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
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