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
signature.asc
Description: This is a digitally signed message part
_______________________________________________ --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
