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Do you know if this can be achieved with the standard sip stack in asterisk? Kind Regards Andrew Colin Converged Telecoms (Pty) Ltd. Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09) Switchboard: +27 (0)10?591 4600 Email:? andrew at convergedgroup.net Web: ?http://www.convergedgroup.net 75 Witkoppen Road, Northriding, Johannesburg, 2169 P O Box 7246, Weltevredenpark, 1715 This communication is confidential and intended solely for the addressee(s). Any unauthorized rev...

...ween extensions using g729 give one way audio As soon as we change the codec to ALAW the issues goes away. Any ideas how to fix this? Outbound calls via a trunk work fine with g729 Kind Regards Andrew Colin Converged Data (Pty) Ltd. Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09) Direct: +27 (0)10 591 4607 Mobile: +27 (0)82 310 3007 Switchboard: +27 (0)10 591 4600 Email: <mailto:andrew at convergedgroup.net> andrew at convergedgroup.net Web: http://www.convergedgroup.net 75 Witkoppen Road, Northriding, Johannesburg, 2169 P O Box 7246, Weltevreden...

Hi Guys I am trying to write a macro for a call return so for example Anyone in the company transfers a call to another extension and it is not answered etc it must return to the person who did the transfer I have got it working but if the call originates externally for example someone calls in to the switchboard and they transfer it then it tries to return to the outside caller. As

Hi Joshua If i put the default_user option per endpoint would it work?? So what exactly does the contact_user option do? I know that in freeswitch there is the option extension-in-contact.We ?basically need to achieve the same functionality? Thanks<div> </div><div> </div><!-- originalMessage --><div>-------- Original message --------</div><div>From: