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[asterisk-users] Asterisk Drop call


 
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roberto.medola at gasp...
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PostPosted: Mon Sep 21, 2020 2:23 pm    Post subject: [asterisk-users] Asterisk Drop call Reply with 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.


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https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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roberto.medola at gasp...
Guest





PostPosted: Tue Sep 22, 2020 11:35 am    Post subject: [asterisk-users] Asterisk Drop call Reply with 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


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lmoreira at dxbrasil.net
Guest





PostPosted: Tue Sep 22, 2020 12:43 pm    Post subject: [asterisk-users] Asterisk Drop call Reply with 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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roberto.medola at gasp...
Guest





PostPosted: Wed Sep 23, 2020 11:04 am    Post subject: [asterisk-users] Asterisk Drop call Reply with quote

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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