Tag Archives: asterisk

FreeSWITCH 1.0.0

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I didn’t see any mention of this on the VoIP blogs I read, but it seems to be worth noting.

FreeSWITCH version 1.0.0 was released on May 26 after about three years of development. I would call it a cross between Asterisk and OpenSER. It’s robust like OpenSER but has more base features similar to what Asterisk provides. Makes sense, as the developer is a former Asterisk developer.

Interesting blog post: How does FreeSWITCH compare to Asterisk?

It’s designed as a soft-switch, not a PBX (see FAQ) but you can add the PBX features on to it that you want. I think this will be a real competitor to Asterisk, sipX and whoever else in the free/open-source world.

Junction Networks offers some cool services including hosted conference bridges, which they’re powering with FreeSWITCH.

7906, meet Asterisk

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The Cisco 7906 with the current SIP firmware load (8.3.4 SR1) connected to an Asterisk system performs very much like it does connected to CallManager via SCCP. Cisco has been pushing for feature parity between SIP and SCCP protocols on their IP phones (see this white paper) since they began to really embrace SIP a couple years ago. Because SCCP implements most of its functionality on the PBX end, and SIP puts most of the functionality at the endpoint, Cisco’s SIP firmware has to implement some of the CallManager functionality and look-and-feel on the endpoint device.

7906 with SIP firmware

Biggest drawback when using SIP is the need to setup a client-side dialplan in order to effect PSTN-like (or CallManager-like) dialing: the PBX knows when you’re done dialing a number and immediately begins to process it. With the SIP firmware you have to either set up a client-side dialplan or press a “Dial” softkey after entering a number (like you do on a cell phone). Cisco’s SIP firmware implements KPML (RFC 4730) to make this smoother, but Asterisk does not.

The interface is nearly identical. There’s a DND key on the SIP phone by default. Hold, call waiting, conference (using the phone’s built-in conference bridge), and attended transfer all work. Unattended transfer does not–I’m unclear as to why not.

I would guess that Cisco’s higher-end phones show more disparity between the SCCP and SIP interface and functionality but on the 7906, Cisco’s current low-end single line phone model, they’ve just about reached their goal of feature parity.

VoIP providers for home use

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Just wanted to mention a few VoIP providers that I use on my home Asterisk system.

  • VoiceStick / i2telecom – Using their Next2Nothing plan (pay-as-you-go) I have a DID number local to my parents. They dial a local number and I pay 1.2 cents per minute plus a couple dollars a month for “911 recovery fee” (for an incoming-only phone number!). Audio quality has been excellent. If it weren’t, my family members would surely tell me so.
  • FWD – every homebrew VoIP system should connect with FWD. Why? Well, because it’s free! I am connecting using both SIP and IAX2. I use it only to connect to toll-free numbers and to test connectivity using some of its test numbers.
  • Gizmo Project – here’s a clever way to get a free DID number thanks to Gizmo Project and Google’s Grand Central. Sign up for a free Gizmo account, set it up in Asterisk and make sure it works. Then sign up for Grand Central. (Need an account? Leave a comment and I’ll send you an invitation.) You can point your Grand Central number to your Gizmo account. Result: People call your Grand Central number (a regular phone number) and your VoIP system receives it.

    Cool, until you decide it’s annoying to respond to Grand Central’s “Press 1 to accept the call…” menu every time your phone rings. Instead, make Asterisk “press 1” for you when a call comes in and before delivering the call to your phone.

    Put a custom context in Gizmo’s peer definition, like this:

    context=custom-gizmo-in

    Then create that custom context in extensions_custom.conf (for FreePBX) or directly in extensions.conf if you’re hand-editing plain Asterisk:

    [custom-gizmo-in] exten => s,1,Answer exten => s,2,Wait(1) exten => s,3,SendDTMF(1) exten => s,4,Goto(from-pstn,s,1)

    (The goto in step 4 sends the call to the generic “incoming” context after the DTMF “1”; modify as necessary.)

  • ??? – I use a cheap termination service (outbound calling) whose terms of service state that “All customers … are specifically prohibited from disclosing to others that they use [the provider’s] service.” I leave it as a mystery for the reader. (However, I’m pretty sure I heard of this provider by word-of-mouth — contract-breacher!)
  • CallCentric – This one is on my radar. I am signed up for their free service and a single “Cheap DID” to test things out. So far I have been very impressed with their customer service (answering a non-trouble question ticket within two hours), the sound quality, pricing, and availability of DID numbers. It seems like they offer DIDs everywhere and can port (LNP) just about any number. I may port my listed home phone number (currently not connected to my VoIP setup) to this carrier.

No mention of the big guys: Vonage, Packet8, or AT&T or Verizon’s VoIP service, for three reasons. One, most don’t allow you to connect to them using a PBX (Asterisk). Two, they’re expensive. And three, if you’re home-brewing VoIP, you are probably a geek or trying to “stick it to the man.” I realize that some of the carriers that look like “little guys” really have big guys behind them, but many of them don’t. They’re just trying to make a little money reselling service from wholesale carriers and adding their own little features to make telephony a little cooler or more fun.

PBX in a flash?

I don’t have any updates at the moment on using Smokeping to monitor the quality of the VoIP network (though I am in the middle of debugging and setting it up, and it looks like it will work very well) but hope to have some graphs to show, soon.

In the meantime there has been some rustling in the VoIP blogs about a new “Asterisk-distribution-on-a-CD” called PBX In A Flash. Ward Mundy of the Nerd Vittles blog (linked on the sidebar of my site) is the man behind this distro, and it looks good. A long-time Asterisk@Home and then Trixbox user, he got tired of the limitations of the Trixbox distribution and decided to produce his own. I’ve downloaded it and plan to evaluate it. I think it will be decent because I like the various Asterisk hacks that have come from Nerd Vittles and I have the feeling this distro will be full of clever configurations and nice feature additions.

At home, I run a well-customized Trixbox, connected to the world with some cheap SIP trunking and connected to me via an analog adapter with cordless phones and softphones. To be honest, there’s no “in a flash” about it — setting up Asterisk, even with nice GUIs and self-installing CDs, is time consuming and requires a lot of research. But it’s rewarding to be able to set up a fairly powerful home PBX using free software, cheap hardware, and cheap connectivity, and have more for $10 or so a month than you could get from the phone company for $40 a month.