I HAVE A DREAM NEW CallerID Superfecta 2.2.4: THE MODULE

[tshif]

I have a small issue with Asterisk Phonebook lookups.

I have multiple DID providers. Two of them pass the CID number as a 10 digit number. The other passes it as 11 digits with a one at the front. 99% of my calls come from the provider that uses 11 digits.

The last script I used I made a small mod to make sure my numbers were all being stored in 11 digits no matter which provider it was. Mostly because of Caching. It was a simple mod to the php.

Since this is a module I don't want to go messing with the code and have it break on an update. Since this module does not cache it would be nice if there was some option to select an 11 digit number for US users.

We hope to begin grappling with dealing with other number formats in the not-to-distant future. Although we currently phrase it as support for UK (and other) numbers, I have added your situation to the discussion - as it seems like this is the likely time and place for it to get dealt with properly. Right now, I don't have a good suggestion for you. Anyone else? (http://projects.colsolgrp.net/issues/show/55)

 

[kelinger]

So close but...

I feel like an idiot because I've been messing with this since 1.x. I have it installed and it passes the debug tests just fine. However, I don't actually GET any text on my phone upon a successful look up (using 2.0 with the latest Trixbox CE). I have my maint username/password entered and I have Superfecta selected as the CID source for any inbound routes. Am I missing something obvious?

 

[tshif]

I feel like an idiot because I've been messing with this since 1.x. I have it installed and it passes the debug tests just fine. However, I don't actually GET any text on my phone upon a successful look up (using 2.0 with the latest Trixbox CE). I have my maint username/password entered and I have Superfecta selected as the CID source for any inbound routes. Am I missing something obvious?

Just to make sure I understand - you GET info in the debug window, but it does not reach your telephones?

Thats a new one on me. Hmmmm. Ok - What kind of telephones are you using? What firmware revision?

 

[vcallaway]

I had a weird issue that might be your problem.

When I filled in the maint username and password it did not store it properly. While watching ( asterisk -r at the command prompt) it would show an authentication error.

The solution was to go into the callerid lookup sources screen and manually put them in. Don't know why the error was there because I checked the mysql database entry and it all looked correct. Just a glitch I suppose.

 

[kelinger]

Just to make sure I understand - you GET info in the debug window, but it does not reach your telephones?

Thats a new one on me. Hmmmm. Ok - What kind of telephones are you using? What firmware revision?

Exactly. Debug shows successful lookups from several of the sources (entering my 10 digit number I'll be calling from) and also shows what it "would" display if it weren't a debug. Yet, actually calling in from the same number I ran in debug displays the correct number on the SIP phone (same 10 digit number) but no text whatsoever.

Currently testing with Grandstream GXV 3000 and Aastra 9480i SIP phones.

 

[hernanp]

phonespamfilter not returning results

I've installed 2.0.1, and for some reason, the phonespamfilter is not working properly.

I'm entering a number that shows 14 complains thru the web, but when debugging, it shows as Not Found...

number is 2044809170...

 

[hernanp]

I've installed 2.0.1, and for some reason, the phonespamfilter is not working properly.

I'm entering a number that shows 14 complains thru the web, but when debugging, it shows as Not Found...

number is 2044809170...

Here are samples of the problem:

Debugging Enabled, will not stop after first result.
Found valid US npa: 204-480-9170

Searching PhoneSpamFilter ... not found
result took 0.4008 seconds.


This is the link to www.phonespamfilter.com showing 14 complaints...

http://www.phonespamfilter.com/search.php?phone=2044809170+

 

[tshif]

Ok your right - your install did not perform correctly for that look up. Here's my results. (At least we know the script works - this is the first number Ive ever tested that was in the system!)

Debugging Enabled, will not stop after first result.
Found valid US npa: 204-480-9170

Searching PhoneSpamFilter ... value of 42 returned ... determined to be SPAM
result

took 0.9562 seconds.

Returned Result would be: SPAM:
result

took 0.9566 seconds.

Please test this source a few times throughout the day. This source should always return a determination - either negative or positive (SPAM, or NOT SPAM) - if it reaches the service correctly.

 

[kelinger]

Exactly. Debug shows successful lookups from several of the sources (entering my 10 digit number I'll be calling from) and also shows what it "would" display if it weren't a debug. Yet, actually calling in from the same number I ran in debug displays the correct number on the SIP phone (same 10 digit number) but no text whatsoever.

Currently testing with Grandstream GXV 3000 and Aastra 9480i SIP phones.

I think I may see the problem. If I go into "asterisk -r" and call my extension, it doesn't even call the CallerID lookup function. If I call a specified extension (like the conference bridge which has a specific inbound route) then it does the lookup successfully.

The problem may be that most extensions would fall into the "Any CID/Any DID" category which, even though its defined as an inbound route, doesn't appear to actually be used.

Thus, the issue may just be that I need some way to get the Caller ID function called when someone is just calling a direct extension from outside.

 

[tshif]

Exactly. Debug shows successful lookups from several of the sources (entering my 10 digit number I'll be calling from) and also shows what it "would" display if it weren't a debug. Yet, actually calling in from the same number I ran in debug displays the correct number on the SIP phone (same 10 digit number) but no text whatsoever.

Currently testing with Grandstream GXV 3000 and Aastra 9480i SIP phones.

Ill test this with a Grandstream GVX 3000 I have here as soon as I can. Remember - you have to select the Caller ID Superfecta as the Caller DI Lookup Source for each inbound route you want it to function with.

 

[tshif]

I had a weird issue that might be your problem.

When I filled in the maint username and password it did not store it properly. While watching ( asterisk -r at the command prompt) it would show an authentication error.

The solution was to go into the callerid lookup sources screen and manually put them in. Don't know why the error was there because I checked the mysql database entry and it all looked correct. Just a glitch I suppose.

After you put the ID and password in the module UI, did you click the "Agree and Save" button? You must, or the General Options are not saved.

 

[kelinger]

Ill test this with a Grandstream GVX 3000 I have here as soon as I can. Remember - you have to select the Caller ID Superfecta as the Caller DI Lookup Source for each inbound route you want it to function with.

Every single inbound route has Caller ID Superfecta selected. I'm thinking that its not reading the "any DID/any CID" route, though and just going directly to the extension.

 

[tshif]

Every single inbound route has Caller ID Superfecta selected. I'm thinking that its not reading the "any DID/any CID" route, though and just going directly to the extension.

Interesting. Ok, you mentioned something about calling from an extension. Extension to extension calls don't use inbound routes. You would have to call from an outside line. You could dial out of your pbx, into your public number, and that should work if you are sending caller ID on your outbound leg.

Does this make sense?

 

[kelinger]

Interesting. Ok, you mentioned something about calling from an extension. Extension to extension calls don't use inbound routes. You would have to call from an outside line. You could dial out of your pbx, into your public number, and that should work if you are sending caller ID on your outbound leg.

Does this make sense?

Sorry... no, I have not been trying internal calls. I've always been trying from an outside line calling IN to my extension. I may have messed up the wording before but that's all I've ever been doing (out to in).

 

[tshif]

Sorry... no, I have not been trying internal calls. I've always been trying from an outside line calling IN to my extension. I may have messed up the wording before but that's all I've ever been doing (out to in).

Ok - please capture the asterisk cli trace that results for one of your inbound calls - and let us see it. Its important to get it from the start in the call being received. It will be clearer if you dont call out of the same PBX your calling into. FOr example, for the test, call in from your cell phone.

 

[kelinger]

Ok - please capture the asterisk cli trace that results for one of your inbound calls - and let us see it. Its important to get it from the start in the call being received. It will be clearer if you dont call out of the same PBX your calling into. FOr example, for the test, call in from your cell phone.

Two files attached. "not-working.txt" is when I call from my cell in to my extension. You'll notice that its not even checking for the caller ID text. The second file is "working.txt" which is what happens when a call into a specifically defined inbound route like our conference bridge (no SIP phone; just an inbound route that directs the call to the conference IVR). This would seem to indicate that the inbound route for default (matches the inbound DID to the extension versus going through the explicit inbound routes) is where something is missing.

BTW, thanks for all the help.

 

[FitnessPlus]

Here are samples of the problem:

Debugging Enabled, will not stop after first result.
Found valid US npa: 204-480-9170

Searching PhoneSpamFilter ... not found
result took 0.4008 seconds.


This is the link to www.phonespamfilter.com showing 14 complaints...

http://www.phonespamfilter.com/search.php?phone=2044809170+

hernanp: If you direct your browser to the results page and hit refresh 10-20 times in short order you may duplicate what I was able to...and that is a 500 error on the page load. An error loading the page is the only thing that should cause the "not found" problem with this source.

I'm not sure if this is a protection that they have put in place to stop abuse, of if their server just gets overloaded easily...so I'm going to request that everyone else doesn't go overload their server with a bunch of requests.

 

[steve54301]

This is the greatest. Thank you.

Here is a thought. How about a way to use this externally. This could be handy for when you receive an unrecognized number on a cellphone and curiosity has the best of you.

Dial an extension; prompted for a 10 digit number; TTS reports back based on the lookup sources pre-setup...

 

[tshif]

This is the greatest. Thank you.

Here is a thought. How about a way to use this externally. This could be handy for when you receive an unrecognized number on a cellphone and curiosity has the best of you.

Dial an extension; prompted for a 10 digit number; TTS reports back based on the lookup sources pre-setup...

Your Welcome!

Interesting suggestion. Remote TTS Caler ID. There are always possibilities. I just wonder how much use this would get. Technically, you can do that through the debug function right now, albeit not over the phone. Everyone, chime in and tell us if you want this feature considered for addition.

 

[blakekrone]

That is an interesting suggestion, I would probably use it as we seem to get some calls on the cells that we don't answer just because we don't know who it was and then when they don't leave a message you have no idea.