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roberto.medola at gasp... Guest
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Posted: Mon Sep 21, 2020 2:23 pm Post subject: [asterisk-users] Asterisk Drop call |
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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; 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.
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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roberto.medola at gasp... Guest
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Posted: Tue Sep 22, 2020 11:35 am Post subject: [asterisk-users] Asterisk Drop call |
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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:
Quote: | 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.medola@gasparimsantos.com.br (roberto.medola@gasparimsantos.com.br)> wrote:
Quote: | 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; 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.
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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lmoreira at dxbrasil.net Guest
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Posted: Tue Sep 22, 2020 12:43 pm Post subject: [asterisk-users] Asterisk Drop call |
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Roberto
Check your router if ALG or similar feature is enabled. Disable and test.
Also, on SNGREP check if both parties are getting ACK correctly after RTP starts.
--
Atenciosamente,
Luciano Moreira
(85)99974-2750
__
Logic Telecom
0800-085-7799 | (85)4042-7799 | (11)4210-7799
Em ter., 22 de set. de 2020 às 13:35, Roberto <roberto.medola@gasparimsantos.com.br (roberto.medola@gasparimsantos.com.br)> escreveu:
Quote: | 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:
Quote: | 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.medola@gasparimsantos.com.br (roberto.medola@gasparimsantos.com.br)> wrote:
Quote: | 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; 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.
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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roberto.medola at gasp... Guest
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Posted: Wed Sep 23, 2020 11:04 am Post subject: [asterisk-users] Asterisk Drop call |
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The problem has been detected.
FXS equipment is causing the fall. Most likely from some bad contact.
Thank you all for your help.
Roberto.
Em 22/09/2020 14:41, Luciano Moreira escreveu:
Quote: | Roberto
Check your router if ALG or similar feature is enabled. Disable and test.
Also, on SNGREP check if both parties are getting ACK correctly after RTP starts.
--
Atenciosamente,
Luciano Moreira
(85)99974-2750
__
Logic Telecom
0800-085-7799 | (85)4042-7799 | (11)4210-7799
Em ter., 22 de set. de 2020 às 13:35, Roberto <roberto.medola@gasparimsantos.com.br (roberto.medola@gasparimsantos.com.br)> escreveu:
Quote: | 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:
Quote: | 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.medola@gasparimsantos.com.br (roberto.medola@gasparimsantos.com.br)> wrote:
Quote: | 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; 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.
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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