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63 posts

Master Geek


# 181381 13-Oct-2015 15:30
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I have noticed that on those occasions when my internet is down (either due to a power outage, or something wrong with our internal settings, or whatever), my Asterisk system can be up and running just fine (according to a check to the Asterisk GUI), but calls to my number will not ring through. The only way to get them to come through again is to use my 2Talk system to make an outgoing call, which seems to "kickstart" the system back into gear. Is this a feature, or a fault? If the former, then that kinda sucks, but if it's the latter, what can I do to remedy it?

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219 posts

Master Geek


  # 1405085 13-Oct-2015 15:33
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Triggering a registration should do it.

If you have a script that is able to determine when you are back up again then it could tell Asterisk to re-register by issuing:

asterisk -rx 'sip reload'

I'm pretty sure that causes a new registration to go out. You can check the registration status with:

asterisk -rx 'sip show registry'

Pete




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Master Geek


  # 1405103 13-Oct-2015 16:10
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That would be great; where would the script go, in my Asterisk PBX or somewhere else?

 
 
 
 


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  # 1405109 13-Oct-2015 16:18
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It's not a fault and should never happen.

What version of Asterisk are you using? It sounds very much like the infamous Asterisk DNS 'bug' assuming you've configured your box with a FQDN for the SIP proxy and are using an external DNS. If so it's just a bad setup causing the issue.






63 posts

Master Geek


  # 1405134 13-Oct-2015 17:14
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I'm on v.11.4.0, and I have to confess I have never become terribly expert at it so managing it or messing with the code is something I am very wary of doing. I have also therefore never upgraded the components, or do so only under great duress.

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  # 1405153 13-Oct-2015 18:19
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It's not your version or lack of updates, it's just that you don't have things configured correctly. It's probably one of the most common issues affecting people new to Asterisk who follow setup guides but don't actually understand what's happening!

Use of a FQDN for the sip peer IP is not recommended. If you are going to use a FQDN then your Asterisk box needs to be running a local DNS proxy or be pointing to a local DNS server. There are huge numbers of posts covering this online if you search for it.






63 posts

Master Geek


  # 1408566 18-Oct-2015 13:01
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The fellow who set up my Asterisk system came back to me (having seen my post on here) saying that the error was his, and that I should change my settings in Asterisk to have the SIP_Registration go to an IP address rather than to a URL. I have done that, and also done the same in the SIP_Peering setup in the trunks for my installation, but the system still deregisters after a while (it is not, as I had previously thought, tied to an internet outage, and instead may go some way toward explaining why it seems that I get so few phone calls ever since this Asterisk system was installed). 

So, if the error was having the URL instead of an IP address, but the problem is still happening (I have not figured out how long has to elapse before the system once again requires a kickstart), what else could be causing this trouble? Is there anyone out there who would like to raise their hand to offer their services to be my VoiP management person?

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Master Geek


  # 1408575 18-Oct-2015 14:00
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SIP_Peering and registration are mutually exclusive. Registration is not needed if you configure peering/trunking (however firewall configuration is!). Mixing the two may be part of the cause of your problems here.  You should use one or the other but not both. Do not choose peering/trunking if your public IP changes occasionally; use registration instead (not as well as).

My best guess here is that inbound calls made shortly after a new registration event are getting through due to the inbound UDP port on the firewall still being open. However the NAT timeout occurs before the registration timeout period and there is a portion of time afterwards during which the inbound UDP can't get in through the firewall again. If calls come in via the trunk (or registration) during that period they will fail due to lack of the INVITE packet getting through to your PBX.

I recommend change your system configuration to either a trunk system with associated firewall rules in place, or a registration ONLY system with a low registration timeout period and keepalives enabled. Alternatively if possible for you to do, moving your PBX to a (non-firewalled) public IP address would probably mask these issues too.


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