<div style="font-family: Arial, sans-serif; font-size: 14px; color: rgb(0, 0, 0); background-color: rgb(255, 255, 255);">Hello,</div><div style="font-family: Arial, sans-serif; font-size: 14px; color: rgb(0, 0, 0); background-color: rgb(255, 255, 255);"><br></div><div style="font-family: Arial, sans-serif; font-size: 14px; color: rgb(0, 0, 0); background-color: rgb(255, 255, 255);">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.</div><div style="font-family: Arial, sans-serif; font-size: 14px; color: rgb(0, 0, 0); background-color: rgb(255, 255, 255);"><br></div><div style="font-family: Arial, sans-serif; font-size: 14px; color: rgb(0, 0, 0); background-color: rgb(255, 255, 255);">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.</div><div style="font-family: Arial, sans-serif; font-size: 14px; color: rgb(0, 0, 0); background-color: rgb(255, 255, 255);"><br></div><div style="font-family: Arial, sans-serif; font-size: 14px; color: rgb(0, 0, 0); background-color: rgb(255, 255, 255);">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.<br></div><div style="font-family: Arial, sans-serif; font-size: 14px; color: rgb(0, 0, 0); background-color: rgb(255, 255, 255);"><br></div><div style="font-family: Arial, sans-serif; font-size: 14px; color: rgb(0, 0, 0); background-color: rgb(255, 255, 255);">Marek</div><div style="font-family: Arial, sans-serif; font-size: 14px; color: rgb(0, 0, 0); background-color: rgb(255, 255, 255);"><br></div><div style="font-family: Arial, sans-serif; font-size: 14px; color: rgb(0, 0, 0); background-color: rgb(255, 255, 255);"><br></div><div class="protonmail_quote">
On Wednesday, November 8th, 2023 at 15:53, John Harragin <jharragin@mw.k12.ny.us> wrote:<br><br>
<blockquote class="protonmail_quote" type="cite">
<div dir="ltr"><div>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. </div><div><br></div><div>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).</div></div><br><div class="gmail_quote"><div class="gmail_attr" dir="ltr">On Wed, Nov 8, 2023 at 8:46 AM Marek Greško <<a href="mailto:marek.gresko@protonmail.com" rel="noreferrer nofollow noopener" target="_blank">marek.gresko@protonmail.com</a>> wrote:<br></div><blockquote style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote">Hello,<br>
<br>
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.<br>
<br>
Marek<br>
<br>
<br>
<br>
<br>
<br>
Sent with Proton Mail secure email.<br>
<br>
------- Original Message -------<br>
On Wednesday, November 8th, 2023 at 1:21, John Harragin <<a target="_blank" href="mailto:jharragin@mw.k12.ny.us" rel="noreferrer nofollow noopener">jharragin@mw.k12.ny.us</a>> wrote:<br>
<br>
<br>
> Marek,<br>
> <br>
> See if calls hang in the system if you encounter another outage<br>
> core show channels<br>
> <br>
> ...if so,<br>
> core set verbose 3<br>
> and see what instructions subsequent calls hang on.<br>
> <br>
> <br>
> <br>
> On Mon, Nov 6, 2023 at 4:44 PM Marek Greško <a target="_blank" href="mailto:marek.gresko@protonmail.com" rel="noreferrer nofollow noopener">marek.gresko@protonmail.com</a> wrote:<br>
> <br>
> > Hello,<br>
> > <br>
> > 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.<br>
> > <br>
> > When looking into logs I see only res_pjsip_outbound_registration.c: No response<br>
> > received from sip provider. Nothing else.<br>
> > <br>
> > In phone log I see:<br>
> > CC_eventProc(event=63(CC_EV_SIG_REGISTER_FAILED),<br>
> > lid=0, par=0, par2=(nil))<br>
> > <br>
> > The phone is Cisco SPA525G2.<br>
> > <br>
> > Thanks.<br>
> > <br>
> > Marek<br>
> > <br>
> > ------- Original Message -------<br>
> > On Monday, November 6th, 2023 at 15:45, Joshua C. Colp <a target="_blank" href="mailto:jcolp@sangoma.com" rel="noreferrer nofollow noopener">jcolp@sangoma.com</a> wrote:<br>
> > <br>
> > On Mon, Nov 6, 2023 at 10:42 AM Marek Greško <a target="_blank" href="mailto:marek.gresko@protonmail.com" rel="noreferrer nofollow noopener">marek.gresko@protonmail.com</a> wrote:<br>
> > <br>
> > > 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?<br>
> > <br>
> > 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?<br>
> > <br>
> > --<br>
> > Joshua C. Colp<br>
> > Asterisk Project Lead<br>
> > Sangoma Technologies<br>
> > Check us out at <a target="_blank" rel="noreferrer nofollow noopener" href="http://www.sangoma.com">www.sangoma.com</a> and <a target="_blank" rel="noreferrer nofollow noopener" href="http://www.asterisk.org">www.asterisk.org</a><br>
> > <br>
> > --<br>
> > _____________________________________________________________________<br>
> > -- Bandwidth and Colocation Provided by <a target="_blank" rel="noreferrer nofollow noopener" href="http://www.api-digital.com">http://www.api-digital.com</a> --<br>
> > <br>
> > Check out the new Asterisk community forum at: <a target="_blank" rel="noreferrer nofollow noopener" href="https://community.asterisk.org/">https://community.asterisk.org/</a><br>
> > <br>
> > New to Asterisk? Start here:<br>
> > <a target="_blank" rel="noreferrer nofollow noopener" href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started">https://wiki.asterisk.org/wiki/display/AST/Getting+Started</a><br>
> > <br>
> > asterisk-users mailing list<br>
> > To UNSUBSCRIBE or update options visit:<br>
> > <a target="_blank" rel="noreferrer nofollow noopener" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
> <br>
> <br>
> --<br>
> _____________________________________________________________________<br>
> -- Bandwidth and Colocation Provided by <a target="_blank" rel="noreferrer nofollow noopener" href="http://www.api-digital.com">http://www.api-digital.com</a> --<br>
> <br>
> Check out the new Asterisk community forum at: <a target="_blank" rel="noreferrer nofollow noopener" href="https://community.asterisk.org/">https://community.asterisk.org/</a><br>
> <br>
> New to Asterisk? Start here:<br>
> <a target="_blank" rel="noreferrer nofollow noopener" href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started">https://wiki.asterisk.org/wiki/display/AST/Getting+Started</a><br>
> <br>
> asterisk-users mailing list<br>
> To UNSUBSCRIBE or update options visit:<br>
> <a target="_blank" rel="noreferrer nofollow noopener" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
<br>
-- <br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a target="_blank" rel="noreferrer nofollow noopener" href="http://www.api-digital.com">http://www.api-digital.com</a> --<br>
<br>
Check out the new Asterisk community forum at: <a target="_blank" rel="noreferrer nofollow noopener" href="https://community.asterisk.org/">https://community.asterisk.org/</a><br>
<br>
New to Asterisk? Start here:<br>
<a target="_blank" rel="noreferrer nofollow noopener" href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started">https://wiki.asterisk.org/wiki/display/AST/Getting+Started</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a target="_blank" rel="noreferrer nofollow noopener" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a></blockquote></div>
</blockquote><br>
</div>