No Internet connection, not even internal calls allowed...is this normal??

[dghundt]

In Webmin | Servers | BIND DNS Server

I see three options to set up. I assume I need to pick the (2) internet server option as opposed to just (1) internal name server?


Update,

Thanks for the tip Joe (and everyone elses' help)

I set up BIND with Webmin, chosing option (2) Internet Server.
Did not do anything else.
I then turned off my dsl modem. Waited until the sip trunk with a fqdn (not a numeric ip address) failed to register. Waited a bit longer. I was still able to use an xlite sip extension to call into asterisk for the time, and was still able to dial out xlite to a zap trunk.
Seems to work well.
If someone else can confirm this, we should make it a sticky. This is a very useful tip.

 

[BenS]

see my reply here:
http://forums.whirlpool.net.au/forum-replies.cfm?t=960451&p=-1#bottom

 

[wardmundy]

Score One for the Down Under crew!

1. Create an extension that you're not going to use with voicemail (let's call it 888 - Net Down).
2. Create a new SIP trunk with the following details:

Outgoing Settings
Trunk Name 0-SipFail
host=127.0.0.1
qualify=yes
secret=make1up
type=peer
username=888

Incoming Settings
User Context SipFailincoming
context=from-trunk
secret=make1up
type=user

Registration String
888:[email protected]

Now go to Tools, Asterisk Info, and SIP Info and be sure your extension is registered. Unplug your router and continue to make internal calls.

Once you're sure it's working, you can take out the Registration String entry to eliminate the "Got SIP response 489 "Bad event" back from 127.0.0.1" errors on your CLI.

 

[dghundt]

Looks like a sip reload while internet is down will cause a problem. What would the expected behavior be if the pbxiaf were rebooted while internet was down?

 

[wardmundy]

According to the Australian folks that have been wrestling with this, rebooting your server or reloading SIP while the net is down would not be a good idea. Unless I really screw something up, we almost never reboot our servers so... at least with PBX in a Flash, this shouldn't be a big problem.

 

[dad311]

1. Create an extension that you're not going to use with voicemail (let's call it 888 - Net Down).
2. Create a new SIP trunk with the following details:

Outgoing Settings
Trunk Name 0-SipFail
host=127.0.0.1
qualify=yes
secret=make1up
type=peer
username=888

Incoming Settings
User Context SipFailincoming
context=from-trunk
secret=make1up
type=user

Registration String
888:[email protected]

Now go to Tools, Asterisk Info, and SIP Info and be sure your extension is registered. Unplug your router and continue to make internal calls.

Once you're sure it's working, you can take out the Registration String entry to eliminate the "Got SIP response 489 "Bad event" back from 127.0.0.1" errors on your CLI.

I didn't have much luck with this fix. I was able to make 1 or 2 calls before my Aastra phone would get "no service" on the LCD screen and calls were rolling to VM. Shortly after that I couldnt make any calls.

Anyone else test this fix?

 

[dad311]

Has anyone had success with this fix?

 

[gbrook]

I have a problematic wireless internet link which goes down 3 or 4 times a day and this worked for me:-

1. Install cache only name server on the Asterisk server (BIND with caching configuration)
yum -y install caching-nameserver
chkconfig --add named
chkconfig --level 345 named on
echo "* IN A 127.0.0.2" >> /var/named/localdomain.zone
service named start

2. Set Asterisk as its own DNS server by adding 127.0.0.1 on the first line of /etc/resolve.conf, like
127.0.0.1
your.other.dns.servers

3. Make sure Asterisk has default settings for file /etc/sysconfig/network
NETWORKING=yes
HOSTNAME=your_asterisk_server_name.localdomain
(Do not change HOSTNAME to your_asterisk_server_name.yourdomain.com )

4. Then in file /etc/hosts, something like
127.0.0.1 your_asterisk_server_name.localdomain your_asterisk_server_name localhost.localdomain localhost

5. Reboot (try without the internet connected)

 

[dad311]

I have a problematic wireless internet link which goes down 3 or 4 times a day and this worked for me:-

1. Install cache only name server on the Asterisk server (BIND with caching configuration)
yum -y install caching-nameserver
chkconfig --add named
chkconfig --level 345 named on
echo "* IN A 127.0.0.2" >> /var/named/localdomain.zone
service named start

2. Set Asterisk as its own DNS server by adding 127.0.0.1 on the first line of /etc/resolve.conf, like
127.0.0.1
your.other.dns.servers

3. Make sure Asterisk has default settings for file /etc/sysconfig/network
NETWORKING=yes
HOSTNAME=your_asterisk_server_name.localdomain
(Do not change HOSTNAME to your_asterisk_server_name.yourdomain.com )

4. Then in file /etc/hosts, something like
127.0.0.1 your_asterisk_server_name.localdomain your_asterisk_server_name localhost.localdomain localhost

5. Reboot (try without the internet connected)

Thx, but I still must be missing something, calls still fail when the modem is disconnected.

 

[Stilllearning]

According to the Australian folks that have been wrestling with this, rebooting your server or reloading SIP while the net is down would not be a good idea. Unless I really screw something up, we almost never reboot our servers so... at least with PBX in a Flash, this shouldn't be a big problem.

Ward,

I have tried what you have indicated but it does not work? Now if you have anything in sip_nat.conf this also throws a spanner in the works as well.

I made a detailed post on how to use a SPA3000 as an alternative, but it looks like the post has been removed or it has gone missing as I cannot find it try as I may. This was the only solution I found worked to date, even if you rebooted or started asterisk.

 

[dghundt]

Not again...

I tried my BIND trick in webmin for my pbxiaf 1.3 orgasmatron II with the dell sc440 and my sip extensions all go down when I lose internet connection. This worked well for me with an older AAH build that I have been upgrading over the years.

Do I need to open a port in my pbxiaf software firewall to allow bind to work? This is the only thing I can think of that would cause the difference I am seeing. I (we) need our sip extensions to ring if the internet is down (zap lines -> sip phone). Thank you!

 

[jroper]

Hi

My understanding is that this is only a problem when you have a sip trunk.

A drastic but simple solution is to drop your sip service provider and replace with an IAX service provider.

Joe

 

[dghundt]

My understanding is that this is only a problem when you have a sip trunk.

yes, this is true. A bug in Asterisk that I think would be a high priority fix, but persists.

A drastic but simple solution is to drop your sip service provider and replace with an IAX service provider.

With the better providers like voicepulse dropping iax support, my list of iax providers grows thinner. I did use this option a while back. ( I am using vitelity and pbxiaf as well).

Do you know if BIND needs a port opened in the iptables firewall? I have found posts that suggest I need to open port 53 for udp and tcp.
BIND fixed this issue for me on an older AAH system (that did not have a firewall).

 

[jroper]

Hi

Logic tells me that no ports should be open, as bind is initiating the request, and the fw rules allow anything out, and only a few things in.

Joe

 

[dghundt]

Thanks.

I may not have installed BIND properly, so still working on this part. However, this did not work for Elastix guys&gals like it did for me.
Temporarily turning off iptables had no effect.
Using the sip trunk via Ward's example had no effect.
I'll try the sipura 'dongle' approach.
After that, guess I'll have to go pure IAX2 again. Shame Voicepulse dropping IAX2.

Does anyone have a reliable fix for this?
We need our pstn lines connecting with our sip extensions even if the internet goes down.

 

[dghundt]

Forgot about seeing this script a while back

Elegant script by middleton at voipuser-
http://www.voipuser.org/forum_topic_5452.html

To work with pbxiaf, I modified a couple things from the original code. I attached the modified (untested) file and will test it out. Would like to see others try too.

I basically had to change the name of the sip registration file, and deleted iax changes since this is not an issue. Make sure in pingtest that you enter a valid website name to ping before you start experimenting.

It seems like a best fit solution aside from actually fixing the asterisk source code:
Allows us to still use sip trunks.
Solution contained within the asterisk box (no sipura dongle).
Minimal downtime using cron job.

I expect this will work and will then post more detailed instructions for installing.

 

[hkgonra]

This happened to me yesterday. My phones all have static ip's on the same segment of the lan that the pbx resides. Internet went dowsn and it showed all the phones as offline in the pbx.

 

[dghundt]

Well it works, kind of

With the pingtest cron job fix, my sip extensions and pstn lines do work, mostly.

Sip to sip extension calling on my lan has some unpredictability with it (variable delays in ringtimes, unanswered callee phone ringing does not stop promptly after caller phone hangs up, variable ring times for direct dialing from inbound ivr). For all but the smallest office, I'd say everyone would go crazy. At least the phones ring. If anyone is interested in testing it some, I can post my steps.

I then put in the sip trunk provider ip addresses instead of a fqdn. Turned off pingtest job. Unplugged internet. System worked the same as above. So there is no point in adding fixed ip addresses since functionality is the same and now I would have to keep track of my voip provider ip's.

I tried setting up the sipura dongle approach, but got tired trying to connect to the web interface. Some other day. I expect that I will still get the odd sip extension behavior.

Think, I'll go pure iax like I did a few years ago..... Hey maybe digium could just fix chan sip? Has the professional product been broken for years with the same bug? If so, wowzers.

 

[wardmundy]

Has anyone tried running your own DNS server on the PIAF box? Seems like that might fix it as well.

 

[dghundt]

From what I have read, attempts at this did not work.
I did have BIND running on my pbx (thanks Joe), adding 127.0.0.1 to the list of name servers.