Re: [Asterisk-Users] double-dial in SIP Grandstream
On Wed, Nov 19, 2003 at 08:27:31AM +1100, Paul Liew wrote: > "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=408. You need > "incominglimit=1" to stop the ringing caused by callwaiting when you are on > the phone. > > Paul > - Original Message - > From: "Bisker, Scott (7805)" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Wednesday, November 19, 2003 12:57 AM > Subject: RE: [Asterisk-Users] double-dial in SIP Grandstream Paul's patch has been working great to stop the call waiting on a system I have with 12 GS phones. The example of the sip.conf for the SIP devices are in the bug report that Paul referenced above. -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] double-dial in SIP Grandstream
"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=408. You need "incominglimit=1" to stop the ringing caused by callwaiting when you are on the phone. Paul - Original Message - From: "Bisker, Scott (7805)" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> 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. > > [] > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
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. [] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] double-dial in SIP Grandstream
Hi, I have even now connected to IAXtel at number 1-700-895-5211 when I am in the office, so Asterisk is great. I just found something strange, which is that if I am already in a connection with my Grandstream and talking, and a second call comes in, it rings on the Grandstream. However, if I am not talking but waiting for dialing, the caller gets a busy signal (good). How can I make sure there is only one call at a time to the SIP phone ? (call waiting could be useful, but I didn't figure out how to do this with the SIP). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
