This worked. Great and thanks -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez Sent: Wednesday, February 07, 2007 5:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] After upgrade to 1.4 transfers don't workproperly
On Wed, 2007-02-07 at 14:12 -0600, Savoy, Kevin - Williston, ND wrote: > I have discovered an issue on my system after upgrading from 1.2.13 to > 1.4. A call comes in on a T1 line and goes to a Polycom 501 SIP phone. > I have confirmed this on multiple phones. When the called person > answers and tries to transfer the call to another extension, the call > successfully transfers, however the person answering the transfer > cannot hear the person that called in, the caller. My dial command > simply is > > > I had exactly the same problem when upgrading to 1.4 and I solved by making sure canreinvite=no is in sip.conf for every phone. > -- Telecomunicaciones Abiertas de Mexico S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users