Get an Android Open Source phone and the new VoIP features now

Thursday, June 10. 2010

Hi all,

The iPhone 4, which was officially unveiled on the June 7th at a presentation in San Francisco, includes several new features that will benefit VoIP users. These are touted as:

Multi-task when making a VoIP call, so customers can access other applications, such as a calendar, during a conversation.

Built-in noise cancellation to eliminate background sounds when making VoIP calls.

Videoconferencing will also be possible between iPhones via the Face Time feature, allowing remote employees to partake in business meetings.

Longer battery life, while the screen resolution has been significantly improved compared to previous models.

The iPhone 4 will be available in the UK from June 24th.

Wow you may think this is amazing, but all these features have been available on Android phones (Google's Open Source mobile operating system) for ages. I've just got an HTC Desire from O2 and it's the No.1 phone in the world right now. It can do all of above and more, plus the Sipdroid VoIP client is free and works on 3G and WiFi using our VoIP services or anyone elses. Best of all it's free software and just works.

Another good blog post covers why the iPhone can never keep up with Android or Open Source:

"Through a bevy of handset makers, Android can offer a variety of phones that will make it difficult for
Apple to beat with just one hardware release a year. While it is hard to

ever go wrong with an iPhone, Android offers a ton of alternative form

factors, price points and carriers: Sprint (NYSE: S) has released the

first 4G phone on Android; T-Mobile has a new competitive Android phone

with a slide-out keyboard; the HTC Incredible sold by Verizon has been

flying off store shelves; and even Google’s Nexus One still boasts some

of the latest hardware. Not to mention new Android phones from Samsung

and LG (SEO: 066570) coming later this summer."

Google now have figures to show that 100,000 open source Android mobile phones are getting activated daily!

Gavin.

Posted by Gavin Henry in Asterisk, FreeSWITCH, Linux, Open Source, Suretec, VoIP at 11:52 | Comments (0) | Trackbacks (0)

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BT VoIP Interconnect approved for our ITSP called SureVoIP!

Thursday, May 27. 2010

Hi all,

Another quick one, just finished a conference call with BT and our credit check and applications are now all complete and approved. We will have our VoIP interconnect with BT live in month or so!!

Cheers,

Gavin.

Posted by Gavin Henry in Asterisk, FreeSWITCH, Open Source, SIP, Suretec, VoIP at 10:24 | Comments (0) | Trackbacks (0)

Defined tags for this entry: Asterisk

, SIP

, VoIP

All Ofcom number allocations are being approved

Thursday, May 6. 2010

Hi all,

Looks like all of them are coming in except critically low areas, but we'll use sub-allocations from our carriers for these areas.

We'll have around 98% UK coverage for numbers in the end. We haven't even finished applying or done 03/07/08/09 yet!

We also got our Communications Provider ID and Carrier-Pre-Select numbers.

More later,

Gavin.

Posted by Gavin Henry in Asterisk, FreeSWITCH, Open Source, OpenSIPS, SIP, Suretec, VoIP at 21:40 | Comments (0) | Trackbacks (0)

Defined tags for this entry: Asterisk

, SIP

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, ofcom

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First Ofcom number allocations are in!!!

Thursday, April 29. 2010

Another quick update. We've just received notice of our first ever number allocations from Ofcom.

They are "Location Independent Electronic Comm Services" which are used for VoIP accounts and we have:

05600 51XXXX

Thanks,

Gavin.

Posted by Suretec in Asterisk, FreeSWITCH, Open Source, OpenSIPS, SIP, Suretec, VoIP at 11:56 | Comments (0) | Trackbacks (0)

Defined tags for this entry: #asterisk #voip

, SIP

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, sip

, voip

Suretec allocated a Communications Provider Identity ("CUPID") code by Ofcom

Tuesday, April 27. 2010

Exciting news!

Our Communications Provider Identity ("CUPID") code application has been processed by Ofcom and we just received our number of 291.

The start of many good things to come with our own VoIP provider SureVoIP (brand and website being worked on now). We're also listed as an ITSP at ITSPA.

Thanks,

Gavin.

Posted by Suretec in Asterisk, Catalyst, Dojo, FreeSWITCH, Jabber/XMPP, Linux, Open Source, OpenLDAP, OpenSIPS, Perl, PostgreSQL, SIP, Suretec, VoIP at 12:13 | Comments (0) | Trackbacks (0)

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Suretec join The Internet Telephony Services Providers’ Association

Friday, March 26. 2010

Just a quick note to say that our application to join the The Internet Telephony Services Providers’ Association has been approved!

More about all this next month.

Thanks,

Gavin.

Posted by Gavin Henry in SIP, Suretec, VoIP at 13:58 | Comments (0) | Trackbacks (0)

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UC Expo 10th March

Thursday, March 4. 2010

A quick note to say we will be attending UC Expo - The UK's leading Business Enterprise Communications Event.

I know we've been very quiet this year, but there's a good reason for this. Keep your eyes pealed for a big announcement in the next few months about our new VoIP provider (Internet Telephony Service Provider) - SureVoIP (redirects to main site for now).

Gavin.

Posted by Suretec in Asterisk, FreeSWITCH, Linux, Open Source, OpenLDAP, OpenSIPS, SIP, Suretec, VoIP at 09:45 | Comments (0) | Trackbacks (0)

Defined tags for this entry: Asterisk

, Linux

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Yubico key has arrived!

Thursday, December 17. 2009

It's taken us a while to get round to ordering one of these Yubikeys from Yubico, but they do look great and have lots of documentation via the wiki.

We're going to be testing them via the pam module for SSH access (and of course using pam_ldap).

So, users in LDAP^[?] via pam, and a Yubikey, let's see how it goes!

(will be interesting for logging in a SIP handset via Asterisk or FreeSWITCH that needs one too and all the others technologies Suretec use and support (Catalyst, Perl, PostgresSQL, Squid and more....). We'll see....

Posted by Gavin Henry in Asterisk, Catalyst, Jabber/XMPP, Open Source, OpenLDAP, OpenSIPS, Perl, PostgreSQL, SIP, Squid, Suretec, VoIP at 15:39 | Comments (0) | Trackback (1)

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Aberdeen’s Suretec Telecom helps holiday dreams come true with open source call recording

Monday, November 2. 2009

Companies at opposite ends of the UK work together to produce innovative software solution – without ever meeting up

Aberdeen-based Suretec Telecom is in the business of supporting a company that makes holiday dreams come true. Specialising in tailor-made holidays, luxury hotels, unique tours and cruises, DreamTicket luxury holidays is a niche holiday provider to the UK travel market, and it prides itself on operating cost-effectively without ‘middle-men’ so it can keep its prices keen. And it’s based in Surrey.

Suretec Telecom knows how important it is for small and medium-sized businesses to be cost-conscious, and it makes sure that its customers get high tech solutions to telecom questions, at a very competitive price.

‘Dreamticket is a very successful niche holiday business,’ says Gavin Henry, managing director of Suretec Telecom, which was launched in 2009. ‘It grew considerably from its early beginnings to around 11 employees in 2004 – and its telephone system is its lifeblood for providing customers with the right holiday to satisfy their dreams. That’s where we come in.’

Continue reading "Aberdeen’s Suretec Telecom helps holiday dreams come true with open source call recording"

Posted by Suretec in Asterisk, Open Source, Suretec, VoIP at 14:20 | Comments (0) | Trackbacks (0)

Defined tags for this entry: Asterisk

, Linux

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, call centre

, callrecording

, sip

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Suretec now have a merchant account!

Friday, October 16. 2009

It's been a long process, almost 6 weeks, what with the new laws regarding PCI DSS compliance as of the 1st of October 2009, PCI DSS Scans, Business plans, website terms and connditions and many more things we have a merchant account at last!!!

This is mainly for our new Internet Telephony Service Provider called SureVoIP which is currently live with various customers but has no front facing sales and marketing site yet, only the admin and customer portal.

We'll keep you posted with a proper press release in due course.

Thanks,

Gavin.

Posted by Gavin Henry in Asterisk, Catalyst, Dojo, Jabber/XMPP, Linux, Open Source, OpenLDAP, OpenSIPS, Perl, PostgreSQL, SIP, Squid, Suretec, VoIP at 13:16 | Comments (0) | Trackbacks (0)

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OpenSIPS 1.6.0 - a newmajor release is out

Friday, October 16. 2009

After almost 7 months from the last major release (1.5.0), OpenSIPS evolves with a new major release, 1.6.0.

OpenSIPS 1.6.0 comes with several critical improvements (DB area, dialog support), but also with new functionalities
(like memchaced support, B2Bua implementation, virtual DB URLs, STUN server, JSON support, new AAA/RADIUS interface, etc).

A complete compilation with all the additions and improvements for *OpenSIPS 1.6.0 *release is available under:
http://www.opensips.org/Main/Ver160

Many thanks to all the people who got involved in this release (and in the overall OpenSIPS project) and contributed with code, with testing and debuging, with patches or reports, with support on the lists, help with packaging and documentation.
I will avoid listing names, not because they do not deserve it, but simply because it will impossible to list list everybody here and I do not what to be unfair with some of them (because I simply forgot a name or because of the limited space).

But nevertheless, I want to thanks you all for out great job and not in my behalf, but in the behalf of people who will find this piece of software a useful tool.

The full Changelog is available here:
http://opensips.org/pub/opensips/1.6.0/src/ChangeLog

To get the OpenSIPS 1.6.0 version, see :

website page: http://www.opensips.org/Resources/Downloads

SF project:
https://sourceforge.net/project/showfiles.php?group_id=232389&package_id=281827&release_id=670379

Note that for the moment only the source tarballs are available. The packages (debs,rpms, etc) will be generated starting now. If anybody can help in generating packages for different distros or architectures, please let me know and I will upload them on the website.

Best regards,
Bogdan

Posted by Gavin Henry in Asterisk, Open Source, OpenSIPS, Suretec, VoIP at 09:21 | Comments (0) | Trackbacks (0)

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Suretec Telecom finds WARP Enterprise 'rock solid'

Tuesday, October 13. 2009

From WARP Enterprise from PIKA Makes Deploying SMB IP-PBX Applications Even Easier:

Based in Aberdeen, Scotland, Suretec Systems has recently launched a new telecom business to meet the needs of small and medium-sized businesses looking for cost-effective and highly featured telephone systems.

“PIKA’s WARP Enterprise PBX platform is the perfect fit for Suretec’s new Telecom business,” said Suretec managing director Gavin Henry. “It is so flexible in how it can be deployed and configured, yet remains a rock solid platform for any enterprise.

“Our customers have all expressed interest in not only the huge cost savings that can be achieved with open source products, but the feature richness, stability and support,” Henry added. “WARP Enterprise meets the cost targets and also saves us recurring integration expenses over the long run, with stable components that don’t churn on a yearly basis.”

Posted by Suretec in Asterisk, Open Source, SIP, Suretec, VoIP at 22:51 | Comments (0) | Trackbacks (0)

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Create an Asterisk hotline/autodial/PLAR without Asterisk!!!

Tuesday, October 13. 2009

Just a quick one to say that when you think you need Asterisk for a customer request, sometimes you don't!!

A question from a customer yesterday:

Do you know of an IP desk telephone that can be used on a
hotline/autodial/PLAR type basis? Ie pick up one phone and the other
rings, etc.

Tricky bit... no call manager of any sort in-between... just the
phones
direct on the network.

Aastra to the rescue!!! Any of the new 67XXi ranges have a feature either per line or globally like so:

Autodial Settings

Autodial Number
Autodial Timeout

Simply add an IP address in the Autodial Number field for line 1 and when you pick up the handset it dials the other phone on that IP directly. No need to put Asterisk in between or any other system, other than pure IP.

Nice!

P.S. I'm sure most of the business grade IP phones out there can do the same.

Posted by Suretec in Asterisk, Open Source, SIP, Suretec, VoIP at 12:07 | Comments (0) | Trackbacks (0)

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Aberdeen-based Suretec launches Telecom division

Tuesday, October 13. 2009

Open source software the key to cost-effective business phone systems, says Gavin Henry, saving tens of thousands of pounds at the outset

Aberdeen-based Suretec Systems, the successful open source IT systems specialist, has launched a telecom business to meet the needs of small and medium-sized businesses looking for cost-effective and highly featured telephone systems. Oil and gas sector companies, along with professional services, travel and public sector organisations, are all taking an interest in the huge cost savings that can be achieved with open source products – and now for their telecoms systems, too.

Aberdeen and Grampian Chamber of Commerce article - Aberdeen-based Suretec launches Telecom division

OilVoice article - Aberdeen-based Suretec launches Telecom division

Sourcewire press release - Aberdeen-based Suretec launches Telecom division

PublicTechnology.net - Aberdeen-based Suretec launches Telecom division

Continue reading "Aberdeen-based Suretec launches Telecom division"

Posted by Suretec in Asterisk, Open Source, SIP, Suretec, VoIP at 09:06 | Comments (0) | Trackbacks (0)

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