Cisco VoIP Ditched for Open-Source Asterisk

An anonymous reader writes "Sam Houston State University (SHSU) is moving 6,000 users off a Cisco VoIP platform to an open-source VoIP network based on Asterisk. One big driver, of course, is cost. From the article: 'We thought that it will be more cost effective in the long run to go with an open source solution, because of the massive amounts of licensing fees required to keep the Cisco CallManager network up and running,' says Aaron Daniel, senior voice analyst at SHSU."

Re:Asterisk really is best bang/buck

[Rob from RPI]

Actually, Asterisk isn't _really_ FOSS, as you have to sign a disclaimer (before you submit code to them) giving them the right to repackage it in non a FOSS way. This is so they can sell the Asterisk Binary Edition, as well as (unclear, to me) licencing issues with Intel Dialogic cards.

OpenPBX.org (nothing to do with my FreePBX project, mentioned above) is a pure GPL fork of asterisk from about a year ago, that they've done significant amounts of re-writing on, including working on a new dialplan language, as well as being able to import a lot of Steve Underwoods work (www.soft-switch.org) with software DSP (eg, soft-faxing, T.38 [fax-over-IP], better DTMF detection) that he will only licence under the pure GPL.

--Rob

Re:SCCP support?

[Rob from RPI]

I was really trying hard not to reply to _every_ post here, but SCCP is an awful protocol. And the 'low end' VoIP phone are all SIP or IAX, so you're barking up the wrong tree a bit. For example - Google for PA1688. This is a VoIP phone _chipset_ that the manufacturers have open sourced the firmware for. You can usually buy PA1688 based phones for about US$50. Or if you want more of an office phone, the Grandstream GXP2000 has a reasonably professional look, and are around US$100 or so. Going up market from there, you're looking at the Snom 320 or 360. Plenty of buttons and lights, and it runs Linux.

--Rob

Re:Asterisk needs improvement.

[Rob from RPI]

That's totally incorrect. OEJ (a leading developer) has taken Asterisk several times to SIP Interoperability Testing meetings, and has acted very proactively to fix perceived or real incompatibilities.

I just did a quick search of the Digium bugtracker, and I didn't see any 'SIP Incompatibilty' bugs there apart from an issue with sipgate.de.

I honestly think you're trolling, or you have no concept of how FOSS works. If there's a bug, you fix it, and if you can't fix it, you report it and someone who can fix it, will.

--Rob

Re:They went just a little bit too cheap....

[HeadbangerSmurf]

I used to work for a company that did a buttload of Call Manager. Well, they still do but I don't. I love Call Manager. It's an incredible platform but it's just so damn expensive. Both systems need care and feeding and I would say that Asterisk needs more of that at this point in time. The Call Manager systems I've worked on ran smoothly and required little to know intervention. Asterisk is a bit more of an attention whore right now but I figure that will change as time goes on. The big thing with Asterisk is the price. Even if we charge a ton for setup we still beat the traditional VoIP phone vendors by quite a bit. We beat one by half earlier this year, $35k to $17k for Asterisk and that was using expensive phones. I've got a CM 3.3 system installed here in town that has been going strong for 4 years. Every so often it needs to be rebooted and I did end up replacing a hard drive in the Exchange/Unity server two years ago, but that's it. It just runs. We seem to reboot Asterisk about once a month right now instead of once every ten months. That and echo tuning on Asterisk is a pain in the rear. I don't think I've gotten one system to be echo free. Of course, that could be because everyone likes to crank the volume on the phone so high I can hear it in the next room. Tom

Re:SCCP support?

[Rob from RPI]

Well yes. Their GXP firmware goes from featureless, to cranky, to bugfix, to feature+, to even more cranky than it was originally. I'm currently running some beta firmware on the GXP on my desk that has all sorts of display corruption issues.

They did, however, get the speakerphone echo well sorted out a while ago. The snoms, on the other hand, do _not_ have echo cancellation in their speakerphone, which means it can't be all that loud. Which leads to user complaints 8-\ However, apart from that minor niggle, yes, the Snoms rock. But they are 2-3 times the price of the GXP's.

If you want good speakerphone, apparently the Polycomm phones are the best.

The reason I don't like the SPA's is that you can't do BLF (Busy Lamp Field - eg, bind an extension to a lamp to see who's on the phone, pick up someone elses call by just pushing a button etc) which is pretty much a prerequisite for any compay upgrading from a Key system. And most of 'em are 8)

--Rob

Re:Asterisk really is best bang/buck

[A.K.A_Magnet]

You seem to have a good knowledge of Asterisk, yet I have to correct you on the fact that Asterisk *IS* F/OSS and *IS* released under the GPL. What you're talking about is giving your copyleft to Digium if you want *YOUR CODE* to become part of the official distribution. Nothing new here, it's a common practice, used even by the FSF which *MAY* change the license then, but you can be pretty sure that the FSF won't change it to a non-copyleft license (while Digium uses it to give non-free licenses), but how do you think they'll change all code from GPL 2 to GPL 3 [not counting GPL 2 or later, since some of the GPL'ed software owned by the FSF (ie you give them your copyleft) hadn't the "GPL 2 or Later" clause and they added it later, since the license can only be changed by an agreement of all the copyleft holders, so it's easier if it's a moral entity like the FSF, MySQL AB, Trolltech the Apache Software Foundation (even if they don't use GPL, they still may want to change their license)... or Digium. And they all ask for copyleft transfer.

My point being: yes, Asterisk is "100%" F/OSS. They just don't allow other copyleft holders in THEIR distribution. Nothing would prevent OpenPBX, to sync with each latest version of Asterisk, but as long as Digium wants to hold all copylefts, they can't include code made by OpenPBX folks. Digium wanting to hold all copylefts is a part of their business model (dual-licensing). Of course, it makes it harder for OpenPBX people to sync because of the two development trees (and I understand why they'd want to keep their copyleft). However, Asterisk remains Free Software. Maybe they're not using the "Open Source development model" at its maximum though, but who cares :). As long as it's Free (with a capital F), it's fine with me.

Re:On the subject of Asterisk

[Rob from RPI]

It's not, but I was working with a couple of gentoo guys to get it in - they seem to have vanished. The way we do an install and check for versions apparently causes a bit of grief. There are, however, gentoo docs on the wiki [aussievoip.com] - However, just checking over them they seem to be a bit lax (well, ok. A lot lax). The CentOS instructions [aussievoip.com] are far more verbose.

I'd love for someone with some gentoo clues to help out!

--Rob

Re:SCCP = Skinny?

[saridder]

But if I remember - and I'm too lazy to look it up - SCCP stands for SKINNY Client Control Protocol, and is a modified, scaled-down (skinny) version of H.323. The original Selsius (company Cisco bought in 1998 which gave us Call Manager) designers didn't have a SIP or other protocol to use back then and H.323 was too much, hence why SCCP was first created.

Re:Asterisk really is best bang/buck

[Rob from RPI]

Well, yes and no. When OpenPBX was forked, there was a fair bit of hue and cry about suing them for Trademark violation, which they resolved reasonably quickly (sed s/asterisk/openpbx/i) and then there was threats about licence violations by linking to openssl.. I can't find the exact message in the digium archive, but here's a link [digium.com] to the same issue being discussed about the freebsd port.

I tend to think that they're a bit over-protective of their code. They release it as GPL to garner community support, then as soon as someone forks it, they're all upset. That does make me a bit grumpy, but I'm probably just overreacting.

(Whilst I'm not claming a coverup, Digium do have a bit of a history of removing things from the archive [digium.com] - That link, admittedly, is a valid reason to delete stuff from the mailing list archive, but it has happened before)

--Rob

Asterisk?

[TCM]

I know everyone hypes Asterisk and Open Source and all that.

But has anyone looked at Asterisk close enough? It's the most horrid piece of software I have seen in a long time. Its configuration is awkward at best and downright inconsistent and nonsensical at worst.

Its documentation is practially non-existent. Nowhere do you find a good documentation written by the programmers. All you have are Wikis and web sites where people try and guess how Asterisk works. Howtos consist of config snippets without explaining what the options mean, let alone explaining the grand scheme behind everything.

Maybe it works after you configured it based on some other guy's experience, but if you want clean and well-documented software, go look elsewhere.

Asterisk seems to be the PHP or MySQL of the PBX world.

</rant>

Re:On the subject of Asterisk

[gremln007]

Hey Rob, I've seen you on a lot of the forums. Great work on FreePBX by the way! I have seen a number of folks posting about TCO and needing Asterisk experts, etc. I just wanted to mention TrixBox (formerly Asterisk@Home). It is a great, EASY way to play with Asterisk in a test or even real environment. You can start out using this and then move on to a plain vanilla Asterisk install if you feel the need for greater control. That being said a lot of people use TrixBox (or Asterisk@Home) as-is. TrixBox with its new better update functionality is really great in my opinion. For those interested, check out their site and download an ISO (http://www.trixbox.org/). Also, there is a version you can run from within VMWare. Sorry, I don't have that link handy but you should find it on the Trixbox site. Jonathan

Re:Asterisk really is best bang/buck

[A.K.A_Magnet]

Woops. My bad, The Apache Software License isn't compatible with the GPL:

This is a free software license but it is incompatible with the GPL. The Apache Software License is incompatible with the GPL because it has a specific requirement that is not in the GPL: it has certain patent termination cases that the GPL does not require. (We don't think those patent termination cases are inherently a bad idea, but nonetheless they are incompatible with the GNU GPL.)

And OpenSSL isn't under the Apache Software License but under the OpenSSL License! So there was a problem with OpenSSL too :).

The license of OpenSSL is a conjunction of two licenses, One of them being the license of SSLeay. You must follow both. The combination results in a copyleft free software license that is incompatible with the GNU GPL. It also has an advertising clause like the original BSD license and the Apache license. We recommend using GNUTLS instead of OpenSSL in software you write. However, there is no reason not to use OpenSSL and applications that work with OpenSSL.

Sorry :)

Asterisk-based 200 wireless/wired phone deployment

[ipstacks]

I just deployed an Asterisk phone system powering ~140 wired Polycom phones and ~70 wireless phones covering 31 acres. Here are some tips from what I learned in this process:

1. Pick a capable vendor for each job you outsource. I looked at Asterisk and decided it is too technical for a Asterisk newbie to build a production system, so I called Digium and they referred me to a dCAP certified Asterisk consultant in my area. Knowing Asterisk is one thing, but knowing how to pull off a great install is more than that. Our vendor developed a workbook that covers many parts of a successful deployment, such as reviewing the network (gear, configs, wiring plant), getting the users (names, current extentions, locations . .), getting the users to think about the dial plan and having them understand their satisfaction with the results is directly related to trying to get it right. When we distributed the phones to each desk, the boxes were labeled and sorted on the pallet this helped save a huge amount of time and allowed us to have the furniture installers help setup phones if we wanted too. Staging the phones: pre-configuring them, having the boxes labeled and sorted on the pallet was well worth doing. The wireless phones we signed out to the employees with some other stuff like work shirts. Having the right vendor to walk us through the process was critical.

2. Pilot your install before you deploy it. The environment I was choosing Asterisk for is an automall. Phones are a big part of the business (as with many) and setting expectations is important. We formed a phone users group to have them decide how we wanted to route calls (dial plan), the idea was to get them involved because it is really theirs to use. Some departments were easy and some were not. Sales was essentially create a call groups for the differnt brands we sell and have the operators transfer them to the appropriate group. Service was much more complicated, but having live operators helps a ton. Parts was easy as well, but all of that needs some serious consideration. Knowing you will get it wrong and tweaking it on the fly will happen, do it and move on.

3. We picked Polycom phones and that turned out to be a great choice, the 601's have six "programmable" buttons and great sound quality (handset and speakerphone). The Polycoms have a two port switch built-in and will trunk with the network switch which means the second port on the phone can be a differnt vlan than the phone. So we have them plugged in/wired like this: [network-switch]---[phone]---[computer]. The phones run Cisco CDP, when the switch detects the phone (via CDP) it assigns the phone as a trunk device and allows you to choose what vlan the phone will be on and what vlan the computer port on the phone will be on. Also you can have a differnet vlan if you were to plug the PC directly into the switch. The setup works well and I could go on and on about QoS, edge marking of traffic and PoE issues but I will stop.

4. The FOP (Flash Operator Panel) is a cool thing, but we had to do some customizing for our needs. We looked at Fonalitys HUD, but FOP works great. You can see which phones are ringing, have voice mail (whether it is new or old), transfer calls by drag and drop, monitor the inbound queues and really not have to touch the phone to work the system as an operator. Nicholas, the guy that wrote FOP is an invaluable resource. He was willing to help and has done a great job. I am asking our vendor and am going to make sure he gets paid in some way.

5. Wireless WiFi phones (OUCH): We chose the Hitachi IPC-5000 and Meru Networks for the AP's. Okay I was getting a little cutting edge here, but hey why not?! Lessons:

Meru Networks ROCKS!! They figured out the roaming WiFi thing for sure!

Hitachi IPC-5000's to be determined: it look like either the phones have a high failure rate or we have a bad batch or something. Also it looks like they aren't nearly as durable as say a cell phone/mobile phone (which is VER