Now, on to the good stuff…

The hardest part of the installation was getting the drivers to compile. I started with a stock Ubuntu Hardy Heron 8.04 with the 2.6.24-16-server kernel. Failed. Updated to 2.6.24-19. Failed. Finally, installed the 2.6.24.3 full source kernel. Failed. Through the entire process I was in contact with Sangoma developer Nenad Corbic who was extremely helpful in getting this thing to run in a very timely manner. Total props to Nenad and the team at Sangoma. A most impressive showing on how it’s done. We had gone from wanpipe drivers 3.0.5 to 3.0.7 to make this work, which is pretty good considering it’s really a beta version driver.

Once the drivers had compiled the rest was a breeze, as anyone who’s done a Sangoma install knows, they even setup the configuration files for you (ok not the dialplan, but please). Just reload Asterisk or FreeSwitch and you’re good to go. Here’s the HowTo from the Sangoma site.

According to the developers the U100 USB FXO does not have HWEC (hardware echo cancellation). But they do recommend using the excellent OSLEC software based echo cancellation or you can use the built in ones Asterisk uses. Just enable “echocancel=yes” in /etc/asterisk/zapata.conf. The configuration is the same as any other zaptel device in asterisk, so I won’t go into configuration files as that has been well covered in many howto’s.

Overall, this device was not bad to work with, even though it was still in BETA. As I stated above, this device nicely fills a niche in the SOHO/SMB space for PSTN connectivity. With some additional polishing up, you will undoubtedly see this device connected to many of the SOHO appliances in the future.

With that being said, can anyone have any thoughts as to how they would use this device?

This information was originally posted on the VoIP Insider blog.

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1. MAC Address: this allows you to simply plug in the Ethernet hardware address of each device you want to give priority to. This is usually listed under the status information on a router. Look around in your router or ATA and you’ll find it. This method is a very easy setup for phones or ATA adaptors. The downside is the don’t give you many entries.
2. Ethernet Port on the Router: If you have a device directly connected to that port then just give it the priority you’d like. That simple. Could be an ATA , a Phone or even a larger switch that y ou connect devices to.
3. TCP/IP Ports: You can enter the ports that VoIP uses, usually 5060 for SIP and 10000-20000 for RTP. Your router will then listen for any traffic on these ports and give preferential treatment to those ports. On the other hand you could decrease bandwidth for BitTorrent type applications.

QoS as found on any consumer router running on a standard Internet Service Provider will ONLY work on upstream/outbound data (data going from you to your ISP). You cannot realistically control the priority of data coming TO you FROM your ISP, since you can only control the data on your side of the modem.

It is true that slowing down the download of data will slow the acknowledgments of that data in a TCP/IP connection, and will therefore slow down (eventually) the transmission of data from the remote. However, this is not a function of QoS. This is a function of TCP/IP. And it will not solve contentions between VoIP and other data, since VoIP data still doesn’t get priority. You CAN help this out by limiting your bandwidth to downloading applications if theiy support it, but this is throttling and not setting priorities.

So, QoS can help a bit and may be worth the effort if you think your network is causing problems with your VoIP.

This information was originally posted on the VoIP Insider blog.

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Asterisk Setup:

The Asterisk setup is easy. Just create standard type=friend extensions for as many phone extensions as you would like to create in sip.conf. A typical basic entry would look like below.

[1001]
type=friend
context=local ; Context for incoming calls for this user
host=dynamic ; This peer register with us
dtmfmode=inband ; Choices are inband, rfc2833, or info
username=1001
secret=1234
disallow=all
allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
;progressinband=no ; Polycom phones don’t work properly with “never”

Create dialplan entries in extensions.conf
exten => 1001,1,Dial(SIP/1001|25)

AudioCodes Setup:

Connect the gateway to a network switch and connect a computer to the same
switch. Then configure the IP address of the computer to 10.1.10.XXX (anything but 10 or 11). Then run your web browser and point it to http://10.1.10.10 and login using

Default IP address: 10.1.10.10
Default username: Admin
Default password: Admin

Click -> “Quick Setup” and change the following:

IP Address => IP address of the AudioCodes 124
Subnet Mask => Set to the correct netmask for your local network
Default Gateway Address => Gateway IP address for your LAN
Working With Proxy => Yes
Proxy IP Address => IP address of the Asterisk server
Enable Registration => Set to “Enable”

Restart the gateway and log back in using the new IP address.

Protocol Management -> Protocol Definition -> Proxy & Registration
Registrar IP Address => Set to the IP address of the Asterisk server
Registration Time => 60
Subscription Mode => Per Endpoint
Authentication Mode => Per Endpoint

Go to Protocol Management -> Protocol Definition -> DTMF & Dialing
Max Digits In Phone Num -> Make it a large number like 32 digits

Go to Protocol Management -> Protocol Definition -> Coders
Add coders as needed You need to set at least G.711U-law

Go to Protocol Management -> Endpoint Settings -> Authentication
Set SIP username and password for each port to match your settings in sip.conf

Go to Protocol Management -> Endpoint Phone Numbers
Enter an extension (phone) number for every used channel to match entries in sip.conf

AudioCodes FXS gateway is ready. Worked here YMMV.

This information was originally posted on the VoIP Insider blog.

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Asterisk is starting to look like a first generation IP PBX. It is apparent now that its dominance in the FOSS VoIP market may be seeing its first real challenges by real players. Asterisk is going to have to start from scratch. They wouldn’t listen to their community, and may now pay the price for that hubris. Can they fix an inherently flawed design? Time will tell.

When I started with Asterisk, the learning curve was awfully steep. I didn’t know that much about telephony, and when I started with FreeSwitch I had to learn an entire new syntax and way of doing things (see -> steep learning curve). A lot of the people that are hanging on to Asterisk are doing so because they really don’t want to have to relearn something all over again. They are comfortable. Others have gained status in a community and don’t relish the idea of having to start at the bottom of the food chain again as noobs. They are comfortable. Asterisk gave me my start with VoIP, and I will always have place for it in my heart. But, I also realize that times and technology change, and so must I. Asterisk isn’t going anywhere anytime soon, but the writing is on the wall, and it reads “Niche Player.”

We here at VoIP Supply will have to react as the market changes or face the very real possibility that our competitors will do so first, and become the market leaders in doing so. To date, VoIP penetration has been limited by scalability. With new systems offering greater capacity the opportunity for growth is certainly there. The enterprise market that was previously cautious regarding VoIP may be a bit more open now. My point is not knocking Asterisk, simply categorizing it for what it is, a decent solution for the SMB/SOHO market. But, as my boss put it: “given the choice between a Mercedes and a Ford for the same price, I’ll take the Mercedes.” Others may feel the same.

UPDATE: The FreeSwitch group is getting ready to release version 1.0.1 soon that will improve the code’s stability and add Automatic Speech Recognition (ASR) and Text To Speech (TTS) to the code.

A little background on what others think of SipXecs is also in order. From the hyper-connected enterprise blog on TMCNet: “Asterisk may be older but sipXecs is better” (a Nortel Guy).

For a little primer on design differences look here.

This information was originally posted on the VoIP Insider blog.

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This is an FXO unit with 4 ports. The setup would be pretty much the same for a 118. I am using a standard vanilla Asterisk 1.4 install. I’ve always preferred a lunchbox Asterisk setup to a user friendly GUI because I find it much easier to troubleshoot when things aren’t working as advertised. This will be the first part in a continuing series on setting up these gateways in different scenarios. My FreeSwitch setup is using the vanilla default profile setup.

We only need to configure sip.conf and extensions.conf to get a working setup on the asterisk end.
Asterisk Setup:

sip.conf : We can use one (type=friend) or two (type=user & type=peer ) entries.

Single or Friend Settings

[pstn]
type=friend
context=inbound
dtmfmode=inband
host=192.168.xxx.xxx ; IP address of MP-114
nat=no
canreinvite=no

Paired or User/Peer Settings

[pstn-out] ; used for dialing out
type=peer ; peers help deliver the calls for us.
allow=ulaw
context=outbound ;not necessary to, but lets us know its function
dtmfmode=inband
host=192.168.xxx.xxx ; (This is the IP of the MP-114)
nat=no
qualify=no
[pstn-in]
canreinvite=no
context=inbound ; Where to deliver the inbound calls in extensions.conf
dtmfmode=inband
host=192.168.xxx.xxx
nat=never
type=user ;we are a user of MP-114 FXO

extensions.conf ; Doesn’t matter much here whether it’s friend or user/peer model

[outbound] ; Context for Outgoing Calls
exten => _NXXXXXX,1,Dial(SIP/${EXTEN}@pstn) ; @pstn-out if you’re using the user/peer model
exten => _NXXNXXXXXX,1,Dial(SIP/${EXTEN}@pstn)
[inbound] ; this is our telephonenumber
exten => _2125551212,1,Answer() ; let the gateway know we’ll handle it from here
exten => _2125551212,n,Wait(1) ; give a sec to get any passed info
exten => _2125551212,n,Dial(SIP/1001,25) ; or point it to your IVR

FreeSwitch Setup: I’m still a noob here. Originally I really completely over thought this one and made far it more complex than needed be. Failed, of course. Then I went super simple with it. Worked! I hope I don’t get flamed for poor design, but it works!
I needed to create an unauthenticated extension in the directory (extensions for asterisk users). This could dicey /unsafe, but it is internal.

/usr/local/freeswitch/conf/directory/default/pstn.xml:

Now for the dialplan settings that make it actually work . There are two place under the dialplan directory I needed to edit.

1. Let FreeSwitch know we have a number for the public to reach.

/usr/local/freeswitch/conf/dialplan/public.xml

2. Make sure the DID is in the default dialplan so FreeSwitch knows how to handle the calls

/usr/local/freeswitch/conf/dialplan/default.xml

First let’s receive calls.

Now let’s make calls. This is for 7 digit calls, but would apply to long distance also.

AudioCodes Setup
Quick Setup:

IP configuration: If you can’t figure this one out there’s little or no chance you’ll get this working. Put on dunce cap and sit in corner.

SIP parameters:
Gateway Name: I use it’s IP address so no dns issues.
Working with Proxy = Yes
Proxy IP address= the IP of the Asterisk box.
Proxy Name= the IP of the Asterisk box.

Protocol Management:
Protocol Definition ->
General Parameters:

Channel Select Mode=Ascending

And make sure SIP ports are set for 5060

Proxy and Registration:
Proxy Name and Proxy IP Address= Asterisk Server

Enable Registration: I didn’t .

Gateway Name and Registration Name: MP-114 IP address

Subscription and Registration Mode: Per Gateway (don’t remember if this matters).

Coders:make sure ulaw’s there

DTMF & Dialing: Max digits-> put a high number like 32

Routing Tables:
Tel -> IP routing and IP-> Tel routing = I used

Dest IP/Phone Prefix =*

Source IP/Phone Prefix =*

Dest/Source IP Address = Asterisk IP Address

Endpoint Phone Numbers: Match channels to phone numbers.
Channels= 1 -4
Phone numbers = your phone numbers

Hunt Group Settings:
I used Cyclical Ascending

End Point Settings:
Automatic Dialing:Destination Phone Numbers should match the numbers you have in inbound context in extensions.conf. In our example -> exten => _ 2125551212,1,Answer()

Advanced Applications:
FXO Settings: Dialing Mode should be set to One Stage.
That should get you up and running. Although little differences in setups can cause major headaches and frustrations, I hope that this gives you a good starting reference point. We’ll be putting this and other guides on our wiki when it becomes available (with screencaps).

This information was originally posted on the VoIP Insider blog.

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FreeSwitch is getting a lot of attention from those in the VoIP and Asterisk community and seems quite promising. Built from the ground up to address issues with scalability and deadlocks within channels, it’s rock solid. It doesn’t have as many applications right now as other software, though its modular architecture will make it a favorite of developers. Think of it as somewhere between Asterisk and Ser from a technical standpoint. Hurdles to adoption with people now using Asterisk will be: its lack of quality, easy to understand documentation, and a need of financial backing (Asterisk is venture capital city). If that happens, look out, “a storm is a comin’.” This is really powerful and flexible stuff we’re talking about here. This is one horse that could pay off at a price for those willing to take a shot.

It’s definitely its own animal. Configuration files are all in XML, which are easily edited with a plain text editor. The command syntax will be completely foreign for anyone coming from planet Asterisk to this brave new world. I wrestled with it for a few times before the light bulb went on and success was at hand. I’ll be posting all my configuration files for a small SOHO setup with an IVR and a couple of phones here. Voicemail is right out of the box and hardly any configuration is needed. If you need help just go to their IRC channel on http://freenode.net/irc_servers.shtml, and ask one of the friendly folks there. They have a neat way of helping people there. You take a number and one of the channel operators will help you with your problem. I noticed in the IRC channel, a lot of familiar names, as will anyone familiar with Asterisk. So, this is where y’all been hiding! I was told by someone there to think of FreeSwitch like Lego’s. The problem with previous telephony software was that the Lego’s were glued together. And they wanted their building blocks, being developers, to be manipulated in any way possible and as free from the core as possible. And that seems to be a main difference with Asterisk, whose core is much larger and tightly integrated. FreeSwitch is still in its infancy and will have a very bright future. One nice feature it provides is higher audio quality. It supports 8 kHz (normal telephone), 16 kHz (widenband or g722) and 32 kHz (ultra-wideband). Now people can fully take advantage of those HD-phones. There is also video support built in and it should be interesting when and what applications will pan out of that feature.

FreeSwitch already supports a whole lot of features, codecs, and protocols:

* SIP, H.323, IAX2

* GoogleTalk and Jabber support

* Centralized user and domain directory

* Stereo call recording

* Software-based conferencing

* Configuration files and call records in XML

* Interactive voice recording and call attendant

* Narrow and wideband codecs

* Voicemail-to-podcast

* Urgent message tags

* Multiple SIP registrations per user account (I like this one-very handy for testing!)

* Multi-tenancy

* NAT support

* STUN support

* Message-waiting indicator

* Presence, SLA (shared line appearance), and BLF (busy line field)

* Zapped and Sangoma hardware support

* Shoutcast streams

* OpenZAP, a common driver and interface for TDM (analog-to-digital) hardware such as Sangoma and Zaptel

From Dave Greenfield’s blog at zdnet.com:

“We replaced a cluster of 10 Asterisk servers with a single FreeSwitch server,” said Chris Parker, director of systems for a large publicly traded CLEC (Competitive Local Exchange Carrier).

Parker says he’s getting several hundred concurrent calls on a single, dual-core box that’s also doing all of the media processing, a computationally intensive task. How’s that for Big Boy Pants?!

This information was originally posted on the VoIP Insider blog.

Need VoIP Phones or a VoIP Phone System? Checkout VoIPSupply.com!