|
Hi
All, I've
an Asterisk Server behind a NAT. Using
DNAT, I've opened port 5060 and all 10000:20000 udp. Sip configured
with externalip and subnet. I've
another site, also with NAT, where I map the rtp port (as defined in the
client) to map to the local client (DNAT). Using
Xlite, this configuration works, it requires using the quality=yes and NAT=yes/always
in the sip ext configuration but works quite well. However,
lately I've purchased a Grandstream ATA Handytone 286 and tried to apply the
same settings but… When
doing an echo test, I can't hear myself, but I can hear the asterisk server
(meaning asterisk can reach the client behind the NAT). When doing
some tcpdump, it looks like some packets are coming from the client to
asterisk, so the network setting looks ok. When calling
to another sip device, with or without canreinvite (yes/no) the rtp stream is
unable to establish it self, no matter where the second client is
(inside/outside NAT). But! When
calling using a zap channel (which is on the asterisk server) everything works!
I can hear the person I'm talking to and he can hear me. I'm a
bit confused….. How could it be that this works and echo test doesn’t? Any
help would be appreciated! Thanks, Ohad |
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