Topic: Asterisk on SME7 (me again)

[cyr]

This book seems to be really good Arne, thank you

[xboxer21]

Hi All,
I have sucessfully installed A4SME, I wanted to upload a welcome greeting message using the digital receptionist feature in AMP, I created a wav file  with a sample rate of 8khz. I tried uploading by clicking on the upload button, the page refreshes after uploading, but when I dial *99 to hear the message, I dont hear anything back. Is anyone here having the same problem. Does anyone know the location where the File will be uploaded to, so that I can manually verify if the file is actually being uploaded.

Thanks

[arne]

Yes and no .. I have had parts of the same problem. Because of my own dial plans I first overided (without my knowledge) all of the numbers that begun with a star. When I realiced this I changes my plan a bit (so all digits in the plan had to have et least 5 digits (XXXXX) so there should be no conflict with the default existing numbers) After that I were able to record and play bach voice messages via the telephone adapter and the standard function of the system. I found these numbers by reading the the default configuration file and the commenst there.

You can make some testing using the echo test and line bussy test. (Did not write down there numbers, onlu used them from the configuration file.)

One other useful chech is to go into ssh shell and write the command "asterisk -vvvr". This gived a lot of info about activity.

I got the "automated receptionist function" to work with most of the tested incomming ip telephony lines. This function did not work for one ip telephony vendor that use a bit unusual format on tha register string. This norwegian wendor gave a problem with this function: http://www.televoip.no On the other hand "the line" this American vendor worked very nice:  http://iax.cc There was some bugs, but I still think this American vendor can be higly recomended.

Info just to indicate thet problems to make the automated reseptionist to work also could be related to the format of the register string og some other more vendor specific problems.

I don't remember at the moment where the sound files are stored. (I think I did find them, but just dont remember now. Will leave a msg if I see them..)

Are your Asterisk@SME running bug free ? Mine were until I decided to replace all the default configuration files with new manually edited files. It stopped working then, even though I had tested the files on a "clean" Asterisk@home.

For me Asterisk@SME have worked 100 % and without a bug as long as I used the standard configuration tools "the right way and as expected".

By the way .. has reinstalled it again.

Best reg Arne.

[chris burnat]

they could be here:
[root@kiwi ~]# ls -la /var/lib/asterisk/
total 152
drwxr-xr-x  10 root root  4096 Dec 11  2004 .
drwxr-xr-x  18 root root  4096 Dec 11  2004 ..
drwx------   2 root root  4096 Sep 17 20:42 agi-bin
-rw-r--r--   1 root root  8192 Nov  9 21:31 astdb
drwx------   3 root root  4096 Sep 17 20:42 firmware
drwx------   2 root root  4096 Sep 17 20:42 images
drwx------   2 root root  4096 Sep 17 20:42 keys
drwxr-xr-x   2 root root  4096 Oct  8 16:12 licenses
drwx------   2 root root  4096 Sep 17 20:42 mohmp3
drwxr-xr-x  10 root root 69632 Oct 20 00:10 sounds  <---    
drwxr-xr-x   6 root root 36864 Oct  9 13:37 sounds.po
Thanks to Stephen Noble who just pass this information tonight.

[xboxer21]

Hello Arne,
I have not completely tested by using any VOIP service provider yet, I was trying to simulate an incoming call by dialing 7777 from an extension.
I will try changing the extensions to 5 digits and retest, will let you know how it goes. So have you replaced your iax.conf, sip.conf & extension.conf manually?

Thanks

[arne]

xboxer21 ->

Yes, I replaced the iax.conf, sip.conf & extension.conf with manually edited files on a ordinary Asterisk@home installation. This worked without a problem. Actually I just made a temperary renaming of the orginal files so it was just to rename back again to restore the orginal configuration.

When I tried to do the same on the Asterisk@SME the installation just crached. I don't know why. Could be just by coincidence.

Arne.

[arne]

burnat ->

Thanks for your info. This is from a Asterisk@home as I just have some problems with the Asterisk@SME (but this is due to some rather irregular testing.)

In the Asterisk@home installation I found the sound file from the digital receptionist here:

/var/lib/asterisk/sounds/custom/

Your info helped me to start the search on the right place.


Best reg Arne.

[xboxer21]

Arne,
I only see a audrey.wav file inside /var/lib/asterisk/sounds/custom/ folder. Do you have any other files in your installation of a@h in the same folder?.
By the way I have signed up for Broadvoice, I will configure my asterisk@SME box to work with broadvoice and let you know how it goes.

Thanks

[xboxer21]

I tried modifying the extensions_custom.conf using phpconfig for Asterisk but I get an error message when I try to save the file. Its says Write failed.
Any idea why is this happening?

Thanks

[arne]

xboxer21 ->

/var/lib/asterisk/sounds/custom/  .. only one file inside that one (and then also some 44 byte template ore sometingg named .wav)

"I have signed up for Broadvoice" I have tried them. Technically it worked very nice. No problems. On the other hand, I did not like their security concept. When I maild them and commented this they only said that they did not concider security to be a problem. I do, so I ended my account.

So security was my reason for changing from Broadvoice to iax.cc By the way, the Broadvoice people treated me very nice, and I got all my money back. I guess that Broadvoice technically might be a slightely bether (more stable) service than iax.cc but I like the security concept of iax.cc very much. (You fill up you account with 20 dollar, and if someone is hacking your account it stops at 0 dollar. Really not much to worry about.)

"I tried modifying the extensions_custom.conf .."

I think the reason why this do not work is because this script is maintained by automated routines. If you try to edit the exctensions.conf in the bottom end, I guess there is a bether chance. (At the Asterisk@home and the Asterisk@SME, I believe this is a more static file that can be edited.)

Best reg Arne

[xboxer21]

Hi Arne,
could you please post your config files for broadvoice,or did you use AMP to configure. Is there anything special we would have to do if my asterisk box is behind a router.

Thanks again

[arne]

xboxer21 ->

Broadvoice is for me now history, I don't have that configuration any more, but I think (believe) I used this guide:

http://geekgazette.com/index.php?option=com_content&task=view&id=20&Itemid=26

Special .. Yes, you will have to forward the right ports. I think default is UDP 5060 and all the UDP range 10000:20000

By the way I have now been able to crash my Asterisk@home as well, so it apear, that if one does enough "incorrect configuration" of any Asterisk server it is possible to destroy it. I think the reason that it crashed were that I tried to edit some of those files that is configured using automated procedures (but I will reinstall it again.)

There has been no crash or problems at all neither with the Asterisk@home or the Asterisk@sme as long as configuration and use have been "acording to the book". "To much" experimenting some times makes the need for reinstalling it all, even though there is no "visible" faults. It just stop working.

I think I did the Broadvoice configuration using the automated and manual configuration at Asterisk@home while the Asterisk@SME only worked with the automated AMP configuration.

There is by the way also a sip_nat configuration file if you want to have the internal extension numbers accessable over the internet. Might be a bit dangerous to use and is not required for basic operation.

Arne

[xboxer21]

Arne,
I did exactly what was mentioned in the link you posted, I can now recieve incoming calls, but cant make outbound calls, My guess is, i need to add a dialplan to the extensions.conf, I found a dialplan on the broadvoice website i will give it a shot & post my results later today. So far i haven't done anything that extensive with my config files so my asterisk has not crashed yet.
and thanks for all your help.

[arne]

Yyr go to the outbound routing. For instance if you make a route 01|. you just will have to 01 in front of the telephone number and it will go out (Trough thar certain "trunk".)

The inbound connection use to be more difficult and tricky than the outbound connection, so I think you have passed the most difficult part of it already.

Arne.

[meanlocha]

I much prefer the idea of not loading up an
Asterisk/AMP (A@H) server on top of SME.  It seems much simpler (to me  to run the two independently and portforward the SIP ports to A@H.  I got Broadvoice
working by a) collecting and setting the parameters
correctly from the Broadvoice account b) portforwarding
UDP on ports 69, 5060:5082, 10000:20000 to the A@H server.

Anthony