Hello,
I confirm server and phones are on the same subnet and the phones are able to
resolve local domain also when internet connection os down. It seems to be the
asterisk bug I referenced before. There seems to be some bolcking resolver in
it.
I do not use database related to asterisk. This should be related to the srv
record resolving. It seems quite random time to trigger the issue. When
inspecting logs after internet problems started the issue appeared in one hour
and several minutes. After restart of the asterisk it reappeared in less than
half an hour. When trying to reproduce I was not able to reproduce for one hour
and a half. So I decided to configure srv_lookups=no. I hope the issue is
workarounded now.
But I think asterisk should be fixed. It should successfully start when the
VoIP providers sip server is not reachable, should recover after it becomes
available. And should work locally when it stops to be responding. The tweak of
creating /etc/hosts entry for the sip server and disabling srv lookups should
not be needed. I hope sometimes theese issues will be addressed.
Marek
On Wednesday, November 8th, 2023 at 15:53, John Harragin
<jharra...@mw.k12.ny.us> wrote:
> Are the phones and the server in the same subnet? You might making note of
> the IPs and just simply try pinging everything with the uplink disconnected.
> Also, if you are using domain names for registration, it is possible a dns
> server must be reachable.
>
> If you are using database for any of your call processing, an unreachable dns
> server can also be the cause of trouble. For some reason, even if you are
> using IP addressing, Mysql will try to resolve a connection and can hang
> (there is a mysql parameter to not resolve addresses).
>
> On Wed, Nov 8, 2023 at 8:46 AM Marek Greško <marek.gre...@protonmail.com>
> wrote:
>
>> Hello,
>>
>> it did not seem the call hung. It seemed it never started. There was no
>> dialplan execution on the asterisk side. It looked like phones were
>> unregistered. Same shows the log posted previously.
>>
>> Marek
>>
>> Sent with Proton Mail secure email.
>>
>> ------- Original Message -------
>> On Wednesday, November 8th, 2023 at 1:21, John Harragin
>> <jharra...@mw.k12.ny.us> wrote:
>>
>>> Marek,
>>>
>>> See if calls hang in the system if you encounter another outage
>>> core show channels
>>>
>>> ...if so,
>>> core set verbose 3
>>> and see what instructions subsequent calls hang on.
>>>
>>>
>>>
>>> On Mon, Nov 6, 2023 at 4:44 PM Marek Greško marek.gre...@protonmail.com
>>> wrote:
>>>
>>> > Hello,
>>> >
>>> > sure I have local DNS server and public resolving should not be needed
>>> > for phone registrations. Running pjsip show endpojnt show the endpoints
>>> > as not in use.
>>> >
>>> > When looking into logs I see only res_pjsip_outbound_registration.c: No
>>> > response
>>> > received from sip provider. Nothing else.
>>> >
>>> > In phone log I see:
>>> > CC_eventProc(event=63(CC_EV_SIG_REGISTER_FAILED),
>>> > lid=0, par=0, par2=(nil))
>>> >
>>> > The phone is Cisco SPA525G2.
>>> >
>>> > Thanks.
>>> >
>>> > Marek
>>> >
>>> > ------- Original Message -------
>>> > On Monday, November 6th, 2023 at 15:45, Joshua C. Colp jc...@sangoma.com
>>> > wrote:
>>> >
>>> > On Mon, Nov 6, 2023 at 10:42 AM Marek Greško marek.gre...@protonmail.com
>>> > wrote:
>>> >
>>> > > It looks like all phones get unregistered, but I am not aware of the
>>> > > cause. Why are get not registered when there is a connectivity between
>>> > > them and asterisk?
>>> >
>>> > Are the REGISTER requests reaching Asterisk (do they show up in a packet
>>> > capture, do they show up in "pjsip set logger on")? It needs to be
>>> > further isolated. How are the phones configured to reach Asterisk? If
>>> > using a hostname, are they still able to resolve it?
>>> >
>>> > --
>>> > Joshua C. Colp
>>> > Asterisk Project Lead
>>> > Sangoma Technologies
>>> > Check us out at www.sangoma.com and www.asterisk.org
>>> >
>>> > --
>>> > _____________________________________________________________________
>>> > -- 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
>>
>> --
>> _____________________________________________________________________
>> -- 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
--
_____________________________________________________________________
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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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