[Asterisk-Users] SIP conections, with RTP not going trough Asterisk

[Ronald Wiplinger]

Tiago Stein D`Agostini wrote:
> Hi, sorry to bother again. But I still cannot make it work. I made all 
> acounts have canreinvite=yes, but found no option in Dial aplication 
> to make the phones exchange RTP directly between them.  Can anyone 
> tell me wich option should I look at? I am stuck with this (probably 
> simple) problem for almost a whole week.
>

What is exactly your dial command?


bye

Ronald Wiplinger
> Thanks for any help.
>
> Ronald Wiplinger wrote:
>
>> Tiago Stein D`Agostini wrote:
>>
>>> Hi,
>>>
>>>   Ie been looking for some time how to use asterisk  to initiate SIP 
>>> connections between 2 IP phones,  but afetr initiated the 
>>> communication making the RTP go directly from one telephone to the 
>>> other, without passing by asterisk. Unfortunately I found no 
>>> explanations of how to do it.
>>>
>>> Does anyone care to give a pointer to any explanation about how to 
>>> do it?
>>>
>> canreinvite=yes
>> and look at the options for dial()
>>
>>>
>>> Thanks in advance
>>>
>>
>>
>> bye
>>
>> Ronald Wiplinger
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