[Asterisk-Users] Problem with Asterisk and Grandstream HT286

[Álvaro Palma]

I've dealing with this issue for a while, and I'd really like to know if 
anybody has experienced the same pain before :-)

I've a lot of Grandstream HandyTone 286, loaded with the latest firmware 
(1.0.8.16) from the GS website. In my sip.conf, this ATA's are 
configured as:

[05]
type=friend
username=05
secret=XXXX
callerid="User 05"
host=dynamic
nat=yes
qualify=yes
disallow=all
allow=g729
callgroup=1
pickupgroup=1
canreinvite=yes

Also, in the ATA's configuration, I've set up the RTP port to random.

The problem is that, without any notice, in some (NOT ALL) of the calls 
between ATA's, they keep mute after the call has started. I've been 
tracking the problem with Ethereal, and it seems to be an issue related 
to the REINVITE sequence sent after the communication has been 
established. I've also been emailing the people of GS, and they tell me 
that the problem is the definition of the REINVITE sequence in Asterisk, 
which, according to them, is buggy, so their solution is to use a "real 
sip proxy, like SER".

Is there something true in this story, or simply the ATA's SIP sequence 
handling is the buggy one? Has anybody suffer this problem before? BTW, 
there's no NAT issues involved in the audio path, since all the 
equipment (ATA's and Asterisk server) are in the same LAN segment.

Thanks a lot for your attention.

-- 
Atly.
Alvaro Palma