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 Gutiérrez 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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