Open Dental Forum

We ordered our new Asterisk server yesterday. For those who are unfamiliar, it's a software that runs on a Linux computer that lets the computer act as a phone PBX. We will have analog phone lines running in, and then IP phones hooked up to it via our LAN. I'm going to be flying tomorrow and again a few days later. It's a perfect opportunity to reread the Asterisk manual. What I'm really interested in is the part where every incoming phone call gets the details dropped into MySQL. From there, the plan is to let OD grab the data and take action on it. I envision the patient being automatically selected based on caller ID. I see all phone calls being recorded and attached to commlog entries. Every commlog entry could have the start time and the stop time of the phone call. I'm just scratching the surface of what we can do with a phone system that's totally software driven and tied to the dental software. Just scratching the surface.

While you are on that thought...think about the possibility of how the reminder calls can automatically be sent out (using some kind of voice recognition or any other software).

As long as asterisk and opendental databases can talk to each other, I think the possibilities of optimizing the workflow are endless. Possibilities of recording all phone conversations with patients is also an extreme possibility..if and when needed.

The biggest question on the table is..what is the biz value of this integration. As long as our strategy is aligned from that point of view, bringing the integration between voice communication and the practice management software will yield great results.

Cheers,

OpenSource

After spending the last 3 weeks learning Asterisk, learning about IP phones, ordering parts, adding more wiring, etc, I have a few more comments to make. I can clearly see that Asterisk is not designed for end users in a dental office. But a prebuilt and professionally installed Asterisk system would be a good alternative to a PBX in a large office. But I have to admit that we are not pursuing this feature necessarily for the good of everyone. We are doing it because we need it to function as a support call center. We keep our customers in an Open Dental database. So when a call comes in, we do indeed need to use caller id to automatically look up the customer, and then record the phone call as well as the start and stop times. We simply need this power whether or not any dental office finds it useful.

I remember hearing more then once it takes awhile to get used to because their isnt any lines, just calls and you have to park them or something...

Going to have my bootable raid array server with hardware detection working this week. Successfully stayed away from the pc for fear of my wife.

Jordan: I have to agree with you. asterisk takes allot of work to setup. I've had partial success setting up trixbox with Rhino R4FXO card and polycom 501 phones.

Luckily these polycom phones can have their firmware update to use on my current pbx system Talkswitch which has lots of cool features. Just not as advaned as a asterisk.

We can't get our phones to talk to the server. So we have to pay someone to come in to fix it. We'll just look over their shoulder while they work, and then we can do it next time.

I can say from experience that asterisk is fun to tinker with...HOWEVER....

I have installed three asterisk based phone systems sold and supported by a prominent vendor and this is my experience.

1. ALWAYS get your phone cards with HARDWARE echo cancellation technology..ESPECIALLY when using analog lines.
2. Unless cost is an issue, DON'T use analog lines and go with a PRI.

I have been using asterisk as a call center with two of the sites linked together. I know in the long run it will pay off, but there was significantly more time spent setting it up than I expected.

Hi Jordan, which Asterisk distro are you using? Have you tried Trixbox or did you start with 'plain' Asterisk?

I've got it running in a dental office (on the same Linux server as OD) and its simple to add phones etc and have them configure with the extension manager built into Trixbox.

Btw, I'm interested in helping with further integration if you go that route. I think that Asterisk, while still young, is going to go places in all businesses.

I echo the comments of the previous poster re hardware echo cancellation, and also add in case its not obvious, that keeping a couple of telco 'hard lines' is very desirable. (See other comments in the much older thread at http://freedental.forumco.com/topic~TOPIC_ID~804.asp and similar.)

cheers, michael

I shied away from Trixbox. I tend to be more of a purist, appreciating the power of lower level interfaces. Perhaps I made a mistake. Especially now that I realize that no dental office is going to be able to set up Asterisk on their own.

All I know, if you can afford to buy a new switch, get one with power over ethernet so your phones can get power from your cat5e cable... Very Cool.

I have decided to keep referring my pals at dfwphones to do phones so have fun with asterik you guys. I will use it maybe one day for myself, but not today.

jordansparks wrote:I shied away from Trixbox. I tend to be more of a purist, appreciating the power of lower level interfaces. Perhaps I made a mistake. Especially now that I realize that no dental office is going to be able to set up Asterisk on their own.

- A good move in general, but I've found that Trixbox is plenty low level enough! There are too many value adds like FreePBX (with its easy extension config and extensive add in modules) and the Endpoint Configuration manager that auto-detects your phones and assigns them their passwords and all other settings when they boot.

You can still get to the underlying Asterisk settings if you feel a need, but when you just want a working phone system at the end of the day, I find I keep coming back to Trixbox.

Its also a good place to start because there are LOTS of people using it, so if you hit a problem, chances are someone has solved it.

I could give you a long list of things I don't like about Trixbox, but after messing with this on and off for 2+ years now, IMO its the best starting point. The other route I keep looking at is stripped down versions like http://astlinux.org and http://askozia.com/ which can run on a machine with no hard disk like the Soekris or Wrap boards. However, AstLinux requires too much config file programming for me and Askozia is still a bit young (but coming up fast!).

You won't regret trying Trixbox - and I'll wager it won't take you long to get a working phone system.

Btw, we've been using Vitelity.com as a provider with good success. I still like Telasip.com a lot, but they cost a lot more than they used to.

MIke:

I've been trying to get my trix box off the grownnd with very little success.

I've got a Rhino R4fxo for my 4 pstn lines going into the building. Its tough at home to get my polycom 501's to connect via my linksys wrt54g router.

I've followed several online manuals and tutorials and still get stuck.

anyways, if you would like to lend a hand please let me know. I do dentistry for a living but love linux.

I just wish there was an ISO package that installed my hardware and worked from the start.

For example, In about 30 minutes I installed untangle.com's firewall/webfilter/router/antivirus/spam blocker without any problems.

I was using smoothwall, but really like untangle and how easy they have made using their linux distribution.

anyways, let me know what you think.

I've found that the Trixbox ISO does install nicely from the start (after it erases EVERYTHING on the machine), but it does take some tweaking to get it going as there are just SO many flavors of phone line adapters (like your Rhino card) and phones. When you throw in an existing network, a router, etc., the things that can go wrong will multiply fast.

I looked at your posts on the trixbox.org forum, and it appears that between your router/network config, the phones and the Rhino card that you're juggling several variables at once. Its understandably a frustrating situation!

Here's a couple of suggestions / comments:

1) using dhcp on your linksys is fine. The only tweak I'd suggest is to have it pre-allocate/assign a fixedIP for your trixbox so that it always gets the same IP. (That will happen anyway most of the time, but not if you reset or power cycle the linksys, which, if the trixbox does get a new address will stop the phones from working)

2) I described how I got my polycom 501 working with TB 2.2 here: http://www.trixbox.org/forums/vendor-mo ... dpoint-mgr
Remember that the phone tftp server has to be set to the IP you've pre-assigned to the Trixbox server, and that you'll have to re-run the setup-polycom (or is it polycom-setup?) if you've changed the trixbox's IP. (Also, this all assumes that all the devices are on the same network - i.e. not the trixbox at work and the phones at home - that setup scenario is possible but the auto-configuration gets more complicated).

3) I suggest you concentrate on getting two phones working first - i.e. so you can call from one to the other, leave and retrieve a voicemail etc.

4) Once you have the phones working, add a VOIP service and get that going too. This is where you may hit issues with one-way audio etc. which are usually related to the firewall/router (i.e. the linksys box).

5) When that's all working, tackle the Rhino card. Test dialing out and dialing in separately and concentrate on one at a time (I suggest going with the one that seems to be working best!) I've seen reports of problems with the Rhino cards that suggest various software patches etc. to fix niggling things, but FXO cards are a pain at the best of times, so its just as likely to be something simple. E.g. see http://www.trixbox.org/forums/trixbox-f ... ap-channel for a simple problem I had with dialing out because the Sangoma card dialed too quickly.

You could swap (4) and (5), but (4) generally happens quickly and its nice to have some success!

I may have some time next week to help out for an hour or so. Beforehand it would be best if you can enable remote access to the box so I can drive it directly. Hopefully you'll have some luck with the above before then!

Thanks for the tip on untangle.com. I'm a devotee of m0n0wall, but I'll check out untangle because the added spam/anti-virus features would be a bonus!

good luck! michael

Mike:

Thanks for the tips. i'll get working on this again.

I was able to get a softphone and the polycoms to leave voice messages in the past.

By the way, what's the port I need to forward for remote access?

Talk to you soon.

Mike:

I have a soft phone setup extension 260 and 1 polycom setup extension 250. I can dial each other but I get the answering machine stating "the person on extension 250 is unavailable, please leave a message"

I can leave a message in either direction and retrieve voice mail, but the phones won't ring.

also, I think its port 22 for remote admin.

Mike:

Here is a copy of my post in trixbox forums:

Hello all:

I have a polycom 501 setup. trunks, extensions and outbound routes setup. HOwever, i cannot dial out.

I get the " all circuits busy" recording.

Here is the output of: asterisk -vvvr when I try to dial out:

any help please??

[root@asterisk1 ~]# asterisk -vvvr
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.2.22, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.22 currently running on asterisk1 (pid = 2784)
Verbosity is at least 3
-- Executing Macro("SIP/250-09e2f700", "dialout-trunk|2|191098735||") in new stack
-- Executing Set("SIP/250-09e2f700", "DIAL_TRUNK=2") in new stack
-- Executing Set("SIP/250-09e2f700", "_NODEST=") in new stack
-- Executing Set("SIP/250-09e2f700", "DIAL_NUMBER=191098735") in new stack
-- Executing Set("SIP/250-09e2f700", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/250-09e2f700", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/250-09e2f700", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,8)
-- Executing Set("SIP/250-09e2f700", "GROUP()=OUT_2") in new stack
-- Executing Macro("SIP/250-09e2f700", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/250-09e2f700", "user-callerid: device 250") in new stack
-- Executing GotoIf("SIP/250-09e2f700", "0?report") in new stack
-- Executing GotoIf("SIP/250-09e2f700", "0?start") in new stack
-- Executing Set("SIP/250-09e2f700", "REALCALLERIDNUM=250") in new stack
-- Executing NoOp("SIP/250-09e2f700", "REALCALLERIDNUM is 250") in new stack
-- Executing Set("SIP/250-09e2f700", "AMPUSER=250") in new stack
-- Executing Set("SIP/250-09e2f700", "AMPUSERCIDNAME=Dr. Aguilar") in new stack
-- Executing GotoIf("SIP/250-09e2f700", "0?report") in new stack
-- Executing Set("SIP/250-09e2f700", "CALLERID(all)=Dr. Aguilar <250>") in new stack
-- Executing Set("SIP/250-09e2f700", "REALCALLERIDNUM=250") in new stack
-- Executing NoOp("SIP/250-09e2f700", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/250-09e2f700", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/250-09e2f700", "Using CallerID "Dr. Aguilar" <250>") in new stack
-- Executing Macro("SIP/250-09e2f700", "record-enable|250|OUT") in new stack
-- Executing GotoIf("SIP/250-09e2f700", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/250-09e2f700", "recordingcheck|20070818-200846|1187482126.16") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20070818-200846|1187482126.16: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/250-09e2f700", "No recording needed") in new stack
-- Executing GotoIf("SIP/250-09e2f700", "0?skipoutcid") in new stack
-- Executing Set("SIP/250-09e2f700", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/250-09e2f700", "outbound-callerid|2") in new stack
-- Executing GotoIf("SIP/250-09e2f700", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/250-09e2f700", "REALCALLERIDNUM is 250") in new stack
-- Executing GotoIf("SIP/250-09e2f700", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/250-09e2f700", "USEROUTCID=") in new stack
-- Executing Set("SIP/250-09e2f700", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/250-09e2f700", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/250-09e2f700", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/250-09e2f700", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/250-09e2f700", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/250-09e2f700", "CallerID set to "Dr. Aguilar" <250>") in new stack
-- Executing GotoIf("SIP/250-09e2f700", "0?nomax") in new stack
-- Executing GotoIf("SIP/250-09e2f700", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/250-09e2f700", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/250-09e2f700", "OUTNUM=w191098735") in new stack
-- Executing Set("SIP/250-09e2f700", "custom=ZAP/1") in new stack
-- Executing GotoIf("SIP/250-09e2f700", "0?customtrunk") in new stack
-- Executing Dial("SIP/250-09e2f700", "ZAP/1/w191098735|300|") in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Goto("SIP/250-09e2f700", "s-CHANUNAVAIL|1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing NoOp("SIP/250-09e2f700", "Dial failed due to CHANUNAVAIL - failing through to other trunks") in new stack
-- Executing Macro("SIP/250-09e2f700", "outisbusy|") in new stack
-- Executing Playback("SIP/250-09e2f700", "all-circuits-busy-now|noanswer") in new stack
-- Playing 'all-circuits-busy-now' (language 'en')
-- Executing Playback("SIP/250-09e2f700", "pls-try-call-later|noanswer") in new stack
-- Playing 'pls-try-call-later' (language 'en')
-- Executing Macro("SIP/250-09e2f700", "hangupcall") in new stack
-- Executing ResetCDR("SIP/250-09e2f700", "w") in new stack
-- Executing NoCDR("SIP/250-09e2f700", "") in new stack
-- Executing GotoIf("SIP/250-09e2f700", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/250-09e2f700", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/250-09e2f700", "5") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/250-09e2f700' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/250-09e2f700' in macro 'outisbusy'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/250-09e2f700'
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1

Mike:

I have made good progress.

I modified the zapata.conf file and this allowed me to dial out.

So, I can dial out from my polycom 501. Cannot receive yet.

In free pbx I did notice that inbound routes is not configured. I think this might have something to do with this but not sure.

I can only dial from the first FXO port. I moved the telephone line to port 2,3,4 and could not dial out of those.

Also, from X-lite softphone I cannot dial my polycom 501 extension. Neither can I do it from my polycom over to the x lite.

The voice mail does work on both xlite and 501.

My next task is to create another extension and hookup a second polycom 501 and see if it can dial to the first polycom.


well.. this is lots of progess for me. i'm excited. learning lots about linux thats for sure.

talk to you soon.

Congrats on getting it going further.

Yes, without an inbound route you won't get incoming calls. Its simple for a zap channel inbound though, so just set one up and you'll be close.

BUT ... I'm still troubled that you can't call from one extension to another, i.e. that you only get voicemail when you do. This means that for some reason trixbox can't find the phones to make them ring so it defaults back to voicemail. You should be able to use the softphone to the polycom and vice versa.

If you can paste the log trace when you try to call extension to extension and/or email/message it to me via this forum, I'll take a look. (Note that I'm offline from Wed thru Sun so I might not answer until Mon/Tue).

Note that until you get extension to extension working, inbound is not likely to work either.

cheers, michael

OK here is the output of asterisk -vvvr

[root@asterisk1 etc]# ztcfg -vvv

Zaptel Configuration
======================


Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)

4 channels configured.

[root@asterisk1 etc]# asterisk
Asterisk already running on /var/run/asterisk/asterisk.ctl. Use 'asterisk -r' t o connect.
[root@asterisk1 etc]# asterisk -Rvvvvvvvvvvvvvvv
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.2.22, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.22 currently running on asterisk1 (pid = 2920)
Verbosity was 1 and is now 15
asterisk1*CLI> zap show channels
Chan Extension Context Language MusicOnHold
pseudo from-zaptel en
1 from-zaptel en
2 from-zaptel en
3 from-zaptel en
4 from-zaptel en
asterisk1*CLI>
[root@asterisk1 asterisk]#
[root@asterisk1 asterisk]# clear
[root@asterisk1 asterisk]# nano zapata.conf
You have new mail in /var/spool/mail/root
[root@asterisk1 asterisk]# asterisk -vvvr
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.2.22, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.22 currently running on asterisk1 (pid = 2920)
Verbosity is at least 15
-- Executing Macro("SIP/250-09c2f800", "vm|260|DIRECTDIAL") in new stack
-- Executing Macro("SIP/250-09c2f800", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/250-09c2f800", "user-callerid: device 250") in new st ack
-- Executing GotoIf("SIP/250-09c2f800", "0?report") in new stack
-- Executing GotoIf("SIP/250-09c2f800", "0?start") in new stack
-- Executing Set("SIP/250-09c2f800", "REALCALLERIDNUM=250") in new stack
-- Executing NoOp("SIP/250-09c2f800", "REALCALLERIDNUM is 250") in new stack
-- Executing Set("SIP/250-09c2f800", "AMPUSER=250") in new stack
-- Executing Set("SIP/250-09c2f800", "AMPUSERCIDNAME=Dr. Aguilar") in new st ack
-- Executing GotoIf("SIP/250-09c2f800", "0?report") in new stack
-- Executing Set("SIP/250-09c2f800", "CALLERID(all)=Dr. Aguilar <250>") in n ew stack
-- Executing Set("SIP/250-09c2f800", "REALCALLERIDNUM=250") in new stack
-- Executing NoOp("SIP/250-09c2f800", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/250-09c2f800", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/250-09c2f800", "Using CallerID "Dr. Aguilar" <250>") in new stack
-- Executing Set("SIP/250-09c2f800", "VMGAIN=") in new stack
-- Executing GotoIf("SIP/250-09c2f800", "1?s-DIRECTDIAL|1") in new stack
-- Goto (macro-vm,s-DIRECTDIAL,1)
-- Executing NoOp("SIP/250-09c2f800", "DIRECTDIAL voicemail") in new stack
-- Executing Macro("SIP/250-09c2f800", "get-vmcontext|260") in new stack
-- Executing Set("SIP/250-09c2f800", "VMCONTEXT=default") in new stack
-- Executing GotoIf("SIP/250-09c2f800", "0?200:300") in new stack
-- Goto (macro-get-vmcontext,s,300)
-- Executing NoOp("SIP/250-09c2f800", "") in new stack
-- Executing VoiceMail("SIP/250-09c2f800", "260@default|u") in new stack
-- Playing 'vm-theperson' (language 'en')
-- Playing 'digits/2' (language 'en')
-- Playing 'digits/6' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'vm-isunavail' (language 'en')
-- Playing 'vm-intro' (language 'en')
== Spawn extension (macro-vm, s-DIRECTDIAL, 3) exited non-zero on 'SIP/250-09c 2f800' in macro 'vm'
== Spawn extension (macro-vm, s-DIRECTDIAL, 3) exited non-zero on 'SIP/250-09c 2f800'
asterisk1*CLI>

Hmm. It does appear to go directly to voicemail without even trying to ring the extension. E.g.:
suggests that its going directly vm. However I could be completely off base and its hard to guess more without seeing your configuration (esp your extension configuration).

Note that there is likely a huge amount more debug info in the file /var/log/asterisk/full. If you use:
when you try to dial, you'll see the debug go by on your ssh screen. Use Ctrl-C to stop the trace.

As a side note, it does appear that your Rhino card is correctly configured for all four ports. If you add a channel for each one, you should be able to dial out on them individually. Most likely the default config you now have will only dial on line 2 if line 1 is busy (and line 3 if line 1 and line 2 is busy etc).

I'm offline for a few days after today, but I'll IM you the contact details of a friend who does this (and other computer things) for his day job. I'm sure he can help you remotely.

cheers, michael

Good article on this topic in the September 4, 2007 issue of PC Magazine. There are several vendors offering "plug and play" options that would allow an average dental office to be up and running very quickly with more features than they are likely to use.

We switched to Fonality PBXtra call center edition a few months ago and have been pretty pleased so far. It is still a young industry but the cost was a fraction of a comparable Cisco or Nortel system. It integrates with Outlook as well as SugarCRM and Salesforce.com. It could also be integrated with OpenDental with some coding from Jordan I suspect...

Best of luck,

Scott Wellwood - COO
EDI Health Group.
http://www.dentalxchange.com
scott@dentalxchange.com

HEy guys I came across this asterisk alternative, windows based and thought I would send you the link. i'll be testing it soon.

http://www.3cx.com/