[Asterisk-Users] double-dial in SIP Grandstream

Paul Liew pliew at atp.org.au
Tue Nov 18 14:27:31 MST 2003


"callwaiting=no" is not supported by chan_sip. Call waiting
enabling/disabling is a function of SIP phones. Unfortunately, GS does not
support disabling call waiting as yet, so I've had to put in a patch to
overcome the problem. Look under
http://bugs.digium.com/bug_view_page.php?bug_id=0000408. You need
"incominglimit=1" to stop the ringing caused by callwaiting when you are on
the phone.

Paul
----- Original Message ----- 
From: "Bisker, Scott (7805)" <sbisker at harvardgrp.com>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, November 19, 2003 12:57 AM
Subject: RE: [Asterisk-Users] double-dial in SIP Grandstream


> Marc,
>
> This is the typical behavior for call waiting.  While you are initiating a
> call, people who call your number will get a busy signal until your first
> call connects.  Once the call connects, the number 2 caller will hear a
ring
> until you pickup.
>
> If you want to disable callwaiting then put "callwaiting=no" in sip.conf
for
> that particular alias.
>
> [<alias>]
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>




More information about the asterisk-users mailing list