[asterisk-users] voice detection during playback

or say one. Then there is a pause of two or three seconds, followed by leave a message ofter the tone. That pause breaks AMD. If I could detect voice while playing a message I could stop playing the message, wait for silence and restart playing the message from the beginning. Regards, David Koski

[asterisk-users] app_amd and voicemail

every combination I can think of for paramters to AMD. Is this possible? Thanks, David Koski [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

[Asterisk-Users] call fails first time, then succeeds

When making a call through voicepulse, I can hear one ring, then the ring tone changes slightly and it continues forever. I think the first ring actually goes trough. If I hang up and try again it works normally. Any clues? Regards, David Koski [EMAIL PROTECTED

Re: [Asterisk-Users] OT: Where to buy a T1 crossover cable for * and channel bank

On Saturday 19 November 2005 07:47 am, Ed Greenberg wrote: --On Saturday, November 19, 2005 10:26 AM -0400 Chris Mason (Lists) [EMAIL PROTECTED] wrote: The crossover cable is different. Best to make it custom. Is it different from an Ethernet crossover cable? Yes. David

[Asterisk-Users] PolyCom SoundPoint 300 and distinctive ring

I am looking for clues on how to configure distinctive ring for a PolyCom SoundPoint 300. Does ALERT_INFO apply? If so, how? Thanks, David Koski [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com

Re: [Asterisk-Users] dialplan defenition

On Thu, 28 Jul 2005 10:30:15 +0100 Joao Pereira [EMAIL PROTECTED] wrote: snip Then I tried: exten = s,1,Dial(SIP/[EMAIL PROTECTED],30,r) I like to do this: ** extensions.conf ** [globals] MYSIP=SIP/mysipphone [mycontext] exten = _74XXX,1,Dial(${MYSIP}/${EXTEN}) ;exten =

Re: [Asterisk-Users] phantom incomming calls from asterisk

is disconnected. Any clues? David Koski David and List, I am having the same problem. I have an * box at my house with 1 zap (pstn on a X100p clone from digit networks) channel and one sip(linksys ATA). I am getting ring on the ATA but there is no call comming in from the pstn

[Asterisk-Users] phantom incomming calls from asterisk

About once a day I have noticed a phantom incoming call with a caller ID of [EMAIL PROTECTED]cut off. When I answer the call there is a dial tone and the call is disconnected. Any clues? David Koski [EMAIL PROTECTED] ___ Asterisk-Users mailing list

[Asterisk-Users] [newbie] configuration for IAX server to server

. Should I use a trunking? Regards, David Koski [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com

Re: [Asterisk-Users] [newbie] configuration for IAX server to server

around the world. and find no problm at all as long as you have good link. Regards Shams On 6/11/05, David Koski [EMAIL PROTECTED] wrote: I would like to connect two asterisk servers using IAX. I am concerned that a loss of connection with the remote server would cause disruption

Re: [Asterisk-Users] [ADMIN]: subscription failure

, 2005 at 07:37:35PM -0700, David Koski said: Would an admin please contact me off list? I tried to subscribe from another address and it failed--I got no email to confirm the subscription. I would rather use the other address and need to know if there is a problem with my mail server

[Asterisk-Users] [ADMIN]: subscription failure

Would an admin please contact me off list? I tried to subscribe from another address and it failed--I got no email to confirm the subscription. I would rather use the other address and need to know if there is a problem with my mail server. Thank you, David Koski [EMAIL PROTECTED] [EMAIL