Gone are the days of having your customer enter their phone number into your website for a click-to-call application! Using Voxeo’s PhonoSDK, you can connect the caller (web customer) to any SIP-based application for FREE! Using Teleku or Tropo, you can transfer your SIP call from Phono to any PSTN “real” phone number in the world or simply interact with the web user with speech recognition and text-to-speech and even record and transcribe browser conversations or interact with multiple users via a browser-based conference call!
Since Teleku also supports hosting Twilio applications running TwiML (and SIP), you could easily run your Twilio application on Teleku’s Ninja platform for FREE and simply have Phono call your Teleku/Twilio extension at sip:email@example.com.
Give it a try! We would love to hear your feedback on placing calls to your Teleku, Twilio, and Tropo applications!
Teleku and Tropo go together like chocolate and peanut butter. Since Teleku was already running on Tropo’s network, there was no integration work required to bring these services together. We are working on Product roadmaps and have identified a number of features that make sense to add to both platforms. In the mean time, customers are free to move across platforms at the same rates as their application needs dictate.
In the process of evaluating the differences between Teleku and Tropo, we have gained a greater appreciation for the Tropo platform and the shear depth and architecture of Voxeo’s private voice cloud. If you are serious about keeping pace with viral growth of your telephony application or handling call volumes that scale from 1 simultaneous call to literally a 100,000 simultaneous calls, Tropo is the platform for you!
Teleku’s Ninja application running on Amazon EC2 was a good experiment in cloud communications running on public cloud computing. For the most part this environment is very stable but we have learned that Amazon’s cloud is optimized for Web traffic not Voice traffic. There is no QoS (Quality of Service) in place for prioritizing voice packets over web packets which causes VoIP jitter and latency at times. Perhaps this will get better in the future. In the mean time, Ninja will continue to remain free with SIP support only.
Tropo’ private Voice cloud is designed to optimize Voice over Web traffic and yield the prioritization of Voice packets over Web packets. The network is also built from the ground up to convert everything to SIP the moment it hits the network. Both Teleku and Tropo support SIP natively which also allows for even lower latency and reduced routing costs of phone calls around the world - even HD voice!
While Teleku and Tropo do not currently offer SLAs, these platforms are running on the same network as half of the Fortune 500 companies’ voice applications which are supported by 100% Voice and SMS uptime SLAs by Voxeo (our parent company). This means that everything other than the Teleku and Tropo Web and Media servers are being powered by the same technology! At $03 per minute and $.02 per SMS message, these platforms are truly great values.
The teams have also adjusted to working well with each other and even started a new project together called THE NEXT BIG THING! We can’t wait to share more information about this game-changing new project in the upcoming months and how it will allow Teleku and Tropo to leap-frog the market. Stay tuned for more details!!!
Twilio, one of our competitors in the cloud communications space (covered by Jason Kincaid of TechCrunch), released an open-source, web-based PBX application called OpenVBX. The name is a little misleading as only the OpenVBX PHP web application that let’s you administer the PBX is open-sourced while the platform is designed to only run exclusively on Twilio’s services at $.03-$.05 per minute - UNTIL NOW!
By releasing OpenVBX, Twilio helped Teleku more than they currently realize. We have always supported the Twilio Markup Language (TwiML). We just needed a good test application that had extensive coverage of the Twilio API to finish our ability to test Teleku’s support for Twilio’s API and TwiML (in addition to our own easier API and PhoneML).
Teleku’s implementation of the Twilio API and TwiML has been enhanced and now more than 90% of OpenVBX’s features running on Teleku with no changes to the OpenVBX code base except for changing the URL of the API from https://api.twilio.com to http://api.teleku.com. This means that OpenVBX is now - well “open”. Customers are no longer “locked” into Twilio hosting to use OpenVBX. Since Teleku ultimately translates PhoneML and TwilML to VoiceXML, OpenVBX now runs on any carrier’s platform such as Voxeo!
So what are the differences? Teleku does not yet support transcriptions but we are working with PhoneTag and MyCaption to implement this feature soon. We also do not store recorded voice files on Amazon S3 rather we form post or email them to the developer to manage (this is a business decision that may change in the near future). Our phone number provisioning API currently does not support toll free numbers but it does support local numbers (just not by area code yet).
On the upside, Teleku runs on any telecom carrier’s network which means that you could deploy your OpenVBX application to Voxeo, XO Communications, I6NET/Asterisk, Plum, Syntellect, TellMe, Genesys, etc. Teleku supports speech recognition on all menu prompts and has many text-to-speech engine options that sound much better than Twilio’s robot voice. We also have a free SIP-based platform (Ninja), that allows free calls from any SIP client (in and out of the platform) as well as cool hacks like routing your GoogleVoice number through Gizmo (SIP Gateway) to Ninja (OpenVBX) for free calling. Our SIP support also allows you to use SIP Gateways such as VoxBone and VoIPJet (or any SIP Gateway provider internationally) to get a phone number (DID) that links to a SIP address for as little as $.01 per minute to run calls through Teleku/OpenVBX. Also, by linking your Voxeo developer account to your instance of OpenVBX running on Teleku, you get your own DID, SIP, Skype, and iNum numbers that ring right into your instance of OpenVBX. Again it’s as easy as doing a search and replace (3 hits) on https://api.twilio.com with http://api.teleku.com before installing the app or by updating the PHP source and twilio_endpoint database record in the settings table of OpenVBX.
We have registered http://OpenVBX.biz and directed our VoiceXML to Voxeo and we are able to place outbound calls, send outbound SMS messages, provision phone numbers (local), add mobile devices, create call flow scripts, link call flow scripts to numbers, and receive inbound calls and SMS messages that execute the associated call flow scrips which are running in TwiML. Other features such as creating tenants and sales/support workflows work as well.
For the good of the telecom industry, we hope that Twilio will allow Teleku to contribute code to this exciting new open source initiative, OpenVBX! We believe that for an application to be considered truly “open”, the customer should not be “locked” into a vendor’s platform. Teleku solves this problem by allowing developers to run ALL of their Twilio applications on any carrier they desire as well as by using lower-cost SIP alternatives to deliver game changing telecom solutions.
Check out this video demo of OpenVBX running on Teleku!
We recently wrote an article on how to direct your Google Voice number to your Teleku application. The only requirement was that you owned a Gizmo account prior to Google acquiring the company. Now new Gizmo registrations are closed.
If you missed out on registering for a Gizmo account, we have a new alternative solution that still allows you to link your Google Voice number to your new Teleku application (extension) to provide free phone calls into your application!
Goto IP Kall and reserve a Free Washington-based telephone number. Once IP Kall emails you the new phone number, log into your new IP Kall account and direct this number one of your existing SIP Gateway services. Add your new IP Kall number to Google Voice - Google will call this number (which will route to your SIP client) to enter the Google Voice security code. Once your IP Kall number is linked to Google Voice, update your IP Kall account to forward its calls to your Teleku application. (SIP Phone Number is your Teleku extension number and SIP Proxy is “sip.teleku.com”). Don’t forget to put a checkmark next to your new IP Kall number in Google Voice to forward your calls.
Many people are interested in how Teleku’s cloud communications platform is built so we’ll explain. There are two parts to Teleku’s cloud services (Web and Voice/SMS):
Our Web Architecture:
Both the Teleku.com web site and its RESTful phone web service APIs are built 100% on Ruby on Rails. Our Web application is hosted on Heroku which is essentially Amazon EC2 cloud hosting for Ruby/Rails applications. The website and its APIs are designed to scale as demand fluctuates. We are using NimbleNodes to monitor our Heroku Dynos to help us scale automagically with demand.
Our Voice/SMS Architecture consists of two platforms (Ninja and Samurai Warrior). Ninja is a stack of open source telephony technologies running directly in the Amazon EC2 cloud. The Ninja stack consists of Asterisk, OpenVXI, VoiceGlue, Flite TTS, and a little custom Sinatra/Ruby glue written by us.
Ninja relies entirely on SIP-based communications to handle its inbound and outbound communications. We have recently integrated GoogleVoice and Gizmo into our platform for both voice and SMS functionality. More details about this integration can be found here. You can also route any PSTN phone number (local or toll free) into Teleku using a SIP Gateway Service Provider. SMS services are provided via direct carrier SMTP traffic and, in GoogleVoice’s instance, RESTful Google APIs. Outbound calls are also supported on the Ninja platform via SIP. SIP Gateways can translate SIP addresses to outbound PSTN phone numbers for as low as $.01 per minute via Gizmo and others.
We offer upgraded telephony hosting services under our Samurai Warrior label through a number of VoiceXML Gateway hosting providers including: Voxeo, Plum Voice, I6NET, XO Communications, TellMe, Genesys Lab, and others. These services offer improved text-to-speech and speech recognition capabilities as well as other services such as: Skype integration, SLAs, and managed phone numbers.
Since our Samurai Warrior offering basically translates Teleku PhoneML to VoiceXML, enterprise customers can also take advantage of this service to run Teleku-based phone applications with their on-premise VoiceXML Gateway infrastructures. Our cloud communications service allows calls to originate and/or terminate on the enterprise’s own VoiceXML Gateway platform. Only our PhoneML web service translations to VoiceXML occur on our SaaS cloud platform. All call traffic, web applications, VoiceXML hosting services, and databases can reside behind a customer’s firewall on-site. We believe that our solution uniquely caters to enterprise customers by allowing them to leverage their existing VoiceXML infrastructure investments, security of data on-site, and engaging their Web developers in building and maintaining enterprise phone application portfolios.
Teleku allows enterprises to take a baby-step into cloud services without the perceived risks that enterprises typically associate with the cloud.
Did you think that it was possible to ask for anything more cool than Google Voice? What if we told you that you could now build telephony applications (both Voice and SMS) using the Teleku phone web service API? It’s for real - oh yea - and it’s out of private beta - no more special invitations!
You can now build cool phone applications like Twelephone, SurfByTel, or even typical business applications like order status systems, find me services, flight reservations, call center apps, emergency notification systems, etc. using your Google Voice number and any Web programming language that you enjoy.
Even if you are not a programmer, Teleku has a basic call flow wizard that will allow you to drag and drop your way to a cool phone application using your browser. What are you waitin’ for? Signup today at http://teleku.com/account/signup and follow the instructions below to link your Google Voice number to our platform and begin building something cool!
If you were one of the lucky ones to get a Gizmo SIP account before Google acquired them, you’re going to love this trick! Google Voice allows you to route your phone number to Gizmo’s SIP services while Gizmo allows you to route your SIP traffic to your Teleku extension. You can also route your Google Voice SMS traffic to your Teleku extension as well!
Voice Routing Steps
1. Log into your Google Voice account. 2. Click on your phone number and add new phone number to your account. Enter your Gizmo phone number (+1747xxxxxxx) and select Gizmo from the drop down list and click Save.
3. Select Gizmo as your primary phone number to ring when someone calls your Google Voice number. (Uncheck all other numbers.) 4. Click on the Calls tab and uncheck Call Screening and set Call Presentation to Off and click Save Changes.
5. Log into your Gizmo account and click on the Call Forwarding tab. Select Forward All Calls and select SIP. Enter your Teleku SIP address (firstname.lastname@example.org) into this field and click Save.
Congratulations! If you followed the instructions properly, you should be able to call your Google Voice number and hear your Teleku phone application!
SMS Routing Steps
1. Log into your Google Voice account. 2. Click on your phone number and then the VoiceMail and SMS tab. 3. Check the SMS forwarding messages to email.
4. Send an SMS message to your Google Voice number and check your Gmail account. 5. Select the new SMS email message and click the More Actions drop down and then Filter Messages like these link. 6. Leave the defaults and click Next Step. 7. Check Forward it to: and enter your Teleku SIP address (email@example.com) into this field and click Create Filter.
Congratulations! If you followed the instructions properly, you should be able to receive Teleku Web service form posts to the URL configured in your Teleku extension when your Google Voice number receives an SMS message!
Tell us what you’re building and we’ll help you promote it!
If you were lucky enough to get a Gizmo SIP account before Google acquired them, you can follow along with me and direct your Google Voice phone number to any application extension on Teleku’s new Ninja platform.
Step 1. Route your Google Voice phone number to your Gizmo SIP account.
Step 2. Forward your Gizmo SIP account to firstname.lastname@example.org
Step 3. Call your Teleku application using your Google Voice number. Remember to turn off call screening and announcing.
Have phun phreaking!
Teleku releases its Ninja!
Ninja is Teleku’s new open source telephony stack running Asterisk, OpenVXI, and VoiceGlue on Amazon EC2’s cloud!
Wanna send someone a link to your phone app? Try http://teleku.com/sip/190. This simple link directs your web page to Teleku and opens your preferred SIP telephony client and dials extension 190 on Teleku’s new Ninja platform. Your SIP client can now also call extensions directly via email@example.com or firstname.lastname@example.org or you can embed a link in your app like <a href=”sip:email@example.com”>Call My App</a>.
"Build it and they will come." - Field of Dreams. You’ve built a voice app and you’re running it on our free SIP-based Ninja platform. Your boss calls and loves it but asks if it can also support Speech Recognition, Skype, iNum, and SMS.
With the swipe of his credit card, you can say “you bet”! Without changing one line of code, you have upgraded your voice app from Ninja to Samurai Warrior running on Voxeo’s extreme voice hosting services infrastructure. All of a sudden your app sounds better with improved text-to-speech and mp3 support and it now recognizes spoken commands and even starts conversing with SMS text messaging clients without changing a single line of code!
This once again proves that Samurais can beat Ninjas.