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how to make asterisk ring

When people call into our phone system  the caller doesn't hear any ringing... (and if they don't know us, they hang up which is a great way of getting rid of tele marketers)

its something to do with my config as it happened after I changed everything to make all the phones ring at the same time.

The way i have the system set up is like this:

1 sip account to a sip provider for our incoming & outgoing calls to the rest of the world

4 internal sip extensions, 3 use x-lite and 1 uses a ATD for an analogue wireless phone.

when a caller dials our phone number all our phones ring at the same time and whoever picks up first answers it.

the config below is just the parts I can see which may be related to this issue,

Cheers to any helpers :)


# this is the extension or whatever you call it of when phone calls come from the outside world, they end up where which makes our phones ring:
 
[from-sipout]
 
exten => 1234,1,Answer
exten => 1234,2,Dial(SIP/tim&SIP/chris&SIP/ata&SIP/deva)
;exten => 1234,3,Dial(SIP/timcell,2;0,tr) ; this calls my cell phone if no one else answers but not used.
exten => 1234,5,Voicemail(222)
exten => 1234,6,Hangup

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timbo007
Asked:
timbo007
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1 Solution
 
timbo007Author Commented:
perhaps I can explain myself better with some more examples... the [from-sipout] is called from where the incoming phone line comes from and here is the parts in the sip.conf which match.. although there is no reference to from-sipout except where it says context...

[sipout]
type=friend                      
secret=XX
username=09974XX                  
fromuser=09974XX                  
fromdomain=tim      
host=sip.kiwilink.co.nz
;usereqphone=yes                  
call-limit=2                        
;outboundproxy=proxy.provider.domain      ;
canreinvite=no
qualify=yes
context=from-sipout
;context=maincontext
;disallow=all
;allow=alaw
insecure=invite

[general]
...
register => 09974XXX:XXXXsip.kiwilink.co.nz/1234 ; 202.180.76.167
...


so that ext 1234 handles all the incoming calls i guess.. which somehow goes straight to [from-sipout]...
you can see i've really messed with it to get it going... mostly trial and error :)
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feptiasCommented:
Try removing the line:
exten => 1234,1,Answer
...and then adjust the step numbering, of course.

The simplest way would be to replace it with this:
exten => 1234,1,Noop
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timbo007Author Commented:
haha, so you mean when it says 'answer' it means it picks up the line and then goes to the next step which just dials all our extensions hence its silent because nothing is being sent back to the caller...? what it originally did was play an mp3 which was the hold music, i'd prefer that to happen, is it easy to do? but hey you've fixed my problem, funny I never tried that, I tried so much else :)
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timbo007Author Commented:
thanks!
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feptiasCommented:
I often use Asterisk as a gateway, converting SIP to E1 or vice versa. For this, you never want the call to be answered by Asterisk, only to be routed onward. If the call is answered by Asterisk, then the caller starts to pay for the call from that point.
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