Toktumi – New VOIP App

Toktumi

They say that when a lot of people start jumping into the bandwagon, then you can consider the activity a success. Take mobile phones, for example. In the beginning, only the relatively rich and important people had to have them. After a while, even the average person used mobile phones. Today, almost everyone carries a mobile phone around. More so, more and more businesses are venturing into this industry.

The same thing could be said of VOIP. In the beginning, there were only a handful of VOIP providers. More so, there were even less people who knew about VOIP and used it on a regular basis. Well, anyone who has been following the goings on in the world of VOIP would know that it has become a success and seems to continue on this path in the unforeseeable future.

Furthermore, if the emergence of new VOIP apps and companies is any indication, then we can declare for sure that VOIP is here to stay. Here’s a new addition to the plethora of VOIP apps available – Toktumi. This app is developed by a “who’s who” of VOIP – Peter Sisson – and is basically a phone service that is PC-based. Toktumi is not that new, if you think about it, as it has been in operation since 2006, albeit through a private beta only. Now, however, consumers everywhere can have a taste of what Toktumi has to offer for free. Of course, it is only free if you call other Toktumi clients. For other phones and other VOIP service calls, charges apply.

So what does Toktumi have to offer? Why should VOIP enthusiasts even think of trying it out when they are already using a great VOIP app? For one, Toktumi is supposed to be a breeze when it comes to installation. They are actually targeting small businesses for their main market and their 5-minute installation should present no problems even for the “un-techie.” Some features of Toktumi include conference calls, assigned phone IDs, inbound calling (but of course!), voicemail, and auto-attendant forwarding.

Rafe Needleman at Webware says that Toktumi is just like Skype – but for grown ups. So what does he mean? He didn’t really expound on this on his article but sums it up as being a treasure because it “solves a real need” without bleeding the customers dry. That makes sense, doesn’t it?

Asterisk on the iPhone?

iphone asterisk
I love Apple – what can I say? I know some of you may not agree with me but here’s a piece of news worth sharing. For all those iPhone fans out there, Asterisk Voicemail for the iPhone is coming out very soon. I ran across a developer’s announcement about his project on ChrisCarey.com. This is what he has to tell you:

Asterisk Voicemail for iPhone allows you to check your voicemail messages on your house or business line from your iPhone. You can think of it as “Visual Voicemail”, but for your Asterisk PBX numbers instead of your AT&T cell number. The technology behind it is Asterisk (The Open-Source PBX), with iUI, Joe Hewitt’s UI interface for iPhone. This software can be installed on any Asterisk server (though you will want to use one that is available via the Internet) and will allow you to check messages in multiple folders, listen to messages, delete messages, move messages, and change voicemail settings – all from your iPhone.

Contact me with any questions or comments.

This software is unreleased. Most of the features are fully functional, but I need to clean up certain portions of the code before releasing it in order to avoid public ridicule. This software will be released under the GPL or some other free license.

I know, it’s more of a teaser right now, but hey, with something like this, all you need is a teaser – for now. I am sure that hordes of iPhone fans using VOIP are going to be thrilled about this. In fact, I have seen several blogs and web sites already featuring Chris Carey’s announcement. It is not a surprise, really. For those who are already using Asterisk and have iPhones (I am not sure about the number of these people), this is something to look forward to.

So what is the potential of this development? I suppose you can describe it in one word –HUGE. Imagine being able to access your voicemail messages anywhere. With today’s trend of everything going mobile, this is the perfect solution to the traveling individual’s needs. Chris Carey, you rock!

Asterisk 1.6 Update

Asterisk 1.6 is starting to shape up with some features of Asterisk 1.2 and 1.4 already successfully merged and new dialplan functions already in place. Listed below is a summary of latest update in the functionality CHANGES file of Asterisk.

AMI – The manager (TCP/TLS/HTTP)
Added functionalities include: TLS support for the manager interface and HTTP server, URI redirect option for the built-in HTTP server, GetConfigJSON (action that returns the contents of an Asterisk configuration file in JSON format), a “Bridge” action which allows you to bridge any two channels that are currently active on the system, and a “ListAllVoicemailUsers” action that allows you to get a list of all the voicemail users setup.

Dialplan functions
Added functionalities include: a DEVSTATE() dialplan function that allows retrieval of any device state in the dialplan, a new option to Dial() for telling IP phones not to count the call as “missed” when dial times out and cancels, LOCK(), TRYLOCK(), and UNLOCK().

CLI Changes
Added functionalities include a ‘core show channels count’ CLI command and the ability to set the core debug and verbose values on a per-file basis.

SIP changes

Added functionalities include: an improved NAT and STUN support, new way of matching incoming requests, “busy-level” for setting a level of calls where asterisk reports a device as busy, new realtime family called “sipregs” (used to store SIP registration data), more support for T.140 realtime text in SIP/RTP, new variables for call transfers, and a new header that is displayed for cancelled calls answered by another phone.

IAX2 changes
Added functionalities include: trunkmaxsize configuration option to chan_iax2, srvlookup option to iax.conf, and support for OSP.

DUNDi changes
Added functionalities include: the ability to specify arguments to the Dial application when using the DUNDi switch in the dialplan, ability to set weights for responses dynamically, dialplan functions (DUNDIQUERY and DUNDIRESULT) that will allow you to initiate a DUNDi query from the dialplan and find out how many results there are as well as access each one.

ENUM changes
Added functionalities are two new dialplan functions (ENUMQUERY and ENUMRESULT) that will allow you to initiate an ENUM lookup from the dialplan access the results without doing multiple DNS queries.

Voicemail Changes
Added functionalities include:  the ability to customize which sound files are used for some of the prompts within the Voicemail application, the ability for the “voicemail show users” CLI command, “tw” language support, support for storage of greetings, and the ability to customize forward, reverse, stop, and pause keys for message playback.

More of the latest updates on my next post. To view the detailed updates including modified and removed functionalities you can check out their changes file.

Buying a VoIP Gateway? Here are 10 Things to Consider

We previously wrote a brief introduction about VoIP gateways recently, and from there we learned some basic concepts about Gateways, which handle the task of transferring voice or data traffic from a circuit-switched telephone network to an IP-based network. We also talked in passing about some of the factors you have to consider when choosing a Gateway for your business or company.

Now here’s a ten-point guide that can help you make that decision on what exactly you need.

1. Cost. The question of cost should be the first thing you should address. For one, your company might have a budget for such equipment, and you might be in the market for different brands or product sets with comparable features. Part of this would be the cost of setting up, maintenance, and even support from your vendor.

2. Hardware vs. Software. Hardware-based VoIP gateways are perceived to be more reliable and secure. Further, these run on their own processors, and hence do not drain computing resources away from existing computers. Software-based gateways, meanwhile cost less, and are easier to update, upgrade or modify as the need arises. It’s therefore a tradeoff between reliability and flexibility.

3. Chassis size. If you’re planning to install a hardware-based gateway, the chassis size is usually indicative of the gateway’s packet processing capacity. Slow processing means poor voice quality and low capacity. This will not apply if you’re opting for a software-based gateway. In that case, it’s the processing capabilities of the computer you will be using that will be important.

4. Capacity. This is in terms of number of simultaneous VoIP calls the gateway can handle without being overworked. The gateway should be able to cope with the regular traffic of your network, and should also accommodate traffic spikes and expansion (if your organization is growing, for instance).

5. Foreign exchange office (FXO) ports. The primary function of VoIP gateways is to convert signals from the public switched telephone network into IP packets. For analog PSTN lines, FXO ports are needed. Small businesses and remote offices would usually need at least four FXO ports.

(continued …)

Paris Hilton Hacks Voicemail Using Asterisk

Sounds like an interesting title, right? Well, apparently it’s an issue that just brought the world of Hollywod closer to geekdom. However, Paris Hilton did not exactly have to know the inner workings of Asterisk to conduct her “hack attack.”

Last August, Paris allegedly spoofed fellow celeb Lindsay Lohan’s caller ID to retrieve the latter’s voice mail. She was also accused of hacking into about 50 other accounts. Or at the very least, someone who had used Paris Hilton’s name was being on a run of mischief.

It seems some mobile networks these days do not bother to ask customers to key in passwords or PINs to retrieve voicemail. All that a user needs to do is to call the network from his/her mobile phone, and the network will connect to the appropriate voice mailbox based on the number that registers on caller ID.

However, inexpensive prepaid services like SpoofCard.com lets users call a toll-free number, key in whatever number they want to appear on caller ID, and dial the desired destination number. This means users can also spoof those numbers, and retrieve the voicemail as if they were the owner of that number.

SpoofCard.com uses Asterisk to run its telephone network. The fake caller ID service provider says there are legitimate uses for its services. For instance, this could be very useful for employees who need to dial into their company telephone networks, but could only gain access if they call in from a number that is recognized being from within the corporate network.

Still, this just goes to show that Asterisk’s outbound caller ID can be misused.

[via TMC]

Back to the Basics – VoiceMail

The voicemail protocol is quite a bit of fun and super easy to use. Let’s just jump right into it.

From the Voip-Info respository, here is how the command will go in your dialplan.

VoiceMail([flags]boxnumber[@context][&boxnumber2[@context]][&boxnumber3])

You will insert the VoiceMail command after the “exten =>” (as per usual)  then unleash all the excitement at the end. Now, there are so many things that you can do with this command that I won’t really get into a lot of it. There is also so much more once you get into the actual voicemail.conf, which we will touch on later.

Where it says “flags” you can place a couple of things. If you put “s” in there, it will skip the usual “Please leave your message after the tone” recording. If you use “u” it will say the user is unavailable. Then “b” is short for busy. You could come up with a script to change this realtime I’m sure.

After your flags, you will put the number of the VM box that you are looking to send a message to. If you leave this portion blank, then you will get a prompt asking for the box number. Simple stuff.

As for the context section, I’ve NEVER had to use that, so we won’t really bother with it.

What about Voicemail.conf you ask? Well that’s another whole article right there. We will touch base really quickly though. Let’s say that you are just trying to set up a basic voicemail box within your Asterierk PBX that doesn’t have outgoing call capabilities. Let’s save as much money as we can :)

Just for ease of use (and because I don’t need to create large networks) I keep all of my voicemail box information under the default context. You can find this under [default] just like you would in the dialplan. Here is a short example of what your box information would look like.

100 => 321,Users Name,email@address.com,pager@address.com,saycid=yes|review=yes|operator=yes

Let’s tear it apart now.

100 – First it the box number, 100. Nothing more to say here.

321 – This is the password for the email box. It can be as long or short as you feel like.

Users Name – This is for CID. What you put here will show up as the callers name.

email@address.com – This one is easy… the email address of the user.

pager@address.com -  This one will go to their cell/pager.

saycid=yes – When the user calls up and checks his VMB, then this will enable the box to read off the callers ID so if they did not leave a number on the message, they will be able to identify the caller.

review=yes -  If the user wants to review the messages again, they can.

operator=yes – This enables the option menu after the messages are reviewed.

Well guys, that’s all we are going to talk about on voicemail today, but check back soon because we will be doing another article on voicemail.conf which is going to be a very informative one. Check back!

Using Asterisk In Your Home – Using Voicemail (Basics)

Today we are going to discuss the basics of using Asterisk as a voicemail server. Voicemail is an extremely easy to learn and use command in Asterisk.

First of all, we need to add it to our dialplan. If you wanted to use it as only voicemail, you can set it to something like this.

exten => 33,1,Answer()
exten => 33,2,Wait(2)
exten => 33,3,Voicemail(225)

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