[Asterisk-Users] Strange Call waiting problems - SNOM 200 Grandstream Budgetone

2004-01-08 Thread Michael
Hi 

I am setting up an Asterisk System in an office environment, Incoming and 
Outgoing calls are working ok, but i am having a few strange problems 
regarding call waiting.

With the SNOM 200 (firmware 2.02t) phones, if you are on a call and a 2nd call 
comes in, the call waiting beep is played and the light flashes, but if you 
hang up the 1st call, instead of the phone ringing, it connects the 1st call 
and the 2nd call together! 

The 2nd caller ends up speaking to the first caller (almost like a conference 
call, but you cannot hear or speak to the 1st or 2nd callers) as you can 
imagine, this could cause a few serious problems :-( . we have not been able 
to recreate this problem with the grandstream, but we are having another 
strange problem with the Budgetone.

if you are on a call on the Budgetone 101 and a 2nd call is received, instead 
of a call waiting beep being played, it rings on the handset speaker! which 
makes it almost impossible to speak to the 1st caller, but if you hang up the 
1st call, the phone rings and it is possible to answer the 2nd call normally.

Anyone got any ideas on what could be causing these problems ?




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Re: [Asterisk-Users] Strange Call waiting problems - SNOM 200 Grandstream Budgetone

2004-01-08 Thread Paul Liew

- Original Message - 
From: Michael [EMAIL PROTECTED]
Sent: Thursday, January 08, 2004 10:40 PM


 if you are on a call on the Budgetone 101 and a 2nd call is received,
instead
 of a call waiting beep being played, it rings on the handset speaker!
which
 makes it almost impossible to speak to the 1st caller, but if you hang up
the
 1st call, the phone rings and it is possible to answer the 2nd call
normally.

 Anyone got any ideas on what could be causing these problems ?

Hi Michael,

GS does not really support callwaiting, and you don't have the ability to
disable from the phone configuration. You can however disable in '*', by
adding incominglimit=1 for each GS phone in sip.conf. Also ensure that you
a username=blah where blah is the same as your phone definition. HTH.

Paul

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