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...t; 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 n...

...and opening the file only the following appears: [2020-08-18 08:46:47.778] Asterisk 17.6.0 built by root @ asterisk-homolog on a x86_64 running Linux on 2020-08-13 22:40:11 UTC\ Em 17/08/2020 18:57, Joshua C. Colp escreveu: > On Mon, Aug 17, 2020 at 6:16 PM Roberto > <roberto.medola at gasparimsantos.com.br > <mailto:roberto.medola at gasparimsantos.com.br>> wrote: > > Hello. > > > I am having a lot of problems with SIP through NAT. So, I decided > to adopt PJSIP. However, I am not able to make the extensions ring > when receiving a call from t...

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

Hello. I am having a lot of problems with SIP through NAT. So, I decided to adopt PJSIP. However, I am not able to make the extensions ring when receiving a call from the queue. I'm using telnet to include the extension and on the asterisk console, it even shows Called PJSIP/6001, but the extension doesn't ring. If I call from extension to extension, it works normally. telenet:

...isable 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...

On Mon, Aug 17, 2020 at 6:16 PM Roberto < roberto.medola at gasparimsantos.com.br> wrote: > Hello. > > > I am having a lot of problems with SIP through NAT. So, I decided to adopt > PJSIP. However, I am not able to make the extensions ring when receiving a > call from the queue. I'm using telnet to include the extension and on the > asterisk...

On Tue, Aug 18, 2020 at 9:00 AM Roberto < roberto.medola at gasparimsantos.com.br> wrote: > Hi Joshua, thanks for answer. > In this particular test my extension is on a simple network. There is no > NAT, just an asterisk running on a virtual machine on a 7- 64bit CentOs. I > am simulating an environment to be able to use PJSIP on my client. And even >...

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 chang...