Displaying 8 results from an estimated 8 matches for "gasparimsantos".
2020 Sep 22
3
Asterisk Drop call
...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 no...
2020 Aug 18
2
Queue don't call Interface PJSIP
...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 th...
2020 Sep 21
2
Asterisk Drop call
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
2020 Aug 17
2
Queue don't call Interface PJSIP
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:
2020 Sep 22
0
Asterisk Drop call
...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 i...
2020 Aug 17
0
Queue don't call Interface PJSIP
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 c...
2020 Aug 18
0
Queue don't call Interface PJSIP
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
> i...
2020 Sep 21
0
Asterisk Drop call
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 change...