[Asterisk-video] Bridging 3g call with SIP

[Vadim Lebedev]

In the sip.conf

in the section describing public ipphone add
canreinvite=false

Thanks
Vadim
Le 10 avr. 09 à 02:46, Reza Fatahillah a écrit :

>
>
> --- On Thu, 4/9/09, Emmanuel BUU <emmanuel.buu at ives.fr> wrote:
>
>> Reza Fatahillah a écrit :
>>> Is there solution to fix my problem?
>>>
>> yes : in sip.conf, you may use two parameters : externip
>> and localnet.
>>
>> The first should be used to describe the private network
>> that you are
>> connected to and the second should be used to specify you
>> public IP address.
>
> Isn't it other way around? externip should be my public ip, and  
> localnet is my subnet?
>
> externip=mypublicip
> localnet=172.x.x.0/255.x.x.0 ;subnet
>
> I tried your method;
> Sip exchange happened at my private ip, the connection ip at SDP was  
> set to my private ip.
>
> But no rtp stream received at my public ip.
>
> I tried this to:
> externip=mypublicip
> localnet=172.x.x.6/255.x.x.0 ;real privateip
>
> I received rtp stream at my public ip, but still cannot see anything  
> at my handphone
>
> Thanks
>
>
>
>
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