search for: jacksch

Greetings, I'm having a miserable time getting Asterisk working with FWD. All the samples show something like... exten => _7., .... How do I get Asterisk to wait until the user is finished dialing instead of trying as soon as it gets the second digit? I can use _7XXX, and dial the FWD 3-digit test numbers fine, but I'd like to be able to dial others... Same problem for outside

Greetings all, My next step is to purchase a nice VoIP phone for my desk. I have a grandstream, and the sound is great, but I'm looking for more of an office style phone, preferably that can handle multiple lines, has a more flexible display (i.e. name as well as number). SIP would be preferable. Any suggestions? Thanks, Eric

Does anyone know of a company that provides German DIDs (preferably Berlin) and termination of calls to Germany at reasonable rates? Thanks, Eric jacksch@tenebris.ca

Hi, How could I possibly inform incoming callers that the number they'd dialed is monitored and recorded. I wanted that when a call-in or call-out is made, a playback will be played to inform caller & callee that thier line is monitored prior to start conversation. Thanks. Angel __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best

Greetings all, I've tried out chanspy, but what I'm really looking for is the ability to interrupt a call (i.e. barge in for emergency purposes). Has anyone found a way to do that with Asterisk? Regards, Eric -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060403/dfe6e99e/attachment.htm

Greetings all, I've been working on having Asterisk put a call through to two different numbers, and give the call to the first one that acknowledges by pressing the 1 key. I found an example on the wiki, but I can't get it working. When I answer the call I hear the message telling me to press 1 to connect, and as soon as the message is done, the call is connected. In other words, it

Perhaps this will help... I have a phone connected to a QuickNet PhoneJack card. When I pick it up, I get a dial tone. When I dial a certain number of digits, the call is processed by Asterisk. The question: How does Asterisk determine how many numbers to let me dial? I'm banging my head against the desk here... _9XXXXXXX lets me make an outbound call, but _9X. only lets me dial 9 plus

The problem I am running into is that when the call to my cellphone is made, it appears as though the call "completes" so it never rolls to asterisk voicemail. Here is my current config: exten => 102,1,Dial(${sipura},10,) exten => 102,n,playback(pls-wait-connect-call) exten => 102,n,Dial(IAX2/asterisk1/9139275900,10,r) exten => 102,n,VoiceMail(u102@default) exten =>

Hi all, For a top quality setup, I will need to install high quality VoIP switches with QoS and PoE. My potential customer should not have any problem with call quality. Experienced folks, Please advice me what switches to install and at what price. I may need it for upto 100 phones. What else should I consider so that phones work without problem along with the computers on the same network?

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Greetings all, I've just put two X100P cards to my box and despite trying everything I can find on the web, I can't get the second one to work. I've edited /etc/modules.conf so that ztcfg doesn't run automatically. To start, I run: /sbin/modprobe zaptel /sbin/modprobe wcfxo /sbin/ztcfg /sbin/modprobe ixj and ztcfg says: ZT_CHANCONFIG failed on channel 2: No such device or