Hello.
Thanks for the reply.
Yes. In the traffic analyzed, the BYE is sent by the originator of the
call, but there is no "human" hangup, but the asterisk one.
BYE is sent, received and confirmed.
I don't know how I could investigate the reason for this BYE.
Em 21/09/2020 17:12, Dovid Bender escreveu:
Is there anything in the Asterisk logs? Which side sends the BYE? Were
you able to capture the traffic with sngrep/wireshark to see if any
side stopped sending/getting RTP? What did the other side see?
On Mon, Sep 21, 2020 at 3:22 PM Roberto
<roberto.med...@gasparimsantos.com.br
<mailto:roberto.med...@gasparimsantos.com.br>> wrote:
Hello
I have an asterisk 16.2.1 on an ubuntu on AWS, which is
experiencing a
drop in call. It does not have a certain time, it is random. The
audio
is flowing normally and the call is dropped.
Has anyone ever experienced this?
My settings changed below:
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = no
tcpbindaddr = 0.0.0.0
transport = udp, ws, wss
srvlookup = yes
directmedia = no
rtcachefriends = yes
externaddr = my ip address
externhost = my domain address ; foo.dyndns.net
<http://foo.dyndns.net>; refreshed periodically
externrefresh = 180
localnet = 172.31.40.21 / 255.255.240.0; AWS NETWORK
localnet = 192.168.0.0 / 255.255.0.0; RFC 1918 addresses
localnet = 10.0.0.0 / 255.0.0.0; Also RFC1918
localnet = 172.16.0.0 / 12; Another RFC1918 with CIDR notation
localnet = 169.254.0.0 / 255.255.0.0; Zero conf local network
localnet = 200.0.0.0 / 24
localnet = 191.0.0.0 / 24
localnet = 201.0.0.0 / 24
localnet = 177.0.0.0 / 24
localnet = 179.0.0.0 / 24
Thanks
Roberto.
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New to Asterisk? Start here:
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asterisk-users mailing list
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