[arm-allstar] Losing connection to SIP provider

[Doug Crompton]

Joe,

 Have you added a registration line in /etc/asterisk/sip.conf like this -

register => xxxxxx:yyyyyyyyy#@washington.voip.ms:5060

Your line after the => probably won't look like this. It will depend on the
provider or what you are connecting to.

Any time you change anything in sip.conf make sure to restart Asterisk -
astres.sh

Then you can go into the Asterisk client from the menu and type -

sip show registry

It should show your registration data and show if you are registered.

Then you can type -

sip show peers

It will give more information about the connection.

In your sip.conf stnaza for your provider or device you can add the line -

qualify=yes

This will monitor the connection and show RTT in the sip show peers command.

See -  https://www.voip-info.org/asterisk-sip-qualify/  for more info.


*73 Doug*

*WA3DSP*

*http://www.crompton.com/hamradio <http://www.crompton.com/hamradio>*








On Sun, May 13, 2018 at 11:30 PM, "KD2NFC, Joe Puma via arm-allstar" <
arm-allstar at hamvoip.org> wrote:

> Hello, I’m a new Allstar, Asterisk user and been working on building a RPi
> node. All is well so far but I setup a sip provider and it all works but
> over time I guess it looses connection and when I try to dial a number it
> fails. I have to reload sip for it to reregister. Is there a keep alive
> option I can set with it monitors the connection and reconnects.
>
> Joe
> Kd2nfc
>
> Sent from my iPad
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