Where is that again??? It is so valuable!

Tbanks Ward, for all you do for us Nerds!!!

" Once you’re logged in, issue the following commands to fix the initial voice prompts with our Stealth Autoattendant:

cd /var/lib/asterisk/sounds/custom
mv nv-greeting.wav nv-greeting.wav.bak
mv nv-menu.wav nv-menu.wav.bak "

I think I fixed my problem by substituting "gsm" for "wav" in the above instructions. (just an FYI)

everything runs more or less correct and I can still log into asterisk-cli.

Even tried redoing the installation, with the same problem. Everything else upgrades and installs successfully except asterisk-cli.

Is this normal? Any ideas?
Thanks in advance

[WM: Restore backkup. Then try again.]

[WM: This is a current limitation in freePBX. You might want to post something on freePBX.org. The more requests they get, the quicker it will get addressed.]

if (strlen($thenumber)10) :
exit ;
endif;

>> Hello again, I changed the beginning of the script as follows and >> it has cleared up my issue. Hopefully this will help someone else.

>> 10) :
>> $thenumber = right ($thenumber, 10) ;
>> endif;

[WM: See #20 above. WordPress has problems with some of the PHP code insertions for obvious reasons. The main one is that it, too, is written in PHP.]

Thanks,
Jeff

Cheers!

P.S. Keep up the fantastic work on trixbox!!!

[WM: Here’s the article. Be sure to read the section on "Making VMware Keep Correct Time." Better yet, read the whole article and also the comments which follow it… especially #10, 17, and 21.]

Telco provides a valid CLID. The Telco Database is more complete than either Google or Anywho. I’d think the flow should go something like this:

1. If there is a CLID name from telco, the call should no be processed through Google or Anywho, but should be processed against Asteridex and the Internal Asterisk Phone Book.

2. If there is no valid entry in either of the above sources, the telco CLID should be passed downstream.

3. If there is an entry in the above sources, then the CLID should be changed to reflect that.

[WM: Good suggestion. We’ll take a crack at it for the next version. Our initial thought was that the lookup probably would not be applied at all to "phone company" trunks, but there may be situations in which the scenario you propose above will work better. Thanks.]

<?php
$thenumber=$_REQUEST[‘thenumber’];

function right ($str, $howManyCharsFromRight)
{
$strLen = strlen ($str);
return substr ($str, $strLen – $howManyCharsFromRight, $strLen);
}
if (strlen($thenumber)<10) :
exit ;
endif;

if (strlen($thenumber)>10) :
$thenumber = right ($thenumber, 10) ;
endif;

Very cool as usual. I just wanted to point out a new toy that I have found to be very useful. SSL-Explorer http://www.sshtools.com/showSslExplorerCommunity.do A very powerful clientless SSL based VPN. Very good for giving remote access to web apps, trixbox, VNC, rdesktop, etc, without exposing anything more than port 443 to the internet. You can even create custom tunnels to alow SIP connection for a softphone. Several authentication methodes are available. Enterprise license add extra features like a full VPN tunnel and more authentication options, including active directory. Easy install via rpm or windows installer. Server requires a recent JRE. Client side needs a semi-recent browser with a recent Java plugin. Also available as a vmware prebuilt appliance, http://www.vmware.com/vmtn/appliances/directory/543

And thanks for all the goodies.

[WM: Try downloading the app again. There was a problem in the initial release.]

Thanks!

I absolutely love the site and have learned a lot in a few short days. Thanks for the great work!

One topic I’d like to see if you can cover is least-cost routing. If I sign up for multiple VOIP providers, I’d like to be able to direct calls based on their rates, which may differ based on day of week, time of day, and destination. In addition, if I have a fixed-time plan (e.g., 500 minutes per month) it would be great to have Asterisk stop using the trunk after that point. Setting the trunk priority in the outgoing routes is a start, but it would be nice to see something more robust. Any interest in tackling this in a future column?
Thanks, ed

[WM: Take a look at last week’s article. VoicePulse offers a service such as you describe.]

I’ve setup CallerID Trifecta on my trixbox and it works for 2 of my DID’s but not the 3rd. All are from VoIPStreet. The 2 it works on are pay-per-minute and the one it doesn’t is a flate rate. Any idea what could be going on? Here’s some logfile output both from the same caller.

Working –

Verbosity was 1 and is now 3
— Executing Set("SIP/2guys-669xxx-09961d58″, "FROM_DID=281657xxxx") in new stack
— Executing Gosub("SIP/2guys-669xxx-09961d58″, "cidlookup|cidlookup_3|1″) in new stack
— Executing Set("SIP/2guys-669xxx-09961d58″, "CALLERID(name)=E P S Software Corp") in new stack
— Executing Return("SIP/2guys-669xxx-09961d58″, "") in new stack

Not working –

— Executing Set("SIP/2guys-669xxx-099c9700″, "FROM_DID=1832252xxxx") in new stack
— Executing Gosub("SIP/2guys-669xxx-099c9700″, "cidlookup|cidlookup_3|1″) in new stack
— Executing Set("SIP/2guys-669xxx-099c9700″, "CALLERID(name)=") in new stack
— Executing Return("SIP/2guys-669xxx-099c9700″, "") in new stack

[WM: Check to be sure both are passing the CallerID number in the same format. Perhaps the "not working" is adding a 1 or +1 to the incoming number.]

Re: 2 DID problem:

Have you tried to differentiate your trunk by adding the number at the end of your registration string?

In my case I own multiple DIDs from the same provider in sip and got many wasted hours to solve incoming route… but when I tuned my registration string and defined specific inbound routes, it worked:
username:passord@ilovenerdvittles.com/my_did_number
johndoe:password@voipautria.com/0123456789

Re: my problem now

I upgraded from 1.2.3 following the 3 PASS recommendation, but just after rebooting, everything seemed fine except the fop.
Now i get msg icon on all extensions, and the down arrow icon for all trunk and extension..

[WM: Trying resaving your trunks and extensions in freePBX. This worked wonders with the original PBX-in-a-Flash utility as well.]

I forgot to ask you about something which is a big problem for all of us who have multiple DIDs from the same provider. I think this would be an issue for many of your readers – so perhaps there is a solution or a workaround…

Here is the scenario…

I have a DID in Melbourne and another in Sydney (Australia for the uninformed) – both of these DIDs have unique numbers and are serviced by the same provider.

Unfortunately the provider only operate a SIP trunk (no IAX) and has the same host registration details for each DID – the only things which differ are the username (number) / passwords for the respective accounts.

The problem I am having is in setting up BOTH trunks to work independently on the TrixBox.

When they are set-up – all is fine – both are registered but unfortunately with same IPs and port numbers (because they have same host). This creates the problem – because TrixBox is unable to differentiate the two trunks when receiving incoming calls. One would expect that the CID for the individual DIDs would be sufficient for TrixBox to know which trunk is being used. The result is that when you try to ring one trunk from the other (or visa versa) the response is a busy message – telling you the party you are calling is on the phone – because it is calling the trunk which instigated the call.

After researching this behavior – I found the following …

“Registering multiple SIP accounts with one SIP provider has been a nightmare in Asterisk. Or, rather, still is. The match-on-IP scheme for peers is a hack to handle registrations, but not a very good hack. If you register for multiple accounts, the incoming calls will all match the same peer. A poor solution.”

… so is there a workaround for this behavior – that can be implemented in the TrixBox?

OR have you already a way of handling multiple SIP registrations to the same host?
Hope you can point all of us with this problem in the right direction!

I’ve stuck out on most of the forums and would be very happy if there IS a solution to this problem.

[WM: If I were you, I’d try posting this on Rob’s freePBX Forum. He may have some ideas particularly since he’s from Australia. Good luck!]

Thanks for the VMware process priority tip! It will keep Trixbox at the front of the processor cycle queue.

Just another bit of feedback for you…

1. The Zork application appears to have all the modules installed – BUT it is not working on completion of the default install of the freepbx 2.2.0. Unless I broke something (I hope not) your input would be most welcome in solving this…

2. In the TrixBox – configuration and administration window – if you select the ENDPOINT MANAGER – it is broken (again in my default install). There appears to be a problem in the trixbox.php code – it hardwires the default mySql root password – so it you change it (as recommended in the freepbx 2.2.0 upgrade) – it creates an undefined function call.

[WM: Thanks for your comment. The MySQL password should NOT be changed on a TrixBox system unless your Asterisk server is in a public area. It causes all sorts of problems.]

[WM: Not yet.]