[asterisk-users] voice detection during playback
Is it possible to detect voice while playing back a message? I am using AMD (Answering Machine Detect application) and it seems to work pretty well but some outgoing messages (on my Sprint cell phone for example) have silence in them. After the initial message of about 20 seconds it says press 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_amd and voicemail
I installed the latest asterisk from svn to get AMD but so far I have not been successful in getting it to leave me a message on my cell phone. The outgoing message goes for about 20 seconds, has about 3 seconds pause (press or say one pause) and goes again for about 7 seconds. I have tried 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://lists.digium.com/mailman/listinfo/asterisk-users
[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] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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/mailman/listinfo/asterisk-users
[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/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 = _74XXX,1,Dial(${MYSIP}/${EXTEN:2}) ; dials only XXX
** sip.conf **
[mysipphone]
type=friend
etc...
David
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Re: [Asterisk-Users] phantom incomming calls from asterisk
I have no telco line, only IAX2 to another asterisk server and one SIP phone. David On Sat, 09 Jul 2005 15:40:02 -0400 John Novack [EMAIL PROTECTED] wrote: Many telcos do an automated once a day or once a week or ?? line test, which can appear as an incoming call to some devices. If you unplug your telco line and the events disappear, perhaps that is what is happening? John Novack John Millican wrote: 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 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. The following is the CLI output when this happens. I know that there is no call on the pstn because i have an emergency phone(frequent power outages) still connected to the PSTN parallel to the * box and it never rings. All the SIP stuff is on an internal lan only. I only call out on PSTN since all I have available here in nowheare land is dial up :-( All work flawlessly except for this one problem. - Starting simple switch on 'Zap/1-1' Jul 8 13:49:23 NOTICE[6150]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... -- Executing Dial(Zap/1-1, sip/677|35) in new stack -- Called 677 -- SIP/677-55a8 is ringing == Spawn extension (default, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Is there anything for Zap like sip debug? My first guess is that I am getting some sort of blip in ring voltage on the PSTN but have no way to prove this. As a posible logic check I unplugged from PSTN, which put zap into Red alarm of course, and then i get no phantom calls. Is there something in the zap driver that shuts down when in red alarm? Any Ideas? John M ___ 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/mailman/listinfo/asterisk-users
[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@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [newbie] configuration for IAX server to server
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 of the main server. Which is the best configuration? The main reason for the remote is to extend the reach of the main server with a few additional phones. 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/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [newbie] configuration for IAX server to server
I have a good link but I don't want the office phone to degrade or have long delays if the link is broken. What is the main advantage of a trunk? Regards, David On Sat, 11 Jun 2005 13:54:30 +1000 Shamsul Arefin [EMAIL PROTECTED] wrote: We already using IAX trunking to connect our servers 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 of the main server. Which is the best configuration? The main reason for the remote is to extend the reach of the main server with a few additional phones. 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/mailman/listinfo/asterisk-users -- Best Regards Shamsul Arefin Saktek Technologies ___ 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/mailman/listinfo/asterisk-users ___ 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/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [ADMIN]: subscription failure
That is what I needed. Thanks! David Walt Reed wrote: Did you go to the web page that is listed at the bottom of every message? Look at the bottom of that web page for the address of the list admin. By the way, that admin address is pretty much standard for ALL mailing lists. On Wed, Jun 08, 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 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/mailman/listinfo/asterisk-users ___ 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/mailman/listinfo/asterisk-users
[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 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/mailman/listinfo/asterisk-users
