Fonolo Widget – Deep Dialing For Your Business
Fonolo for business lets you embed your companies phone menu (IVR system) right on your website, or in your mobile applications, with just a few simple lines of HTML code.
Fonolo automatically maps out your phone system (and stays up-to-date with any changes you make)- You will get happier customers, fewer misdirected calls, and better feedback from your callers. Best of all, you do not have to change anything in your existing phone system.
Here's a really simple example for our fake "Fonolo Airlines" company:
More details about Fonolo for business can be found here. We're running several live trial runs with a handful of companies- if you're interested in getting access to the Fonolo widget, contact us.
Fonolo named to Time.com’s “50 Best Websites 2009″
We're pretty excited here at Fonolo today- We were named to Time.com's "50 Best Websites for 2009".
Other websites on the list, include: Twitter, Google, Skype, Amazon and Flickr- which is great company to be in- we couldn't be happier.
It's great to see how well Fonolo resonates with the public. It’s further proof that we’re addressing a point of pain felt so deeply by so many people.
You can read the whole article here.
All new Fonolo interface launched!
I'm very excited to announce the release of the new Fonolo consumer web interface. This new web interface not only has a new look and feel, but has improved usability, and many new features and enhancements.
Some of the enhancements to the web portal include:
Anonymous Deep Dialing
Anonymous, in the sense that you do NOT have to sign-up for an account with Fonolo, to try the service out. Search for the company you want from the list of over 500 companies in our database, right from the main page of our website, and click to start your call- no sign-up required. no special software. no special phone- all for FREE.
QuickTones
Ever been on a phone call, and they ask you for your account number? or if you have a Frequent Flyer number, but you can't remember it, and the piece of paper you have it written down on is long since gone?
Store frequently used account numbers, frequent flyer numbers, PINs, etc, in the Fonolo system, for one-click access while on a call. Simply select the tones from your list, and Fonolo will send the tones on your call, as if you had pressed the buttons yourself.
Heads-Up Display for your calls
While a call is active, you'll have full control over adding notes or recordings, with our new "Heads-Up Display" at the top of your screen. This control panel remains active during the life of your call, even if you navigate to other sections of the site.
Improved Call Recordings and Notes
When you make recordings during a call, Fonolo will organize them on a master timeline for that call, along with any notes that you added. Now you can flag the important parts and easily find them later.
Try the new Fonolo website today! it's completely free, and easy to use.
Handling SIP URI Dialing in Asterisk
Asterisk, by design, is very "extension" orientated- that is, if you want to dial an end-point, it requires an extension to route the call to. These extensions (defined in the asterisk extensions.conf file), can be extensions registered to phones, DIDs (XXXYYYZZZZ), or simply usernames assigned to users by the network administrator. Extensions are used for both incoming, and outgoing phone calls.
For example, if I place a call through my SIP phone to 1-444-555-1212, then asterisk will look up the "extension" 14445551212 in the extensions.conf file, to determine how to route the call.
Similarly to how e-mail works, an artifact of these extensions is a direct SIP address, which is basically your SIP extension @ your SIP server- so, if my phone was extension 555, and my SIP server had the IP address 192.168.1.1, then my SIP address would be: sip:555@192.168.1.1, and (if it's not implictely blocked), I can be dialed directly using that SIP address.
But, because Asterisk is so extension orientated, it doesn't easily allow for outbound dialing, using remote SIP addresses; If I try to dial the address sip:444@somedomain.com, Asterisk will immediately strip off the host portion (@somedomain.com), and try to route the call based simply on the extension "444"- which, since it's an extension on a remote server (@somedomain.com), it won't be able to route it locally, and will fail.
The solution? Well, it's not ideal, but Asterisk provides the ability to use wild cards in the extensions.conf file (it refers to them as "patterns") when doing extension look ups; this is handy when you have blocks of extension or DID's- you can use the wildcard to map like extensions to the same config, keeping your config file small.
The issue, is that the wild card only compares against the local part of the SIP URI, which can look like almost anything, including other phone numbers.
First, define some general config
[general] ; ; Your termination provider (defined in sip.conf) ; TERM_PROVIDER = SIP/company_peer ; ; The IP address of this asterisk server ; ASTERISK_IP = 192.168.1.1
The "ASTERISK_IP" address is important later, as we'll use it to validate outgoing SIP addresses.
Then in your default config, you should have something like this configured already- handling NANPA style dialing, and international dialing
[default]
;
; Dial NANPA style phone numbers directly
;
exten => _1NXXNXXXXXX,n,Dial(${TERM_PROVIDER}/${EXTEN},60)
exten => _1NXXNXXXXXX,n,HangUp()
;
; Dial international numbers directly
;
exten => _011.,n,Dial(${TERM_PROVIDER}/${EXTEN}, 60)
exten => _011.,n,HangUp()
After all the specific matches are done, then add:
;
; very last, assume anything else is a SIP URI
;
exten => _.,n,GotoIf($[${SIPDOMAIN} = ${ASTERISK_IP}]?unhandled)
exten => _.,n,GotoIf($[${SIPDOMAIN} = ${ASTERISK_IP}:5060]?unhandled)
exten => _.,n,Macro(uri-dial,${EXTEN}@${SIPDOMAIN})
exten => _.,n,HangUp()
;
; if the call doesn't match anything
;
[unhandled]
exten => s,n,Congestion()
The reason this has to be last, is because matching the extension "_." will match *anything*- basically it's a catch-all. You're saying that if it doesn't match anything else before this, then assume it must be a SIP URI.
This section also compares the ${SIPDOMAIN} variable to your ASTERISK_IP address; this ensures that only SIP URI's with remote hosts are processed as SIP URI's. If the host matches our ASTERISK_IP address value (ie- it's a local extension), then it should have already matched something above this catch-all config.
Pass any SIP URI dials to the uri-dial MACRO, merging back together the extension and the SIP domain value.
;
; handle dialing SIP uri's directly
;
[macro-uri-dial]
exten => s,n,NoOp(Calling as SIP address: ${ARG1})
exten => s,n,Dial(SIP/${ARG1},60)
This solution works, but isn't ideal, as it will match anything that didn't already match; a better solution would be for it to NOT strip off the SIP domain, and allow for using a regular expression of some kind to check the extension, but there is currently no better way of handling this in Asterisk.