Newbie question abou privacy manager


#1

Hi all,

Is there a way in AAH 2.5 to enable privacy manager on ALL incoming calls? I know you can do it at the extension level in AMP but I would like it to run on all incoming calls before they hit the IVR. Any ideas? I don’t mind editing the conf files :smile: .

Thanks,

Tom


#2

Ick. Well, good news, bad news: in AAH, you can do this in the inbound routing section (incoming DIDs), there is a checkbox for this. The bad news: only VOIP DIDs go through there, so if you have a PSTN line, you’re out of luck. Also, it only invokes privacy manager, and i’d like it to invoke zapateller too (nuke the telemarketers…) Also, some of the VOIP providers send weird strings like ‘asterisk’ or ‘UNKNOWN’ when the caller has blocked CID, so I ginned up some changes to extensions_custom.conf that forces all inbound calls through. I don’t have access to my AAH config right now, but if you want, I can post what I did later…


#3

Thanks for the reply! Fortunately I have my pstn forwarding to my VoIP number so the first item is not a problem. I found the option for privacy manager in inbound routing. What I’m trying to figure out is if you can enable it sooner. I want privacy manager to be active the moment * answers the call. I can’t seem to find a way to do that. Also my provider is TelIAX, as far as I can tell they don’t modify incoming CID. I’d appreciate anything you would be willing to post :smile: I can use all the help I can get!

Thanks,

Tom


#4

Not sure what you’re needing here. The inbound routing code runs very early in the process (asterisk picks up and goes to the inbound DID stuff). Why is that not good enough?


#5

Hmmm…

The way it seems to be working now is that the unknown call comes in, * answers it and the caller get the IVR. After the caller makes a chioce in the IVR and tries to transfer to an extension is when the privacy manager activates. I need to double check my configuration when I get home from work, I think I’m missing something simple. Hmmm…

Tom


#6

you’ve got something buggered up then. If you look at extensions_additional.conf (i think), you can see where it adds the contexts and rules for the inbound DID. privacy manager should be getting invoked there if you checked the box, certainly WAY before the IVR runs!

since telasip runs asterisk, they probably have the same bug i ran into (where they send funky strings, so PM doesn’t do the right thing…)


#7

Yep, I surely do have something wrong. I found the PM entries in extensions_additional.conf in the [ext-did] group, here’s a snippet (I replaced the phone numbers with n’s):

[ext-did]
include => ext-did-custom
exten => nnnnnnnnnn,1,SetVar(FROM_DID=nnnnnnnnnn)
exten => nnnnnnnnnn,2,PrivacyManager
exten => nnnnnnnnnn,3,Goto(aa_1,s,1)

As you can see it is the very first item after the context and include. I think the problem may be the context I have on the trunk, it’s from-pstn. I think because of that it is not even seeing this and using the incoming calls from pstn settings. I haven’t been able to figure out how to add PM to that context or what other context to try in the trunk settings. Any ideas?

Tom


#8

dswartz,

What were those changes for the extensions_custom.conf you mentioned? I found the following in my log file (PN’s replaced with n’s):

Mar 8 22:53:21 DEBUG[2496] chan_sip.c: Checking SIP call limits for device myloginid
Mar 8 22:53:21 DEBUG[2496] chan_sip.c: build_route: Contact hop:
Mar 8 22:53:21 VERBOSE[7558] logger.c: – Executing SetVar(“SIP/myloginid-287e”, “FROM_DID=nnnnnnnnnn”) in new stack
Mar 8 22:53:21 VERBOSE[7558] logger.c: – Executing PrivacyManager(“SIP/myloginid-287e”, “”) in new stack
Mar 8 22:53:21 VERBOSE[7558] logger.c: – CallerID Present: Skipping
Mar 8 22:53:21 VERBOSE[7558] logger.c: – Executing Goto(“SIP/myloginid-287e”, “aa_1|s|1”) in new stack
Mar 8 22:53:21 VERBOSE[7558] logger.c: – Goto (aa_1,s,1)
Mar 8 22:53:21 WARNING[7558] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_EQ, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:
= ANSWER
^
Mar 8 22:53:21 WARNING[7558] ast_expr2.fl: If you have questions, please refer to doc/README.variables in the asterisk source.
Mar 8 22:53:21 DEBUG[7558] pbx.c: Expression result is ‘0’
Mar 8 22:53:21 VERBOSE[7558] logger.c: – Executing GotoIf(“SIP/myloginid-287e”, “0?4”) in new stack
Mar 8 22:53:21 DEBUG[7558] pbx.c: Not taking any branch
Mar 8 22:53:21 VERBOSE[7558] logger.c: – Executing Answer(“SIP/myloginid-287e”, “”) in new stack
Mar 8 22:53:21 VERBOSE[7558] logger.c: – Executing Wait(“SIP/myloginid-287e”, “1”) in new stack
Mar 8 22:53:22 VERBOSE[7558] logger.c: – Executing SetVar(“SIP/myloginid-287e”, “LOOPED=1”) in new stack
Mar 8 22:53:22 DEBUG[7558] pbx.c: Expression result is ‘0’
Mar 8 22:53:22 VERBOSE[7558] logger.c: – Executing GotoIf(“SIP/myloginid-287e”, “0?hang|1”) in new stack
Mar 8 22:53:22 DEBUG[7558] pbx.c: Not taking any branch
Mar 8 22:53:22 VERBOSE[7558] logger.c: – Executing SetVar(“SIP/myloginid-287e”, “DIR-CONTEXT=general”) in new stack
Mar 8 22:53:22 VERBOSE[7558] logger.c: – Executing DigitTimeout(“SIP/myloginid-287e”, “3”) in new stack
Mar 8 22:53:22 VERBOSE[7558] logger.c: – Set Digit Timeout to 3
Mar 8 22:53:22 VERBOSE[7558] logger.c: – Executing ResponseTimeout(“SIP/myloginid-287e”, “7”) in new stack
Mar 8 22:53:22 VERBOSE[7558] logger.c: – Set Response Timeout to 7
Mar 8 22:53:22 VERBOSE[7558] logger.c: – Executing BackGround(“SIP/myloginid-287e”, “custom/aa_1”) in new stack
Mar 8 22:53:22 DEBUG[7558] channel.c: Scheduling timer at 160 sample intervals
Mar 8 22:53:22 VERBOSE[7558] logger.c: – Playing ‘custom/aa_1’ (language ‘en’)
Mar 8 22:53:24 DEBUG[2496] chan_sip.c: Auto destroying call ‘002da92109e4f5f74fc5995b7b2d956c@127.0.0.1’
Mar 8 22:53:25 DEBUG[2496] chan_sip.c: Auto destroying call ‘7124c1191419faeb36870b412bfdf518@127.0.0.1’
Mar 8 22:53:26 DEBUG[7558] channel.c: Scheduling timer at 0 sample intervals
Mar 8 22:53:26 VERBOSE[7558] logger.c: == Spawn extension (aa_1, s, 9) exited non-zero on ‘SIP/myloginid-287e’
Mar 8 22:53:26 VERBOSE[7558] logger.c: – Executing Hangup(“SIP/myloginid-287e”, “”) in new stack
Mar 8 22:53:26 VERBOSE[7558] logger.c: == Spawn extension (aa_1, h, 1) exited non-zero on ‘SIP/myloginid-287e’
Mar 8 22:53:26 DEBUG[7558] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
Mar 8 22:53:26 DEBUG[7558] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES (‘2006-03-08 22:53:21’,’“Unknown” ‘,‘Unknown’,‘s’,‘aa_1’, ‘SIP/myloginid-287e’,’’,‘Hangup’,’’,5,5,‘ANSWERED’,3,’’)
Mar 8 22:53:26 DEBUG[7558] chan_sip.c: update_call_counter(myloginid) - decrement call limit counter

So if I’m understanding this PM is looking at the call and thinks that Unknown is a good CID. That would seem to be pretty much the same as the problem you described. Sorry to be so dense! What do you think, same problem?

Tom