[Asterisk-Dev] native bridge woes

[Fran Boon]

mjr-asterisk at ranney.com wrote:
> Native bridging sure seems like a great idea, except every time a call
> is bridged by my asterisk server, the audio cuts out of a second.
> Unfortunately, it cuts out right as the person is saying "hello" so
> its hard to tell who answered the phone or what they said.  I'm using
> Cisco 7940/7960 phones.  Is this a problem with asterisk or a problem
> with the phones?  Is there any way to do native bridging without the
> audio dropout?

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)

That works very nicely for me...

F