Run ethereal and look the dump, prehaps A) the SIP invite doesn't match
the correct IP & port B)try turning on Asterisk's NAT fix C) send the
dump to me :)
-Adam
Alejandro Guti?rrez wrote:
> Hi!.
> I am testing firefly and I can say it's a great
> program, but I have a problem.
> When I use Sip and I activate the "canreinvite" option
> in Asterisk, I can't hear anything.
> My network is the following:
> -Two Firefly clients with SIP. Each firefly is in
> different networks behind NAT.
> -One Asterisk server with a public IP.
>
> First, I tested my network with canreinvite=no.
> Everything was perfect, the voice quality was quite
> good.
> After that, I changed to canreinvite=yes, and I
> could't hear anything.
> I thought that my routers might be stopping the voice
> streams, but I ran Ethereal and I could see the voice
> was arriving to my boxes.
> With IAX, canreinvite works but nowadays SIP phones
> are majority :(. Any ideas?.
>
> Thanks in advance.
>
>
>
>
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