similar to: TE110P + PRI incoming + outgoing extensionsquestion

I just got a TE110P up on an XO PRI - everything looks good so far. We've been given a block of 23 numbers for the PRI. If I explictly set the incoming extension in extensions.conf like: exten => 1153,1,Answer or: exten => _XXXX,1,Answer I can get the incoming call. If I try and do: exten => s,1,Answer I'll see something like this: -- Extension '1153' in context

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 shows -Blocked Call- PrivacyManager always gives

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 the FXO interface connected to the home phone line. When I get an

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

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, 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 of no CID the caller gets dumped into the

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 tried again. Would you believe it worked fine. Has anybody else come

Hi Lists, I have one box with two FXO and two FXS ports, it is running asterisk inside. I have one sinle POTS line connected to the one FXO and two phone sets connected to the FXS port. Extension 6003 is asigned to one fxs and 6004 is asigned to another fxs, the two extensions can call each other, they can both make/receive PSTN call, but they can't make PSTN call simultaneously. Is it

Hello-- I've been playing with the privacy options on my home/home-office system since August last year, and have some results, gleaned from my CDR records, which over the last 13 months, number a total of 8672, which includes incoming, as well as outgoing calls. Before I start spitting out numbers, let me note that with the current setup, I haven't had to tell a single telemarketer

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

Hi all, I'm looking for some rudimentary insight on GotoIf() which seems to be failing on me in my dial plan. All I basically wish to do is block a particular caller. Sounds easy enough, 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

I am attempting to write in incoming context for calls. 1. If the caller id is given and it is not black listed it will Playback a greeting and then right the phone or go to voicemail under busy or unavailable conditions 2. If no caller id is given, then Privacy Manager will ask for the number. I am testing 6145551212 to see if the black list will work 3. If a caller id is given, and it is

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

Hello, I have a rather odd problem with Asterisk detecting faxes. I have two POTS lines coming into the box (TDM400P). Line 1 is for voice, Line 2 is fof fax. When I set them up with channel => 1-2 in zapata.conf, all is fine, but as soon as I have two channel => definitions, Asterisk is unable to detect faxes. The fax line is not supposed to ring local phones, so the most obvious

I upgraded my server from Fedora Core 4 to Fedora Core 5. I was wondering if anybody else has run into the problem and know's the fix? I recompiled asterisk and if I don't have the /usr/lib/asterisk/modules/codec_g729a.so file in place it works. I use or used to use the licensed G729 Codec from Digium. This is the error message from asterisk -vvg: [app_playback.so] => (Sound File

I've setup the following in extensions.con: exten => 2200,1,Ringing exten => 2200,2,Wait(2) exten => 2200,3,Answer exten => 2200,4,Zapateller exten => 2200,5,Macro(stdexten,2205,SIP/2205) This works as expected if I dial from a SIP phone on my desk. However, if I dial in from the PSTN (through a SIP provider) it fails while trying to play ths SIT with: Apr 8 18:53:12

Is it possible to give a caller three goes at an extension number then hangup? exten => s,1,Zapateller(answer|nocallerid) exten => s,2,PrivacyManager exten => s,3,Ringing(1) exten => s,4,NoOp(${CALLERID}) exten => s,5,SetMusicOnHold(random) exten => s,6,Background(silence/1) exten => s,7,Background(thank-you-for-calling) exten => s,8,Background(silence/1) exten =>

I'm trying to build a test suite so that I can run "calls" through and verify the call results. I've made a cross over cable and linked my 2 ISDN30 ports together. So now I can dial out on span 1 , and to receive the call on span 2. in the context for span 2, I have the following: <snip> ; #1 "answer" a call and play music 000XXX : ring for a random period,

Hi all, I have the following in my extensions.conf: exten => s,4,GotoIf($["${CALLERID(number)}" = "8585979857" | "8585970327"]?15:5) The numbers listed above are known spammer numbers. However, when I call from any other CALLERID, it still directs me to s,15 which is the Hangup() application. Here are logs from the asterisk CLI: -- Executing

I am looking to install a web interface for Asterisk to transfer calls and look who's on the phone. If anybody has a working web interface please let me know. I installed the www.asternic.com (operator) But when I bring up my web browser it says transferring data and does not bring a browser. -----Original Message----- From: asterisk-users-admin@lists.digium.com