Re: [asterisk-users] asterisk 18.17.1 unreachable

2023-05-11 Thread Antony Stone
On Thursday 11 May 2023 at 21:15:50, Jerry Geis wrote:

> I have 4 devices that I connect here local and there is no issue.
> I have those same 4 devices connecting from another location across the
> internet.
> 
> They all boot up, connect and register I can send audio to them and they
> play.
> - then at times they show UNREACHABLE and I can no longer send audio.
> then they come back online again and are OK.
> 
> I'm using old chan_sip,  I tried changing qualify to no - that did not
> help.
> 
> What might I adjust to keep these SIP units alway ON ?

1. qualify makes no difference - that's just a question of whether Asterisk 
checks whether things are still connected or not.  It doesn't make them 
connect (or not) any differently.

2. check the routers that these devices are connected through and see if you 
can increase the UDP connection tracking timeout.  If a device (phone) doesn't 
send a packet within this time, the router will forget the NAT association 
between the (private IP-addressed) phone and Asterisk (out on a public IP 
address), and that means that when Asterisk sends an Invite to the phone, the 
router fails to send it on to the phone, so the phone doesn't know about it.  
Only when the phone then re-registers, does the router then refresh its 
connection tracking timeout, and all is well again (for a while).

3. Reduce the re-register interval on the phones, so that they refresh the 
connection tracking timeout on the router more frequently than it forgets.


Antony.

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[asterisk-users] asterisk 18.17.1 unreachable

2023-05-11 Thread Jerry Geis
I have 4 devices that I connect here local and there is no issue.
I have those same 4 devices connecting from another location across the
internet.

They all boot up, connect and register I can send audio to them and they
play.
- then at times they show UNREACHABLE and I can no longer send audio.
then they come back online again and are OK.

I'm using old chan_sip,  I tried changing qualify to no - that did not
help.

What might I adjust to keep these SIP units alway ON ?

Thanks


Jerry
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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

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