search for: zapatel

I am having issues with Privacy Manager and Zapateller. If I set callerid="" on a sip user zapateller sends the tones If I set callerid="Anonymous" <8475551212> zapateller doesn't send the tones If I call from a phone after dialing *67 zapateller doesn't send the tones In the last 2 cases, the display on the phone...

Does anybody know why I get this error when using zapateller: WARNING[1209214400]: File rtp.c, Line 327 (ast_rtcp_read): RTP Read error: Resource temporarily unavailable It's scrolls until a sound is recived from the line, then it plays the zapateller tones. /Mike

Hi All I'm tracing a very strange problem which I could reproduce with different versions up to 1.2.5 (sorry, didn't update to a newer one yet). Scenario 1: Problem does not occure. ============================= Sip Phone registered directly to the Asterisk. exten => i,1,Zapateller() exten => i,n,Playback(invalid,noanswer) exten => i,n,Hangup() Works like expected. I dial an invalid number, hear the SIT tone and then the Announcement. Scenario 2: Problem does occure. =========================== SIP 'trunk' to another SIP PBX (or another Asterisk). exten...

Been tinkering and found Zapateller appears to be answering when I didn't expect it to. I have something like so: [incoming] exten => s,1,NoOp ;Zapateller(answer|nocallerid) exten => s,2,PrivacyManager exten => s,3,Dial(${RING},20) ... I have a 1x1 analog * installation with a couple IP phones too. I've got th...

Hi All, In theory if I do this; exten => s,1,Zapateller(nocallerid) exten => s,2,Privacymanager exten => s,3,Dial(a bunch of SIP extensions) My callers should only hear the anti-telemarketing tones if they call from a line that has no caller*ID and then get offered an opportunity to enter it, right? What I'm finding is that in the event o...

Hi, I have been using Zapateller with a TDM400 no problems at all, but recently I have ported our BT number to a VoIP provider, and have a strange problem. When I phone our number I first get the BT unavailable three tone sound, and then it actually connects the call via IAX2. So, I disabled zapateller in the dialplan and tri...

Hey all...I'm just getting my * setup and right now all I have is an FXO but no FXS. I wan't to get rid of telemarketers by having * pick up the phone if there is no CID present, give the caller the Zapateller tones and then ask the user to input their phone number via Privacy Manager (yes I realize that this won't get us any where given that I can't re-ring the phones without FXS but that'll be my next purchase) This is what I have in my extensions.conf: [incoming] exten => s/,1,Zap...

My PSTN line doesn't allways hang up properly after it goes to voicemail. The problem occurs when a caller hangs up during the initial greeting. Even though the hangup accured, voicemail continues to record, usually a fast busy and/or a teleco generated "please hangup now" message. After the voicemail.conf 'maxmessage=180' expires the line simply stays offhook. The hardware

...ling) 2. RE: G.723 (Steven Critchfield) 3. Re: Voicemail storage in DB (James H. Cloos Jr.) 4. Voicemail config from database (AJ Grinnell) 5. oob to inband dtmf over rtp (James H. Cloos Jr.) 6. OT appologies to list (Linus Surguy) 7. Re: OT appologies to list (Brian Cuthie) 8. Zapateller issues (Mark Phillips) 9. RE: Zapateller issues (Sean Cheesman) 10. RE: Zapateller issues (Darrin Johnson) 11. Re: Re: Voicemail storage in DB (James H. Thompson) 12. Re[2]: [Asterisk-Users] OT appologies to list (Stephen Karrington) --__--__-- Message: 1 Date: Mon, 12 Apr 2004 13:08:...

...e my name from the list as a control measure. I'll let someone else do that dirty work. The NO-CALL list, tho, is not completely affective, though. Charities, political parties, government recruiters, those with remote "pre-existing business relationships" are still clear to call. 2. ZAPATELLER. The Asterisk Zapateller application, which plays the SIT (Special Information Tones) (the dah-dee-die tone, usually followed with the female voice, "The number you have dialed...") to those with no caller id, is the first app run on incoming calls on both my business and home lines. H...

Relax your PRI is fine. What Xo is sending you is 4 digits os DID. If for example you have 1130-1153 as the last 4 digits of your Number you can use this to rout your calls. exten => 1130,1,Goto(ivr,s,1) Exten => 1140,1,Goto(extensionss,100,1) Exten => _X.,1,Zapateller Is the above config. XXX-XXX-1130 with go to the s extensions' first priority in the ivr context. XXX-XXX-1140 will go to extension 100 in the extensions context XXX-XXX-???? Will get a Zapateller tone. Any questions? > -----Original Message----- > From: asterisk-users-bounces@l...

..., but my ternary operator/plan currently is not properly being implemented. Can anyone spot where I'm being a momo? All extensions get forwarded to the following macro: [macro-forward] ; arg1 = phone number ; arg2 = timeout ; arg3 = extension (voicemail) ; arg4 = mobile number exten => s,1,Zapateller(answer|nocallerid) exten => s,2,PrivacyManager exten => s,3,Wait(1) exten => s,4,GotoIf($[${CALLERID(number)} = "15552221313"]?15:5) exten => s,5,GotoIf($[${LEN(${CALLERID(number)})} = 4]?6:8) exten => s,6,AGI(didextlookup.agi|${CALLERID(number)}) exten => s,7,Set(CAL...

...ist and the relevant device structures are present. lsmod | grep ztdummy gives:- ztdummy 5160 0 zaptel 186916 1 ztdummy rtc 12372 1 ztdummy Where I'm stuck is I am now at a loss as to how to configure my /etc/zaptel.conf and /etc/asterisk/zapatel.conf files:- Zaptel.conf contents are:- # Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order #Span 1: ZTDUMMY/1 "ZTDUMMY/1 (source: RTC) 1" (MAST...

...The db shows: asterisk*CLI> database show blacklist /blacklist/<1010987/18887975686number> : 1 /blacklist/<name/number> : 1 /blacklist/unlisted/6145551212 : 1 asterisk*CLI> exten => 2129,1,Wait(1) exten => 2129,2,Zapateller(answer|nocallerid) exten => 2129,3,NoOp exten => 2129,4,PrivacyManager exten => 2129,5,LookupBlacklist exten => 2129,6,Dial(Zap/4,5,Ttr) exten => 2129,7,Answer exten => 2129,8,Wait(1) exten => 2129,9,Playback(personal/hello) exten => 2129,10,Playback(personal/i-am-not-in-...

does anyone have a recorder GSM file that emulates the Telco's "if you are a telemarketer please hangup now" recording? I don't see one in the sounds dir. the ZapATEller works great for computerized callers but if a human hears this message asking them to go away they have to. Isn't that right? Dave

...nks that CallerID is present (when it isn't). I get this on the console: == Primary D-Channel on span 1 up -- Accepting voice call from '' to '0711234567' on channel 0/2, span 1 -- Executing Ringing("Zap/2-1", "") in new stack -- Executing Zapateller("Zap/2-1", "answer|nocallerid") in new stack -- Executing PrivacyManager("Zap/2-1", "") in new stack -- CallerID Present: Skipping -- Executing SetCIDNum("Zap/2-1", "") in new stack -- Executing LookupCIDName("Z...

...erID: ${CALLERID}) exten => s,n,Hangup exten => fax,1,Set(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif) exten => fax,n,rxfax(${FAXFILE}) exten => fax,n,Hangup exten => i,1,Hangup exten => h,1,Hangup [incoming] include => parkedcalls exten => s,1,Answer exten => s,n(zapateller),Zapateller(nocallerid) exten => s,n,LookupCIDName exten => s,n,NoOp(CallerID: ${CALLERID}) exten => s,n(ring),Dial(SIP/2000&SIP/2001&SIP/2002,30) exten => s,n,Voicemail(u2999) exten => s,n,Hangup exten => s,n(ring)+101,Voicemail(b2999) exten => s,n,Hangup exten =&g...

...censed G729 Codec from Digium. This is the error message from asterisk -vvg: [app_playback.so] => (Sound File Playback Application) == Registered application 'Playback' [app_dumpchan.so] => (Dump Info About The Calling Channel) == Registered application 'DumpChan' [app_zapateller.so] => (Block Telemarketers with Special Information Tone) == Registered application 'Zapateller' [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator) == Registered translator 'ilbctolin' from format ilbc to slin, cost 7 == Registered translator 'lint...

...OSE[4196] logger.c: -- B-channel 0/1 successfully restarted on span 1 lspci -v 06:03.0 Communication controller: Digium, Inc. Wildcard TE405P (2nd Gen) (rev 02) Flags: bus master, medium devsel, latency 32, IRQ 201 Memory at 30004a00 (32-bit, non-prefetchable) [size=128] /proc/zapatel/1-4 with 5 being ZTDUMMY Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" B8ZS/ESF ClockSource Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" B8ZS/ESF RED Span 3: TE4/0/3 "T4XXP (PCI) Card 0 Span 3" Span 4: TE4/0/4 "T4XXP (PCI) Card 0 Span 4"

Hey all...I'm brand new to * and I want to convert my home into a pbx type setup. I've figured out that I want a Wildcard X100P to bring my single POTS CO into my Linux box. My problem is that I'm sure sure what I need to do to get my analog phones connected up into a structured phone system. It *looks* like I can go the route of the Cisco Analog -> VOIP for about $100 on ebay.