I'm using Asterisk a SIP Server for a lot of GrandStream HandyTone ATA's. Each one of them is configured in sip.conf as: [1234567] type=friend username=1234567 secret=1234567 callerid="ATA 1234567" host=dynamic nat=yes qualify=yes disallow=all allow=g729 canreinvite is set globally to YES. When one ATA calls another, I see the next traffic on Ethereal (just shown the sequence between one ATA and Asterisk) 1.- ATA1 -> INVITE -> Asterisk 2.- ATA1 <- 100 Trying <- Asterisk 3.- ATA1 <- 180 Ringing <- Asterisk 4.- ATA1 <- 200 OK <- Asterisk 5.- ATA1 -> ACK -> Asterisk 6.- ATA1 <- INVITE <- Asterisk (REINVITE, with the IP of ATA2 as Connection Information) 7.- ATA1 -> 200 OK -> Asterisk 8.- ATA1 <- ACK <- Asterisk Until here, all looks normal. However, the strange is that, IN SOME CASES (NOT ALL THE TIME), in packet 9 of the sequence, I see the following: 09.- ATA1 <- INVITE <- Asterisk (REINVITE, with the IP of Asterisk as Connection Information) 10.- ATA1 -> 200 OK -> Asterisk 11.- ATA1 <- ACK <- Asterisk Is this normal behavior? In what cases Asterisk generates that second REINVITE? The problem with this is that, in this case, my ATAs lost sounds, like if the RTP traffic is going to the moon instead the other party or the SIP server (I'm not using transcoding in this case, since both ends support G729) Thanks a lot for your attention and help. -- Atly. Alvaro Palma