similar to: Prompting for number when CID number not sent?

Hello Instead of using PrivacyManager, I'd rather use my own dialplan to prompt the user for a ten-digit number if they called while blocking CID. This code does prompt the user, but 1) hangs up if the user didn't type the ten digits before the timeout 2) if the user did type the right number of digits, it still hangs up instead of Returning and then jumping forth to the "cid"

Hello About Record(), ATFT 2nd Edition says that "if the filename contains %d, these characters will be replaced with a number incremented by one each time the file is recorded." Problem is, the documentation doesn't explain how to refer to this filename later in the dialplan :-/ In this particular example, I want to move the file to the web server's /htdocs so users can

Hi folks, My v1.6 Asterisk system logs all Call Detail Records to a PostgreSQL database, including those handled by the Privacy Manager. Unfortunately, even though I can use the CLI to see the information being submitted by anonymous callers to satisfy the demands of the the Privacy Manager, that information is not recorded in the database. Instead, all that is written to it: clid:

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 all, Is it possible to display or print variables in Asterisk (e.g. in the CLI) for debugging purposes? For example, I'm using two different types of SIP phones: the Snom M3 and the Siemens S675IP. However, when anonymous callers submit a number to the PrivacyManager, only the Siemens displays the new CID correctly; the Snom shows "unknown" (even though the new CID looks

Hello I've been googling for a couple of days now, but still can't figure out what to put in zapata.conf to get it to report CID. Unless I'm mistaken, France uses ETSI FSK for CID method and bell 202 as CID FSK Standard: http://img219.imageshack.us/img219/7207/linksys3102cid1jj7.jpg http://img219.imageshack.us/img219/4625/linksys3102cid2ld5.jpg Does Zaptel support those on Digium

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

Is it possible to reject all incoming calls that do not have a CID? Could I do something like that (modified version from the book): exten => 123,1,GotoIf($[${CALLERIDNUM} = ]?20:10) exten => 123,10,Dial(Zap/4) exten => 123,20,Playback(abandon-all-hope) exten => 123,21,Hangup( Alternatively, what's the privacy.conf file for? What does it mean for a user to have to chances to

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 having a problem with a polycom 301 not picking up a ZAP call. Below is the CLI output of the call. I have: TDM400 with 2 FXO lines Asterisk 1.2.14 Polycom 301 When I dial the first ZAP line, I choose an extension that rings the polycom, polycom rings and I can pick it up and the call is bridged. When I call my second zap line, the polycom rings, but I cannot pickup the

Hi all, using Skype for Asterisk I have the following problem. In my dialplan I need to have a CID matching (example.skype) over a DID (test.skype) : [example] exten => test.skype/example.skype,1, NoOp(nothing) exten => test.skype/example.skype,n, Hangup() Where test.skype and example.skype are Skype business account. In this case, when I get a : CLI> show dialplan example I get: [

Hi list, After successfully configuring Linksys SPA3000 and SPA3102 devices as Asterisk PSTN gateways, the only thing I can't get working is the PSTN Caller ID. The analog and SIP phones I've used can both display CIDs for internal calls, while the analog model also displays CIDs correctly when attached directly to the PSTN line. However, when PSTN calls come in via the SPA

Hi Guys, I have several * servers connected to T1 PRI's from various service providers in multiple locations the US. All the * servers use the same hardware with the same OS and * version. When connected to 5ESS Switches, using the NI2 (national) PRI protocol, the CID name and number come across fine and populate into the * CDR fine. I connected to a DMS-500, NI2 (national) protocol and

I'm thinking of doing an app to work with the CID that's gotten from the Zap channel. All the CID's I see from within the US are 10 digit numbers. I'm out in the rural areas of the US, and no-one ever calls me from overseas. If they did, what would the CID look like? What does the CallerID look like overseas? How many countries provide it? murf

Hi List, I need to modify the callerID name of the call coming back when a parked call returns to the extension that parked it when it times out. Looking at app_parkandannounce.c /* Now place the call to the extention */ snprintf(buf, sizeof(buf), "%d", lot); memset(&oh, 0, sizeof(oh)); oh.parent_channel = chan; oh.vars =

Hello I use the Linksys 3102 to connect Asterisk to a POTS line, and XLite on XP as an SIP client: http://img694.imageshack.us/img694/1421/3102asteriskxlitecid.png The problem is that by default, Asterisk doesn't rewrite the CID name + number in incoming calls, so that XLite displays whatever name I used in the 3102 and the extension the 3102 uses to register with Asterisk. How can I tell

Hi, I'm trying to configure a Linksys/Cisco SPA8000 talking SIP to both a local Asterisk server and also with a trunk directly to a VOIP provider. Everything works great, except I'm having a problem setting the outbound caller ID to a value different from the SIP username/authname. The SPA8000 has SIP setting for Display Name, User ID, Password, and Auth ID, as well as a "Use Auth

With new enough libnbd, we already support vsock by virtue of uri=; however, it's also nice to allow direct exposure of the nbd_connect_vsock() api. Signed-off-by: Eric Blake <eblake@redhat.com> --- plugins/nbd/nbdkit-nbd-plugin.pod | 29 +++++++++++--- plugins/nbd/nbd.c | 63 ++++++++++++++++++++++++------- 2 files changed, 73 insertions(+), 19 deletions(-) diff --git

??? Usually phone companies set the outgoing CallerID for you but recently we got control over that and are now setting the outgoing Calleir ID ourselves.? My problem now is that the CDR will put the outgoing CID in the CDR instead of the extension that dialed and that causes problems for reports.? What is the proper way to set outgoing CID and keeping the original extension number in the

Hi there. Is it possible to have different sip users have the same CallerId number in sip.conf. I need this because we got multiple companies on this Asterisk box. Company A's internal numbers: CID: User: 1000 - User 1 2000 - User 2 3000 - User 3 4000 - User 4 Company B's internal numbers: CID: User: 1000 - User 5 2000 - User 6 3000 - User 7 4000 - User 8 Is this allowed? Regards