[Asterisk-Dev] native bridge woes

[mjr-asterisk at ranney.com]

Fran Boon <flavour at partyvibe.com> writes:

> Always 'Answer' & then 'Wait,1' before starting whatever is required.
> 
> e.g.:
> exten => 101,1,Answer
> exten => 101,2,Wait,1		; Allow VoIP phones a chance to initialise
> exten => 101,3,Dial(SIP/101,20)

Sadly, this did not work for me.  As soon as the call is answered on
the other end, it native bridges, along with the audio dropout.

    -- Executing Answer("SIP/3122618252-9178", "1") in new stack
    -- Executing Wait("SIP/3122618252-9178", "1") in new stack
    -- Executing Dial("SIP/3122618252-9178", "SIP/NUMBER at HOST") in new stack
    -- Called NUMBER at HOST
    -- SIP/HOST-8938 is making progress passing it to SIP/3122618252-9178
    -- SIP/HOST-8938 answered SIP/3122618252-9178
    -- Attempting native bridge of SIP/3122618252-9178 and SIP/HOST-8938

Any other suggestions?
-- 
Matt Ranney - mjr at ranney.com