SOLVED IncrediblePBX AsteriDex Click-to-Dial

ericlee1

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Hi,
I can't figure out how to use Asteridex4, I installed the Incrediblepbx whole enchilada, which comes with asteridex pre-installed. I would assume it should work out of the box, but that doesn't seem to be the case.

when I click on an entry, I get a pop up saying calling SIP/701 .. and then the popup disappears, but my EXT never rings.

what steps am I missing to set this up, I'm using SIP extentions, also tried with PJSIP but no luck.

UGH..
:helpsmilie:
 

MGD4me

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First, do you have an extention 701? If yes, then it should work.

The procedure, is that you find an Asteridex number you wish to call and click it, as you did. The PBX will ring which ever extension you had preselected in the field, upper left corner. Eg SIP/701. or SIP/3000, whatever extension you will be calling FROM. That extension should ring first (yours), and after you pick up, the destination number is dialed next to complete the transaction.

Try using another extension, and enter that SIP/xxx number in the field mentioned.
 

ericlee1

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I have EXT701, and its not working.. my EXT never rings.. This is the pop up I get:

Extension SIP/701 is ringing now.
When SIP/701 answers, call to 18002478726 will be placed.

phone is set to SIP, also tried PJSIP.. EXT will not ring.. just tried on 3 different installs I have going.

nothing pops up in the log files, so the command is totally being ignored I take it?
 

ericlee1

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ok found this in the error log

manager.c: 127.0.0.1 failed to authenticate as 'admin'

I haven't installed any 3rd party apps, so not sure why the something is out of sync

can someone confirm that its working on their install, I found a reference to amp111 as a password in one of the config files, but I don't have that setup anywhere.

???
 
Last edited:

ericlee1

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Still can't figure out how to get past this error code:

manager.c: == Manager 'admin' logged on from 127.0.0.1


any suggestions? tried a few fixes, but no luck..

:(
 

KUMARULLAL

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I tried everything. I also installed and compiles asterisk 13.22 and 13.23. It falls back to asterisk 13.7
I think for GVSIP to work, you need to be on asterisk 13.22 and above.
I had to resort to Ward's suggestion of creating a Hiformance GVSIP host and create trunks on the client PBX to connect to GVSIP host for inbound and outbound calling
 

JoeOIVOV

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Still can't figure out how to get past this error code:

manager.c: == Manager 'admin' logged on from 127.0.0.1


any suggestions? tried a few fixes, but no luck..

:(

Any luck? my experience is the same as yours. I setup a lot of these boxes and I have yet to see this work. Every now and then I try to see if it works but it only ever gives the error message: 'admin' logged on from 127.0.0.1
 

wardmundy

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@ericlee1 @JoeOIVOV Look in /etc/asterisk/manager.conf and write down the admin password. Then plug that into line 74 Secret field of /var/ww/html/asteridex4/callboth.php.

Or here's a 3-liner to do it for you:
Code:
oldpw=`cat /var/www/html/asteridex4/callboth.php | grep Secret | cut -f 2 -d ":" | cut -c2- | cut -f 1 -d '\'`
newpw=`cat /etc/asterisk/manager.conf | grep secret | cut -f 2 -d "=" | cut -c2-`
sed -i "s|$oldpw|$newpw|" /var/www/html/asteridex4/callboth.php

IMPORTANT: Make certain that one of your outbound routes includes a 1NXXNXXXXXX dial string entry, or calls will fail after you answer extension 701 to begin the call.
 
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JoeOIVOV

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the 3-liner worked great, thanks for that!

Now, its dialing for the first time, on the caller ID shows Anonymous, but it requires me to dial SIP/ infront of the extension and click save before it will call. It doesn't simply accept an extension. Is there is any quick commands to repair this or is it intended?

I ended up discovering from previous troubleshooting that this could be a issue but for the first time it does ring and connect. Thank you for this!
 

JoeOIVOV

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I found this by restamp:

To make the selection permanent, go into the /var/www/html/asteridex4 directory and change line 25 of index.php to "PJSIP/701" (or your preferred extension). Then change line 11 of config.inc.php similarly.

When I change to SIP/ it allows me to type in the extension after the / and click save and it works just fine, only thing is seeing SIP/ would be nicer for a radio button for SIP/ or PJSIP/ prefix. or an internal switch so we wouldn't have to see it but, it works so I'm just fine. I will use this a lot as I can add webex phone numbers with their call in pins. Works great.
 
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Joe

Thanks I never had the time to dig into this one. This did the trick. For anyone who might follow. When you change the extension number from 701 to your extension it does save. It just always shows extension 701.
 

JoeOIVOV

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I found this by restamp:

To make the selection permanent, go into the /var/www/html/asteridex4 directory and change line 25 of index.php to "PJSIP/701" (or your preferred extension). Then change line 11 of config.inc.php similarly.

When I change to SIP/ it allows me to type in the extension after the / and click save and it works just fine, only thing is seeing SIP/ would be nicer for a radio button for SIP/ or PJSIP/ prefix. or an internal switch so we wouldn't have to see it but, it works so I'm just fine. I will use this a lot as I can add webex phone numbers with their call in pins. Works great.

So I found this: makes these settings appear/work even better:

nano /var/www/html/asteridex4/callboth.php

and update this:
fputs ($fp, "Channel: $IN\r\n"); to this fputs ($fp, "Channel: SIP/$IN\r\n");
This allows you to use the extension field without remembering to type SIP/ or seeing this in the box.

When I go back into AsteriDex my previous extension is even still saved there, and I can change it quickly and click call and it works perfectly!
Now if only I can change the caller ID from [email protected] to something like [email protected] that way my recipients arn't so confused about the call...
 

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