Nhadie wrote:
> Hi,
>
> I encountered something i can't understand. I've setup 2 extensions.
>
> [100]
> type=friend
> host=dynamic
> nat=yes
> secret=100
>
> [101]
> type=friend
> host=dynamic
> nat=yes
> secret=101
>
> and on extensions.conf
>
> exten => _1XX,1,Dial(SIP/${EXTEN}|30|t)
> exten => _1XX,n,Hangup
>
> This dial plan is ok, audio connects both ways.
> but when i had a typo error, i forgot the 't' option, only one way audio
> when i call, 't' option is used to transfer call how come it affected
> the audio?
>
> thank you in advanced
>
> regards
> nhadie
>
The 't' option is one that requires Asterisk to be in the media path of the
call
(so that Asterisk can tell when the transfer DTMF has been pressed). In order
to
stay in the path, SIP reinvites are disabled for the call. Without the 't'
option, Asterisk will send reinvites to the phones so that their media does not
go through Asterisk at all.
In order to figure out why there is one-way audio, you would need to provide a
sip debug of the call. Based on the fact that you have "nat=yes" for both SIP
friends, I'm guessing that there's some sort of NAT issue here, but I can't be
certain.
Mark Michelson
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