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.medola at gasparimsantos.com.br > <mailto:roberto.medola at 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. > > > -- > _____________________________________________________________________ > -- 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 > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200922/5082734d/attachment.html>
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 at gasparimsantos.com.br> escreveu:> 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.medola at 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; 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-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200922/47d1cc50/attachment.html>
Thanks Luciano. But there is no active ALG on the modem. Attached the call flow, including the ACK. Em 22/09/2020 14:41, Luciano Moreira escreveu:> 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 at gasparimsantos.com.br > <mailto:roberto.medola at gasparimsantos.com.br>> escreveu: > > 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.medola at gasparimsantos.com.br >> <mailto:roberto.medola at 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. >> >> >> -- >> _____________________________________________________________________ >> -- 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 > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200923/8139ade0/attachment-0001.html> -------------- next part -------------- A non-text attachment was scrubbed... Name: fluxo.png Type: image/png Size: 90361 bytes Desc: not available URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200923/8139ade0/attachment-0001.png>
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:> 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 at gasparimsantos.com.br > <mailto:roberto.medola at gasparimsantos.com.br>> escreveu: > > 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.medola at gasparimsantos.com.br >> <mailto:roberto.medola at 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. >> >> >> -- >> _____________________________________________________________________ >> -- 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 > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200923/2aa68e73/attachment.html>