Gentlemen,

While you figure out what is wrong with the script, would you mind if I posted this to the Wiki to save some other unfortunate soul the frustration of having the expensive and highly lauded device they have purchased for use with linuxmce not working? I mean, I am thrilled that it works for some or most, but it did not work here and took quite a bit of time to figure it out--and it did not have to do with changing the mac range. Unless I hear otherwise I will go ahead, but please let me know if a fix is nearing completion.

Many thanks for all your help.

Best regards.

I found one other post on this in the forum from last Fall which no one seems to have followed up on. see http://forum.linuxmce.org/index.php?topic=6575.0 The Outlook sync program listed as Outlooksync.msi is missing from the /usr/pluto/installers directory--although there is a link to it on the admin home page. Does anyone have this file so that it could be posted here or in the wiki? It seems that it used to be included and used to work. see http://wiki.linuxmce.org/index.php/OutlookSync So far, the importcontacts.zip which is in the installers directory does not work--repeatedly crashing.

The object here is not so much to use Outlook but to get an idea about how to get contacts into the mysql database without having to do them one by one. I'm kind of surprised that there has been so little interest in this.
Any help or suggestions greatly appreciated.

ok

Problem solved:

So that no other 7970 users will have to endure what I had to in order to get this working, here is the simple solution that our friend from Orange County either refused to divulge or simply did not know the answer to:

Linuxmce automatically sets up the Cisco 7970 but apparently there are bugs that prevent it from completing this setup occasionally. In my case,everything worked except that you could not call the phone from an internal extension. What you need to do to get this to work:

1) In the web admin got to Wizard\Devices\phones and then scroll down to where the Cisco 7970 is listed in the description column. The bolded number to the left is the pluto device number that has been assigned by the setup script copy that as well as the info in the phone type and phone number boxes.

2) Go to Advanced\Configuration\Phones setup in the column. Under the FreePBX logo you will see "Extensions" click on that. On the right side of that page will be the list of extensions. If you do not see an extension with with the phone number you copied in step 1, you will need to add it.

3) In the drop-down box next to the word "Device" choose, Other (custom) Device, and hit submit. This will take you to a screen that will allow you to fill in various fields.  For the
Display Name put whatever you want --I put Cisco 7970
Outbound CID put something like "pl_142" <210> where 142=the pluto id number from step 1 and 210=the phone number from step 1. The formatting with " and <> needs to be included.
Under Device Options next to Dial put SCCP/210 where SCCP=phone type from step 1 and 210=phone number (this is the secret sauce to getting asterisk to recognize that it is a sccp extension it is dialing.
Under voice mail and directory you will need to select "enable" from the drop down box as there is an apparent bug--at least in my version of FreePBX which is too much trouble to fix.

All the other fields can be left blank or default.

Now hit submit. It will register you SCCP extension and you will see it on the right side of the page.

Click on it. Go back and disable the Voicemail in FreePBX so that you use the linuxmce voicemail instead.
Hit submit again--this time it will accept it without balking that you do not have voicemail.

Now at the top of the page you will see an orange box on a blue background "apply configuration changes" click on that and apply the changes in the next screen.

Go to another extension and call your SCCP Cisco 7970 phone. It should be working now.

Please note, this is the quick and dirty fix using SCCP based on my particular problem. From everything I have read, SCCP is fraught with problems and SIP is the way to go with asterisk--but requires that you either buy a license for the firmware or order the phone with SIP. Perhaps the next release of linuxmce will support SIP configurations of this phone as well.

Happy dialing

OK, well thank you for your input.

I have tried what you describe several times (3 to be exact) with the same result: everything works except for calling the cisco 7970 extension (210 in this case) from an internal extension on the asterisk server. In other words, extensions 200 through 209 can NOT call 210 (the Cisco 7970)  They get the message "Your call can not be completed as dialed." But 210 can call 200 through 209, New York, Switzerland, any where in the world with no problems.

I'm not quite sure why repeating the steps you have suggested again and again is the only solution since we have repeated evidence that it does not work in this situation. Albeit, it should.

Does anyone have the experience to suggest why asterisk is telling the caller "Your call can not be completed as dialed?" in this situation? The sccp.conf file looks fine. Conversely, does anyone have a copy of how their Cisco 7970 extension is set up in FreePBX that they would post?

Thanks in advance.

Well thank you for the information.

My experience with non pnp phones in linuxmce is that you do need to manually set it up and I have been able to do this relatively easily, which is why I have been asking about how the SCCP page is configured so that I can take care of that first and see pretty quickly if that fixes the problem. If not, I will at least have more info from asterisk as it tries to dial the extension.

I agree that there is something wrong, but none of the things you mention have been changed from the default, and everything else is working fine so I am at a loss about how to approach the problem other than follow up on my last post and see if it is a bug in FreePBX.

However, I do appreciate that you did respond to some of my queries, at least I know what it is supposed to do with the same freepbx that I have, although I am a bit confused about your last "No" as to whether you see an extension set up for the 7970 or you are unwilling to copy or attach that page to this forum.

Again, any help from anyone greatly appreciated as this is quite baffling.

OK. so when you go to Advanced\Configuration\Phone Setup from within the Web Admin on the upper left hand corner of that frame under the FreePBX logo, what version does it say you are using?

If you click on Extensions Under the "Basic" heading on the left under that logo, do any of your cisco 7970 phones show up on the right side? If so, when you click on the link to the Cisco 7970, what does it tell you about how it is set up? Is it a custom phone or something else? Would you be willing to copy or attach that page to this forum as a guide for other Cisco 7970 users?

Many thanks for any help you can offer.

Well, that is good news. Would you mind posting what your freepbx page for that extension looks like in 7.10?

I'm starting to think it is a freepbx problem because when I go in and try to add an extension manually, on submit, I get:

"Error: Trying to write null voicemail file! I refuse to contiune!"

Googling this suggests that this is a bug in one of the updates:

http://www.trixbox.org/forums/trixbox-forums/help/problems-creating-extension

This would suggest that when the script tries to add the 7970 as an asterisk extension it hits an error that is undetected on the UI and just dies.

I am using:

FreePBX 2.2.0beta3 on 192.168.80.1

Which freepbx are you/should I be using in 710? Perhaps I upgraded freepbx when I shouldn't have? I remember there were alerts to upgrade some time ago within the interface. If so, is there any way to roll back to the version I should be using from within the files in linuxmce or elsewhere?

If this is the problem, it would be good to share it with the other unsuspecting upgraders out there.

Again, many thanks to everyone for their input.

Thanks. Tried your suggestion but got exactly the same result:

No extension in FreePBX--but it does show up in linuxmce as a phone;
can dial out and call other extensions but when you try and call it (extension 210) you get the voice message "your call can not be completed as dialed"

Is there a way to force the linuxmce list of phones to sync with FreePBX list of extensions?

Or, do you have a page you could attach to this post showing how it should be set up in FreePBX and I will add it manually?


BTW, when I do a sccp show devices in the asterisk CLI I get:

NAME             ADDRESS         MAC              Reg. State
================ =============== ================ ==========
SEP001B54525B8F  192.168.80.227  SEP001B54525B8F  Ok       

which proves that it has registered with asterisk.

Orbiter also works great.

Just need the rest of the extensions to be able to dial it and I am home.

Any suggestions greatly appreciated.

No I did that (the got religion part of the post). the MAC address has been added to the template and the phone works fine for everything except receiving calls from other extensions on Asterisk. Do you have a sample extension page from freePBX of the how the Cisco 7970 is installed as an extension? I will just add it manually if I know how to configure it.

Many thanks for your reply.

Hi,
I was under the impression that the extension would be automatically registered in FreePBX (as the other pnp phones are) but it is just not there. Since it is SCCP I am unsure about how to add it manually as there is no default for SCCP. Could anyone please give me a hint?

The problem is that my other SIP extensions can not dial the 7970 because (I think) it is not showing up in FreePBX as an extension. I can dial out and can dial other extensions from the Cisco 7970--they just can't dial me. BTW, Asterisk shows it as registered and sccp.conf shows all the info correctly.

Other than that, once I got religion and actually added my MAC range to the template as described in the Wiki, the script seemed to work flawlessly. Did something just not run?

Any help greatly appreciated.

After playing around with spandsp I did not find it that reliable. I tried the IAXMODEM and hylafax approach and it seems to work much better. The real problem for me was getting it set up correctly between linuxmce and freepbx as there are too many instructions on the web that caused me to unnecessarily or incorrectly duplicate files resulting in a buggy IAXMODEM at first.

The view from 30,000 feet:

1. Use Linuxmce to create an IAX2 extension from Wizard Devices Phones by selecting IAXy adapter as the type--this will create the freePBX extension automatically.

2.use apt-get install to install the IAXMODEM software. Create a file named ttyIAX0 in /etc/iaxmodem that looks like this:

device /dev/ttyIAX0
owner uucp:uucp
mode 660
port 4570 (you want it to be 1 or more above 4569 which it binds to)
refresh 1800
server 127.0.0.1
peername (whatever number linuxmce created)
secret the (same number as the peer name)
cidname (Whatever you want to show up when receiving a fax)
cidnumber (your fax phone number)
codec ulaw (if you live in the US)

3. Go to freepbx and on your fax extension change the port to 4570 from the default--depending on your context, you may want to change that there as well

3. use apt-get install hylafax-server to get the server installed

4. from the command line run "faxsetup" this will setup hylafax and then ask you if you want to add a modem. Say "yes"

5.In the modem config type in ttyIAX0 when it asks you where your modem is--go through the rest of the steps answering the default except where you need to fill in specific info about your location, phone number etc. it will test you modem and set it up for you as well as create all the files to have it automatically start on boot.

6. create a file called FaxDispatch in /etc/hylafax that looks like this:

FILETYPE=pdf;
SENDTO=(your email address);

7. Go to the tutorial on linuxmce wiki about postfix (just search on that word); I just followed the instructions and it worked perfectly. I found postfix was great for emailing me the faxes.


8. Go back to freepbx and set up an inbound route to the IAX2 extension you have created--In my case I used a pstn trunk and only let it go to that extension number so I didn't need to do any fax detection. If you need fax detection there is a lot of info out there on how to do it with zaptel or nvdetect. Also create an outbound route for anything dialed from that extension if you need to.

9. Reboot the machine.

10. login to machine and do a sudo -s to become root. type asterisk -vvvvvr to get to the asterisk CLI. Type "iax2 show peers" to see that your modem is registered and hylafax is running. Send yourself a fax (or have a friend do it) so you can watch the fax come in. If everything went well, the phone will ring and connect to that extension then nothing will happen until it disconnects. You will then receive an email with your fax attached.

If anyone has more to add or sees some mistakes here, please add them as a reply. Eventually, this would be a good tutorial topic for the wiki...

Thank You for your response.

On LinuxMCE 7.10 to get rxFax and txFax (the applications that spandsp installs) into asterisk I had to:

apt-get install asterisk-app-fax

then if you do a:

asterisk -vvvvvvr
core show applications

they are there.

Apparently 7.10 left them out of the package. From this point I believe that you can choose whether to go with Asterfax or Hylafax or something else. I will try and post if there are any problems.

I am really enjoying the capabilities of Asterisk in linuxMCE. One problem has been discovering where the Asterisk application directory (usually /usr/src/asterisk/apps) is so that I can add functionality to the Asterisk app such as faxing through my Digium card connected to a pots line. Can anyone enlighten me as a newbie how Asterisk is built into LinuxMCE and how I might patch the existing Asterisk source with spandsp? I searched the forum and found only one 2005 reference which suggested that it would be built into future releases. However when I go the asterisk CLI and do "show applications" it is not listed. Many thanks for any help in advance.

I tried it and it worked. However the video quality is much worse than just using the the html page to pull it up.
Is there a way to get the "speak" button to work? When I click on it the image goes away and I get an "End Broadcast" button. When I click on that I get a "call dropped normal clearing" notification.

Thanks.