similar to: What to use instead of LookupCIDName?

I have achieved much of what I wanted to achieve using Wine, with one exception. I was hoping to be able to use Shockwave content, but despite installing Firefox and the Shockwave plugin, it does not work - it seems to stop after the promotional film before the actual requested content. I know there are "pay" solutions to this problem, but was wondering if anyone had found a free

Dear all, I tried to make a call with extensions.conf. exten=> _00[1-9].,1,Dial(zap/g1/${EXTEN}) exten=> _00[1-9].,2,NoOP(ANSWEREDTIME=${ANSWEREDTIME}) exten=> _00[1-9].,102,Hangup But the 2 and 102 will not be executed. So I can get the correct answered time via 2. Is any idea about it? Is it the problem of my ZAP channel's configuration? My zapata.conf is as below:

I am now at the point where I have one more thing to install before I can scrap my Windows box - and it's Leisure Suit Larry's Greatest Hits and Misses. I have installed the DOS-based parts of it in dosbox, and also used sarien and freesci to handle some of it, but there is a significant part of this collection that needs Windows. Trying an installation with the default Windows version

Hi, I am using asterisk 1.4.18. I am using it for inbound only call center. The SIP phones are X-Lite. Right now when a call is proxied by Asterisk to X-Lite the agent only sees asterisk written on its CLI screen. I want the agents to be able to view the callees number. Is there any way to make this happen. Regards Syed Nasruddin -------------- next part -------------- An HTML

Hello, I'm having a problem in voicemail check attempts from SIP-based phones. I've searched a ton of docs but don't see anyone else having a similar issue. I have a TDM22B with two non-sip phones connected to it as well as several SIP phones including a GXP-2000 and some X-Lites. Users of the real phones in the same context can pick up and dial *8 to get to VoiceMailMain() just

Hello, I am using the SetCIDNum dialplan application on 1.2 and 1.4.0; I've tried it on 1.4svn 56126 and it does not recognise this application. Any idea?... Thanks! __Yehavi:

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 I have a working wine installation one one computer including several installed programs. Now I would like to move the .wine folder, which as I understand it contains all the configuration details and windows file and registry, to another conputer. What is the easiest way to do this? I installed already the same wine version (0.9.3) on the other computer. How can I move the .wine

Hi all, With the gamut of FXO cards out there, I'm looking for a recommendation for home use. I have a nicely working Asterisk 1.4 system that just requires an FXO card to connect my NTL PSTN to it. My previous X101P clone seems to have kicked the bucket. Any suggestions would be greatly appreciated. Regards Kyle -- Kyle Gordon kyle@lodge.glasgownet.com http://lodge.glasgownet.com

What are peoples experience with the reliability of the TDM400p. Specifically in the 2 FXO, 2 FXS configuration, which is the 022 (?) model. Is this board prone to random failures? joe a.

Hi All, If I had a large block of code, eg: [outgoing-pstn-gradwell] ; the caller ID convertion assumes that the last two digits of the callers id ; are mapped to the last two digits of the PSTN number. exten => _0.,1,ExecIF($["${RECORDOUTBOUND}"="TRUE"],Monitor,wav|${TIMESTAMP}-${CA LLERID(num)}-${EXTEN}-${UNIQUEID}.WAV) exten =>

I am trying to install RepliWeb R-1 Console, but it is failing. I have a self-extracting installer that I used on my Windows-based laptop but when I try to run that with Wine, this happens: I do: wine R1_win.exe Some extraction happens, then: File not found C:\windows\temp\rw_products>.\check_before_install.exe err:ole:CoGetClassObject class {88d969c0-f192-11d4-a65f-0040963251e5} not

A few months ago, I tried to get Datakam Player (Registrator Viewer) to work on my Linux system using wine, without success. I have since installed a newer wine, and it now starts and can open the files created by my dashcam, but it does not play them - the video area remains black, there is no sound and the speed and location displays do not change. Short of trying an even newer wine, which

Godd evening! What happens if someone calls and asterisk doesn't "Answer()" itself, but another analog phone does? Can I somehow catch this situation in my dialplan. I have an ISDN line, but with it I got a box with an adapter for good old analog phones. This doesn't seems to be directly connected to the ISDN line asterisk sees. But somehow, it must know, that the call

I have set my Asterisk server up to connect to my Jabber server and send messages with the caller ID details in them to the recipients of incoming calls - this is working very nicely. There are a few other things I can think of right now that I would like to send to Jabber but as yet I do not know whether they are possible. They are: (a) a count of messages in a voicemail box - triggered

Greetings list, I'm trying to establish if there's an issue whereby certain telcos in certain countries have not updated the London, UK numbering plan to include some parts of the 020 3 range, despite it being in operation for some two years now. To help with this, I'd be most grateful anyone outside the UK could make a test call to +44 203 3393 7389. This is a simple test number

This sort of message is usually preceded by some magic tones that allow direct marketing application to immediately drop a call to a dead phone number. What is the proper terminology for the tones? Where can I find information about how this is implemented? -- Drew Einhorn

Hello, I'm trying to fill CID data into the astdb using AsteriskWin32's asterisk.exe, to no avail: The batch file stops after the first line, and just waits: ---------------------------------------- rem c:\cygroot\mystuff>import.bat rem rem c:\cygroot\mystuff>C:\cygroot\bin\asterisk.exe -rx 'database put cidname 123 "My cellphone"' rem rem Asterisk module

Hello, Sometimes, when a call comes in from the PSTN through our VoIP gateway, the information that is sent to our web page that logs calls includes the original CID name instead of the one that is we expect to be rewritten on the fly using Asterisk's LookupCIDName: ================= ;extensions.conf [internal] exten => group,1,LookupCIDName exten =>

Hi Most of our customers have generic names like "Hospital", so I need to rewrite their caller ID name by looking up the number in a database on the Asterisk server, and rewriting the name such as "Reading Hospital" so that we know who's calling. Any idea if this can be done with Asterisk, and how to do it? Thank you.