Dinesh Nair wrote:
> On Thu, 11 Oct 2007 21:32:28 -0300, Alejandro Facultad wrote:
>   
>> But when I establish a call between two Wengophones, I sniffer the 
>> communication and I realize that SIP+RTP packets go among clients and 
>> Asterisk always (RTP does not bypass Asterisk).
>>     
>
> asterisk will no do a reinvite when it needs to be in the middle of the
> media stream for any reason, such as call transfers being activated or
> call recording being activated. what's the arguments to the Dial
> application you're giving to asterisk ?
>
> (actually, this should be sent to the asterisk mailing lists instead)
>
>   
Hi, I write in name of my colleague Alejandro. We have Asterisk with
"canreinvite=yes" and "nat=no" for every user of sip.conf file and this
is our dialplan you request (with the Dial application):

[company]
exten => 1111,1,Dial(SIP/user1)
exten => 1112,1,Dial(SIP/user2)
exten => 1113,1,Dial(SIP/user3)
.......

The Advanced options of the Wengophone control panel are by default, and
I check/uncheck the "Register through proxy" account option but it's the
same (by the way, we don't know the use of this option and it's not
explained anywhere).

Any help to get RTP data packets bypassing the Asterisk and go among SIP
clients ???

Thanks a lot for your support.

Alejandro



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