[asterisk-users] anoyingly answers already in use pstn line

[Gleim, Jason]

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of Jack Bates
> Sent: Friday, October 17, 2008 4:48 PM
> To: asterisk-users at lists.digium.com
> Subject: [asterisk-users] anoyingly answers already in use pstn line
> 
> I am using Asterisk and an X101P card as a glorified answering
machine.
> We have a residential PSTN line with about six phones connected to it.
> Like an answering machine, I want Asterisk answer the line *only* when
> an incoming call is not answered after four rings.
> 
> This mostly works. My extensions.conf is at the end of this message.
> 
> The problem is that Asterisk will sometimes answer the line when
> someone
> is already talking on one of the six phones connected to it. Sometimes
> Asterisk will answer the line and start playing the greeting in the
> middle of a conversation! This is especially a problem when I am
> talking
> on the phone to an automated system, because although I hang up the
> phone I am talking on, neither the automated system nor Asterisk will
> hang up.
> 
> I have not yet discovered a pattern to when Asterisk answers the line.
> It always answers after four rings, but it sometimes answers when
> someone is already talking on one of the phones connected to the line.
> 
> In a perfect world, Asterisk would be the only thing connected to the
> line, and all our phones would be Asterisk extensions. Unfortunately
we
> do not currently have the required VoIP phones or FXS interface...
> 
> Is there any way to make Asterisk less flaky, and answer the line
> *only*
> when an incoming call is not answered after four rings?
> 
> ---
> 
> [default]
> 
> exten => s,1,Wait(20)
> exten => s,n,Answer
> exten => s,n,Background(recordings/coop-greeting)
> exten => s,n(instruct),Background(recordings/leave-message)
> exten => s,n,Background(recordings/enter-extension)
> exten => s,n,Background(recordings/dial-by-name)
> exten => s,n,Background(recordings/visit-website)
> exten => s,n,WaitExten
> 
> ; General delivery mailbox
> exten => #,1,Voicemail(6000)
> exten => #,n,Goto(s,instruct)
> 
> ; Dial by name
> exten => a,1,Directory(default)
> 
> ; Entering an invalid extension replays the instructions
> exten => i,1,Playback(invalid)
> exten => i,n,Goto(s,instruct)
> 
> ; Timeout goes to voicemail
> exten => t,1,Goto(#,1)
> 
> exten => 6003,1,Macro(stdexten,6003,SIP/cstewart)
> exten => 6004,1,Macro(stdexten,6004,SIP/mhockley)
> exten => 6005,1,Macro(stdexten,6005,SIP/jbates)
> [...]
> 
> 
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Others may wish to chime in and confirm or deny this but the card is
probably getting confused by you loading the line with the other phones.
I know most of the analog cards I've worked with (which does not include
the X101P) really get cranky if there is anything else hanging off that
line. The only solution I've seen to the problem is to change things
around so that the card is the only thing on the line.

In know you said you haven't switched to IP or FXS but is there a reason
why? Your problem would go away and you would be able to leverage all
the features of Asterisk if you just got a single ATA. Something like a
Linksys PAP2T-NA can be had for around $55 USD. Disconnect your PSTN
line at the entrance bridge, run it into the X101P, and plug the PAP2T
into the house. It is convenient and doesn't require any changes in
internal wiring. (You might have to run a few wires if the bridge is on
the back of your house.) No need for new phones or anything. Granted,
all the internal phones would be on one extension but you have that
situation now... And with the ATA you've solved your problem. As the
need arises, get more ATAs or IP phones or whatever and build out your
internal phone network.

Jason