similar to: Cisco Remote-Party-ID / Bug #2012

Hello! I've got a question regarding the Q.931 Setup-field Calling Party Number. It contains five things: Type of number, Number Plan, Presentation and Screening indicators and the actual number. Our provider uses some of those to decide if the numer should be presented or not to the outside world. I've done a crude hack in our GnuGK to always change those so that our numbers are

Hi list! I'm trying to get PrivacyManager working but for some reason it always thinks 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

An attachment called (WORM_SOBIG.C) in an email that appears to have been sent from your email address to (arrowcomp at xtra.co.nz) contained the virus (WORM_SOBIG.C), which has been deleted. If you do not believe you were the actual sender, the Klez virus is likely to be the culprit. The Klez virus works by forging the 'From' address inside the virus infected email, which means you can

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, I am running into some problems here with PrivacyManager. We used to use it without any issue, but now there seems to be several problems. We are currently running Asterisk 1.2.4. First, it seems that if the user does not press the pound (#) key after entering their number, PrivacyManager will fail. I have the minlength set to 10, and

Hello! Anyone with some insight into the oh323 channel driver please shed some light on the code block below from wrapendpoint.cxx. When enabling trace on the channel driver i get this, for me, strange debug info: WrapH323EndPoint::OpenAudioChannel: Direction => PLAYER, Buffer => 320 WrapH323EndPoint::OpenAudioChannel: FrameSize 8, FrameTime 8, TimeUnits 8

Hi all, Previously, I had the PrivacyManager working for me exactly as would be expected, but after upgrading the OS to Debian lenny and Asterisk to v1.4.21.2 that's no longer the case. Anonymous callers are still confronted with the PrivacyManager, but now no matter what I set the minlength value to, e.g.: exten => jaap,n,PrivacyManager(1,1) ... (I'm not using a

I'm building an app that will do the following: 1. Force the caller to record their name. 2. Dial the party to call. 3. Play a short menu: 1 = Accept Call 2 = Decline Call, go to VM if available 3 = Accept Call forever, never ask again 4 = Decline Call forever, block number, get rid of caller 4. do things based on that choice. I'm

I have been having problems getting PrivacyManager to work correctly. Right now I am running the 1/21/05 CVS but I have been unable to get this to work on asterisk-stable either. You can see from the debug below that the inbound call is arriving via IAX2 and both the CALLING NUMBER and CALLING NAME are both set as "Unavailable". However, PrivacyManager executes and determines that

Hi all, I try to use the hdf5 package. I downloaded the Windows binary version, and it claims it requires hdf5dll.dll library, so I download the whole hdf5 1.4.3 package from NCSA. Unfortunatelly, it crashes R. I suspect it is because the package was probably compiled against version 1.2 of hdf5 (as evoked in the help file?) or any other previous version, so I downloaded the source of the package

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

Hi all, My latest Asterisk system is based on Debian squeeze with Asterisk 1.6.2.6-1 and SIP only. One of my favorite features that I had working with Asterisk 1.4 is the PrivacyManager. However, this was not straightforward, because anonymous SIP calls arrive with ${CALLERID(num)} = "anonymous", instead of being blank. So, to get it to work I added the first three rules to

Hi I can't for the life of me work out why this is not working. When in the campon contect if you hit a DTMF key 2 you get moved to the exten => 2 defined in the mainmenu context not the exten => 2 defined in the campon context. What is wrong? The same happens if you hit key 1. [campon] exten => _*1XXX,1,Answer exten => _*1XXX,2,SetCallerID(${CALLERIDNUM}) exten =>

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, 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 encountered a DTMF issue. My scenario: Access carrier-----sip----> Asterisk-1.4.25.1-----sip---->CiscoGW-----ISDN----->TDM Switch the access carrier sends to Asterisk out of band (rfc2833) dtmf, Asterisk forwards it with SIP INFO method to Cisco gateway, but on TDM switch every digit is duplicated. Is it possible that the carrier sends inband along with rfc2833? Kind

Hi, why would Asterisk core dump with the following test dialplan extension ? exten => 8100,1,Answer() exten => 8100,n,Set(CALLERID(all)="") exten => 8100,n,PrivacyManager() exten => 8100,n,GotoIf(${[${PRIVACYMGRSTATUS} = FAILED]}?:nocid) exten => 8100,n,NoOp(Number is ${CALLERID(num)}) exten => 8100,n,Hangup() exten => 8100,n(nocid),Playback(vm-goodbye) exten

I'm having a problem with Asterisk picking up the Zap/1 and thinking its a new call when instead I've already been on the phone talking to a person. This is not my ideal setup and currently I have just an FXO card and Asterisk is in parallel with my phone system instead of being in the front. I'm not sure if the problem would be fixed by adding an FXS card and putting my phones on

I've got a bit trouble with callerid and zaphfc cards. Basically zaphfc doesn't add the "0" in front of national numbers (haven't tried a international call yet). With chan_capi that allways worked fine, however i had to define the national and international prefixes in capi.conf. Is there something similar in zapata.conf ? Here is my zapata.conf: [channels]

Hello, >The ONLY issue I have is that I don't get ringing dialback so >calling out gives a silence until the other party picks up Have you turned on early B3? S,1,Dial,CAPI/12345678:b${EXTEN}|30 (always early B3) >(plus the recent changes to locks in * required a tweak to the >chan_capi source to match). could you post this tweak here? I'm stuck with CVS-STABLE ;-(

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