[Asterisk-Dev] native bridge woes

[brian]

Does seem to drop out for like half a second?

If so then look in rtp.c for a usleep of 500000 and change it to like 5000
to see if that helps :)

bkw

> -----Original Message-----
> From: asterisk-dev-admin at lists.digium.com [mailto:asterisk-dev-
> admin at lists.digium.com] On Behalf Of mjr-asterisk at ranney.com
> Sent: Friday, May 21, 2004 3:01 PM
> To: asterisk-dev at lists.digium.com
> Subject: Re: [Asterisk-Dev] native bridge woes
>
> 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
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