ENUMPlus

Im using 1.8 and when I goto /var/lib/asterisk/agi-bin/enumlookup.agi I do not see the | character in the file. I will poke around and see if I can figure it out.

Thanks
Robert
 
Well that is because it does not exist "in" the file - it actually is the data sent back from ENUMPlus.

All the file really does is get your api key, send the request - then if the result is not empty it populates your dial variable with whatever is returned from the curl request.

I could in fact run an update in the ENUMPlus database to replace all "|" with "," - unfortunately that would delimit all <=1.2 Asterisk users.

I will have an update to the module for strict dial plan conventions within the next 48 hours.

Thanks for spotting this - I was always comfy with the loose dialplan scripting, I have been meaning to update for a while now.
 
Dynamic DNS not working

I set up dynamic DNS at ENUMPLUS.org. It stopped working about a week ago. I checked my firewall with tcpdump and it looks like packets are passing through from ENUMPLUS to my PBIAF server. I tried running the curl command from the command line with no results. Is anyone else having problems with this DNS service?
 
Dynamic DNS for incoming

I'm talking about incoming calls using ENUMPLUS dynamic DNS.
 
Just started playing with SIP/URI and ENUM. I havn't found much in the way of installation documentation (if it doesn't exist I will happily jot something down). Can someone please scan thru the steps I followed and let me know if I missed something:
  1. ENUMplus Module install was very straight forward, got 0.2.3 set up with zero issues. Registered and got API from enumplus.org
  2. I manually created an ENUM trunk with appropriate dial rules no problem. There is nothing much to configure here, does the FreePBX ENUM trunk just "know" that it is using the enumplus module for lookups?
  3. My PBX is not visible to the 'net, so I have created a personal SIP/URI using voip.ms and redirected it to my voip.ms DID. I have confirmed that this is working via a softphone.
  4. Registered two of my DID's with my new enumplus.org account, verified the numbers via the callback feature and verified that the SIP/URI is working properly.
Questions:
  1. What happens with my DID registrations on enumplus.org? Do those entries get shared with other enum lookup directories (e164.arpa, e164.org, e164.info, enum.org, etc.)?
  2. I have successfully tested my new ENUM outbound route by dialing some ENUM test numbers listed on voip-info.org and I got the "ENUM lookup successful" blurb then the outbound connection. If I dial one of my own DID's registered with enumplus.org, I get the "ENUM lookup successful" but immediately get "All circuits are busy". Is it possible to initiate and recieve the same ENUM call on the same PBX?
*edit* (Using Asterisk 1.4 and FreePBX 2.6)
For testing I have created an outbound route that is dedicated to my own extension. All my dialed numbers go to this outbound route which has ENUM as the first trunk then other conventional trunks after. If I dial a number with a successful ENUM lookup, call is routed via the URI and all is well. Most calls I make, of course, do not have an ENUM lookup and I would assume that the call would just pass over the ENUM trunk and go to the next defined trunk in this outbound route. That is not the case however, it seems like if the ENUM lookup fails the call will go out using another trunk, possibly via another outbound route. Anybody know what's going on here?

Lorne

Answers :
1. No, sorry there is no API available (that's why ENUMPlus exists, it searches the top lookup sources)
2. Check the CLI when dialing yourself, it will probably say somewhere "482 Loop Detected" - Try calling through a VPN back to your PBX or from a friends house and I'll bet it works.
 
DynDNS

Aster1sk,
DNS records pointing correctly now. Thanks!
 
All files can be found here

I failed to update the manifest (xml) file that describes the version, will be updated with 0.2.5 (this week hopefully).

I was also just reminded about a little bug with newer versions of Asterisk and the (|) Pipe / (,) Comma dilemma. This will also be fixed in 0.2.5 - I'll update here when it's ready.
 
Is the asterisk 1.6 documentation appropriate for 1.8?

I realize this is pretty simple, but without adequate documentation alot of people aren't going to get it going. Looks like a fairly complex dialplan for an old version of asterisk.

How about some documentation for 1.8?:confused5:
 
Been a while.

My apologies for the late reply - I'm currently updating the enumplus module for newer versions of Asterisk as well as re-thinking the site. There are a few problems I've been experiencing (with user input, not the actual service) and think I could remedy the 1.8 upgrade with some simple settings in the user panel (ie: replace | with ,).

I'll post back as soon as I've figured out how I want to approach the upgrade.
 
Might as well do Asterisk 10 (PIAF-Red) while you're at it. The dialplan code seems to be roughly the same as Asterisk 1.8.
 
I kept getting a stray (|) character at the end of the enum sip string.
I added this line to fix it.

$result = shell_exec($cmd);
# Added to get rid of |
$result = substr($result, 0, -1);
if (!empty($result)) {
 

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