Category Archives: Telephony

Handling SIP URI Dialing in Asterisk

1 Reply

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.asterisk

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.

Gizmo5 747 Area Code

Leave a reply

I was playing with the Gizmo5 service today- we’ve had a few customer now ask if they could have Fonolo call them back at their Gizmo5 address, instead of a regular PSTN number. Their service, which is very Skype-ish, aims as providing cost saving on international phone calls, as well as provides free calls between Gizmo5 users.

One of the first things I noticed (after installing the Gizmo5 client), was that their system allocates each user a “SIP” number, which they made the unfortunate choice of formatting like a standard ten digit phone number (XXX-YYY-ZZZZ)- not only that, but they used the area code 747!2t_gizmo-logo

Unfortunately, they don’t have any stake in the *real* 747 area code controlled through NANPA– even worst, looking at the NANPA database, the 747 area code is already allocated as an overlay to the 818 area code (in California), and is set to be put in service May 18th, 2009 (~3 months from now).

Now, the Gizmo5 site doesn’t seem to make any mention of this, and there are only a few posts out there I can find- but they all simply say that the “SIP number” was never meant to be used as a normal PSTN number- it can only be called from one client to another, or as a full SIP address (by tacking on @proxy01.sipphone.com)- if that’s the case, it seems really weird that they opted to use phone numbers, instead of just usernames.

I’m not sure how they plan to handle it when *real* PSTN numbers in the 747 area code start getting allocated to people in California, and then one of their users with a DialOut package tries to call one of them? I’m sure it’s not going to take long before phone numbers start being duplicated, and they’re going to have to solve this issue.

Not to mention general confusion; I’ve read post after post about people being confused about why they can’t dial their 747 phone number from PSTN number.

As far as Fonolo support, I was hoping to simply trunk any calls to the 747 area code over SIP to the Gizmo5 system, as if they were just regular phone numbers; in testing, it seems to work well, but I’d just run into the same issue they will, come May.

I’ll have to just add them into the system as generic SIP addresses- hopefully I’ll have this working soon.

White Spaces in Canada?

2 Replies

I’ve been of the opinion for years, that at some point soon, people will no longer have to pay for Internet access; that it will devolve from a product itself, into simply a product delivery system.

Not only that, but all our services (cell phones, home phones, TV, and of course, “Internet access”) will be delivered over this network- which, if everybody is connected (end-users and businesses alike), will open up huge opportunities for competitive companies in all sorts of markets.

And, of course, it will all be that much easier if this magical Internet connection was fast, and wireless; just give your new TV some power, and it will connect to your services provider automatically using it’s wireless card; forget about cellular- your new portable VOIP phone will connect over the same wireless network.
dtv
No longer will whole provinces/cites/states be subject to incumbent providers, who’s only grip on our patronage, is that they happen to be the current guardians of the copper.

Well last Nov, the FCC voted unanimously to approve the use of the unused airwaves between broadcast TV channels (aka the “white space”), for wireless broadband service for the public (which, I have to say is pretty surprising that the FCC would support something good for the people, and not special interest groups)- of course, it was only after years of testing and poking and prodding by the Wireless Innovation Alliance (of whom some notable members are Google and Microsoft).

The spectrum itself is also ideal, as it has a much longer range than wifi, which means fewer base-stations to cover larger areas, which ultimately means lower costs to operate- so while it’s been hinted at, it doesn’t say this wireless Internet connection is going to be free- but definitely affordable (cheap even)- and one step closer to free.

On the coat-tails of the FCC, the CRTC (Canadian Radio-television and Telecommunications Commission) is meeting on Feb 17th to discuss doing the same for Canadians, but Canada isn’t set to shut off analogue TV until February 2011- which means a few years until it’s even possible.

Maybe I can fake my geo-location, and get access through Buffalo. 😉

VoicePHP by TringMe

Leave a reply

TringMe, an Indian start-up just released its new VoicePHP service, which allows you to develop voice applications using PHP.

This is definitely cool stuff, though I see quite a few issues with implementation. HTTP, is, by design, a non-persistent transaction, and a phone call requires persistency while the call is in progress; therefore, there is no way for PHP alone, to handle “Voice” requests.

The only way to really make this work, is to have an external server (effectively a PBX of some kind) that handles the call traffic, that is simply polled (in some fashion) by PHP on a per request basis. Those requests could be polled via Ajax, or even a comet-like system, but it’s still not handled directly via PHP. I assume this is where their VoicePHP “Server” comes in.

voicephparch

So, in this sense, it sounds simply like a PHP wrapper around the TringMe REST API, which is something that can already be done in other ways, like accessing the Asterisk Manager interface via PHP using asterisk-php-api, but with one interesting twist-

It looks like (according to their FAQ) you can access the audio streams directly from PHP- which is pretty interesting, depending on how they’ve implemented it. Given the (generally) short transaction lengths of an HTTP request, I’m not sure how you would realistically pull audio from a real-time source (like a phone call), or play-back an audio clip on to a call, unless you used some sort of scheduling system.

ie- use PHP to decode and audio file (say .wav or .mp3), and then send the file to the VoicePHP server, which schedules the audio clip for playback, and returned immediately. It’s then played over the call in “real-time”. PHP could then poll the VoicePHP server to get status about the playback.

I’ve signed up for a beta account to play around with it, hopefully I’ll see something from them soon.

SIP Softphone for iPhone

Leave a reply

I’ve been using an iPhone for quite sometime now- I bought one just before the 3G’s came out- I was going to wait, but I figured I’d be able to pickup a first gen pretty cheap with the 3G’s coming out, and I wanted to get an unlocked one anyway, as I didn’t want to end up getting stuck with the Rogers plans/term contract.siax

I just installed a new app the other day, that gives you a native SIP/IAX2 client for your iPhone- which is pretty awesome, and something I’ve been waiting (and considering writing) for a while now. It’s called SiAX, and currently only available through Cydia (jailbroken phones).

The app works fairly well- the audio quality is good, and seems to be pretty solid. Unfortunately, you only seem to be able to receive calls through the app while you have the application running; it doesn’t maintain any SIP registration information when you close the app, which is a little weird. But for a quick call here and there, especially to avoid roaming charges on the road, it’s pretty decent.

I’ve actually disabled my data plan with Rogers, as it’s a rare occasion that I’m not around a wifi connection- even when I’m travelling, there’s usually at least one Starbucks around. Of course, this was only after incurring hundreds of dollars in roaming data usage.