RE: [Asterisk-Users] Sending DTMF after a call is set up

: [Asterisk-Users] Sending DTMF after a call is set up Bill Hamlin wrote: > I'm using Dial to place a call to a PBX. But then I want to wait a few > seconds and dial an extension. Dial doesn't return until the call is > disconnected though. Try this posting: http://www.voip-inf

RE: [Asterisk-Users] Sending DTMF after a call is set up

sterisk-Users] Sending DTMF after a call is set up n Fri, 18 Feb 2005 16:06:54 -0500 "Bill Hamlin" <[EMAIL PROTECTED]> wrote: > > I'm using Dial to place a call to a PBX. But then I >want to wait a few > seconds and dial an extension. Dial doesn't retur

[Asterisk-Users] Sending DTMF after a call is set up

I'm using Dial to place a call to a PBX. But then I want to wait a few seconds and dial an extension. Dial doesn't return until the call is disconnected though. I also want the caller to not hear any audio until the DTMF has been sent. This gets the caller to the right place and he doesnt have

RE: [Asterisk-Users] SIP client cannot connect to Asterisk

Try setting the SIP signalling port in your client to something other than 5060 (eg ) and run tethereal on your Asterisk box to see if you're getting packets on . -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of K Wong Sent: Monday, December 27, 2004

[Asterisk-Users] restricting SIP access to asterisk

How do you set up Asterisk to allow SIP call requests from specific IP addresses? We have no control over what account (From: header) is used. We want to be able to allow calls based on the IP address the INVITE comes from, not the account. Is there a way to do that? __

[Asterisk-Users] astcc newbie question

not 10 digits. How come it thinks it is 12 digits? I set both the Published number and DID in the Brand to 9175954700. Was that the right thing to do? Maybe it's not recognizing the DID? Thanks, Bill Hamlin ___ Asterisk-Users mailing list [EM

[Asterisk-Users] Asterisk as PSTN gateway

re has any installations using this and what their opinion is about it (does it work? how's the audio quality? and so on). Thanks, Bill Hamlin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBS

[Asterisk-Users] 3-way calling

laced with the *+(some key). Is this implemented? How would I configure this? Thanks for any help, Bill Hamlin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update opt

RE: [Asterisk-Users] question about CPU usage

el > Emerging Voice Technology Inc. > Palo Alto, California and London, England > > Email: scott "at" evtmedia.com > URL:www.evtmedia.com > > > > >-Original Message- > >From: [EMAIL PROTECTED] > >[mailto:[EMAIL P

RE: [Asterisk-Users] Snom 200

You must have port mapping in the Linux NAT that allows the SIP-level packets to get to the * Server, so you need to add a port mapping for the RTP packets. I may be wrong but I think * sends RTP on the same port that it receives RTP on, so once the phone sends some RTP to * then the RTP coming ba

RE: [Asterisk-Users] question about CPU usage

TED] > [mailto:[EMAIL PROTECTED] Behalf Of Steven > Critchfield > Sent: Monday, March 22, 2004 4:07 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] question about CPU usage > > > On Mon, 2004-03-22 at 14:49, Bill Hamlin wrote: > > I've had my asterisk

[Asterisk-Users] question about CPU usage

e but otherwise polling like crazy? Thanks, Bill Hamlin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Problem playing the first voice mail prompt

t;. I have a system that has no Zapata cards at all. Do I need to have one? Any ideas as to what may be wrong? Thanks, Bill Hamlin Globalnet ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users