similar to: Zapateller Answering?

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

While setting my first couple IP phones, I set their voicemail buttons to an extension that runs VoicemailMain. exten => 8500,1,Wait(1) ; voicemail exten => 8500,2,VoicemailMain ; exten => 8500,3,Hangup ; I would like to be able to pass the mailbox number allowing each phone to go in directly but I'd rather tno have

One of the suggestions I got for dealing with my analog FXO woes was to look into using a Voicetronix card instead of the Digium. They're more expensive but if it worked better it may be reasonable. They indicated they were compatable with the CVS HEAD version of *. Can someone share with me their experiences with Voicetronix OpenLine and/or OpenSwitch cards with *? I'm specifically

New to asterisk so please be gentle. I'm guessing I'm among a number of recent additions to the list after the article in Linux Mag. I gotta say I'm *very* intersted in the project and will be doing lots of reading shortly. A couble quick questions first... How suitable is Asterisk for use as an IVR providing callers with textual data out of a database? Can it be combined easily

In case it helps others, a couple gotchas that I had to dig and beat my head against to get past... - building on FC2, use "make linux26" when building asterisk - also for FC2, a couple links in /usr/src named linux and linux-2.6 and pointing to /lib/modules/2.6.8-1.521/build/ were needed to build - yes, the power connector on the TDM400P needs to be connected to the power

Looking to outfit a 8-person office shortly and am looking for IP phone recomendations. Saw recent threads and the wiki with all kinds of suggestions and notes on configs but little on the "feel" and sound quality. I have a PolyCom SoundPointPro analog desk/speaker phone and the Cisco 7940 here. How to the other business-class phones compare, from a user's point of view, to these?

In my search for a better analog FXO interface, I'm looking into the Voicetronix cards. Can someone let me know if they've been using the OpenSwitch cards with Asterisk? Also, can anybody comment on the differences in FXO performance between the OpenCall and OpenSwitch cards? Thanks in advance, Paul -- Paul A. Dugas Dugas Enterprises, LLC email:

Seems strange to me that I should be having this problem but I'm stumped. I'm still playing with a Voicetronix OpenSwitch6 card and Asterisk (development CVS). I using a dialplan that's derived from one on another running installation that's using Digium FXS/FXO interfaces instead of the OpenSwitch card. It seems that as soon as a user on an FXS interface dials 7 digits, a call

I've got a single IAXy installed in a little office nearby and got a call from someone on site a finew mintues ago. Apparently they couldn't make a call on that extension. They'd pick up the phone and get nothing; no dial-tone. Running "iax2 show peers" showed the unit still registered and I could ping it too. Didn't try to call it while it was doing this; sorry. The

Hi all, Happy Wednesday! Got a 7940G off eBay to tinker with and, per the wiki, contacted Cisco about a service contract. Being just little ol' me, they won't deal directly and blew me off (I hate dealing with Cisco, Nortel, etc as a small-business person...) Again per the wiki, I hit CDW.com and found CON-SNT-CP7960 for $9.14. Looking for support for the 7940 (not 7960), I just

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

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

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

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

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

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 =>

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've been struggling with a test * install for a couple months now in a small office and am just about ready to give up on it. It's not that the system itself is a problem. I've got everything (attendant, voicemail, FXS extensions, Cisco and Polycom hard-IP phones, and 2 VOIP carriers) working except for the frigging analog FXO interfaces. These things are driving me completely mad.

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

My Asterisk fax detection used to work, but no longer does. OK. So, here's the deal: 1. It appears that the "faxdetect" command cannot be applied channel-by-channel in zapata.conf anymore, as Asterisk appears to the last "faxdetect=" command to ALL channels. 2. My stations are detected and sent to the proper extension; i.e., when I send a fax from one zap extension