Hi,
If 't' is set on Dial command, but then i set canreinvite=yes on the account
[100]
type=friend
host=dynamic
nat=yes
secret=100
canreinvite=yes <--- if i set this
would asterisk still stay in the path?
regards,
nhadie
Mark Michelson wrote:
> 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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