SAIL Community Supported PBX

This issue was fixed in sail_6.2.0-22.  It is not yet on the distro library but you can fetch and install it manually from Github here

https://github.com/aelintra/sail6

Hi,

Would you mind running the install again please?   We'd missed all of the module combinations.  Apologies.  I thin it should work for you now.

Yes, it is available now and the install instructions can be found here

http://sailpbx.com/mediawiki/index.php/SARK_V6.2.0_install

Before you do this install you should back up your existing config and download it to your desktop before you proceed.   Better yet, install it as a new side-by-side image and apply a backup from your existing system.

In version 6 systems you may encounter the above message when adding a new extension.   The cause is a URL change in the location of the wireshark public mainufacturers' database.   We are working up a fix in source but it will take a few days.  In the meantime here is a module fix you can apply directly to your system.

The module which fetches the manufacturers database is

/opt/sark/scripts/getmaclist.sh

Replace the module with this one

#!/bin/sh
# get a list of macs amd manufacturers
#

[ ! -e /opt/sark/www/sark-common/manuf.txt ]  && touch /opt/sark/www/sark-common/manuf.txt
chown www-data:www-data /opt/sark/www/sark-common/manuf.txt
curl -L -s "https://gitlab.com/wireshark/wireshark/-/raw/master/manuf" > /tmp/manuf.txt
if [ -s /tmp/manuf.txt ]; then
        diff /opt/sark/www/sark-common/manuf.txt /tmp/manuf.txt
        if [  "$?" -ne "0" ] ; then
                mv /tmp/manuf.txt /opt/sark/www/sark-common/manuf.txt
                logger SARKgetmaclist - updated manufacturer MAC DB
        else
                logger SARKgetmaclist - manufacturer MAC DB up to date
        fi
else
        logger SARKgetmaclist - **** Could not fetch new manufacturer MAC DB ****
fi

This module gets run every Sunday at 00:00 hours so it will self heal if you leave it to its own devices.  However, you can prime it by simply running the module.

sudo sh /opt/sark/scripts/getmaclist.sh
     

That's unfortunate.   Asterisk can only listen on a single port for SIP.  Look here for a full explanation

https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample

From 6.2 you can set the SIP listen port in the Net->IP Settings panel and SARK will switch it for you.  However, there are consequences to doing that.  Most, if not ALL, SIP carriers only listen on 5060 so you'll need to deal with that if you want to talk to one.   You can do that by setting port=5060 in the trunk definition for the connection.   However, Asterisk will still not listen on 5060 so it will need to be a trunk you register with in order for it to work.   Moreover, you will need to set all of your downstream phones to register with Asterisk using 7070.   This varies from manufacturer to manufacturer so you'll need to read up on it.

Finally, if your remote SARK is simply communicating with another Asterisk then you might consider using IAX to do the trunking, thus sidestepping the issue altogether.   

Very naughty of the modem/router to be using a well-known port for some other purpose.     

 

It can be a few things. 

Is this during an upgrade?

Did you port a previous database backup?   

The BLF template setting has been exposed in 6.2.0-18.   The 6.2 installation procedure has changed and I will post a write-up in the main wiki showing how to install it.

Good question,  it was simply never exposed in 6, I'm not sure why but we can look at doing it for you in a near point release.

Thanks for your input.

SARK 6.2 beta is up in the repos so you can install it if you wish.   There are quite a few new features and I'll cover some of them below.

Support for ARM 64 bit
There is an image available for the HPE on arm64.  This allows 6.2 to run with the new AWS EC2 Graviton enterprise ARM instances (and any other arm64 cpu running in native 64 bit mode). 

Support for Ubuntu netplan
As you probably know, Ubuntu went a bit off piste with their networking and introduced Netplan (I think in 16.04). To be fair, it made their lives a lot easier when dealing with cloud deployments but it caused us all a bit of a problem when defining networks.  Anyhow, 6.2 runs natively with Netplan.  Furthermore, 6.2 will always run with two IP addresses on the prime NIC, one dynamic and one static.   It will always grab a DHCP address if one is available and, in addition,  it will use a second user-defined static IP address.   The static address is specified in the networks section and it is used for SIP operations.  The benefit of this dual strategy is that you can more easily move the PBX between networks without tears.  Whatever network it finds itself in, it will always attempt to grab a DHCP address so you won't be locked out if you forgot to change the static IP.

The only exception to this is if the instance is running under AWS, in which case it will simply use the given address it gets from AWS.   In this case, you won't be given the option of adding another address.

HTTPS Certificates
You can load and install your web certificates through the browser app

HTTPS provisioning with certificate checking
You can load your phone manufacturers' certificates and have the provisioning engine validate provisioning requests against them.   

Non standard SIP ports for extensions
You can provision a non standard (non 5060) SIP port for your phones.

Provisioning support for Fanvil phones
Still a bit of a work in progress but it will get a basic config Fanvil up and running

Upgraded wallboard
Re-written with more detailed information on live call flows

   

Hi there

Under normal circumstances, localnet gets set every time the box boots.   There is a systemd task called debsetlan which calls a routine in the generator to set the correct values.   Once done it exits and isn't active again until the next restart.  You can find the code at /opt/sark/generator/setip.php

I'm puzzled as to why it hasn't worked correctly for you.  First place I would probably look is at permissions and ownership in /etc/asterisk.   Everything should be set to 664 asterisk:asterisk. 

Setting it yourself is fine so you did the right thing to get it chugging.

   

Thanks for the feedback.
We'll spin up -62 and take a look.

It's not much different to -60, changelog shows a couple of changes, none of them major.  In any event, you needn't start again.  -62 will go onto the repo shortly as the final 5.0.0 release so you will be able to use apt-get to update it.   

sail (5.0.0-62) UNRELEASED; urgency=low
  * Tagged as 5.0.0-62
  - fixed typo in *61* greeting playback

 -- CoCoSoft <admin@aelintra.com>  Thu, 7  Mar 2019 18:00:00 +0000

sail (5.0.0-61) UNRELEASED; urgency=low
  * Tagged as 5.0.0-61
  - added user recovery (S200) for the techs
  - added more Yealink phone types

 -- CoCoSoft <admin@aelintra.com>  Fri, 5  Mar 2019 18:00:00 +0000


I'm not sure why that should be but it could be a finishing error on our part.   Can you try flashing -60 instead?  It's also on the library.  We'll see if we can duplicate your results here using -62.



   

ssh default login

UID root
PWD sarkadmin

Web interface is HTTPS

Web default login

UID admin
PWD sarkadmin

Ensure the S200 is in the same subnet as wherever you are logging in from.   SARK firewall is initially set to accept ports 80/443/22 only from the local network (127.0.0.0/24)

Yes, there is an image on the package library in the wiki at sailpbx.com

Download it and use imagecopy or DD to flash your card.  The image requires a 4Gb flash card.

If you can do a regular internal call to another extension and it flows audio OK then the issue is outside of the LAN.    If you've checked and set all the values I gave you then you can run tcpdump to create a PCAP file and make a call.    Then you can see where the RTP is going by using wireshark to analyse the PCAP file. Something like this will do



tcpdump -i {name of your NIC} -s0 -w dumpall.pcap



Download the PCAP and show it to wireshark.