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

[Alex Mosburger]

Hi Ronald!

Please check if the following points are NOT activated. 
* is not using direct phone to phone RTP streams if:

-) either of the clients is configured with canreinvite=no
-) the clients cannot agree on a common set of codecs and * needs to
perform codec conversion
-) either of the clients is configured with nat=yes
-) * needs to listen to DTMF tones during the call (for transfers or any
other features)

Hope this helps,
Alex

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tiago
Stein D`Agostini
Sent: Montag, 17. April 2006 13:13
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP conections,with RTP not going trough
Asterisk

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.

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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