Telium Support Group

The license file contains a ‘signature’ to verify the license has not been changed. You are inadvertently changing the license file; there are subtle differences between Linux text files and Windows text files. (In particular, Windows uses CRLF while Linux uses CR only.) You don’t need to know the details, but even opening the file in Notepad and then saving it again is changing the file!

This time go to your mail client and save the attachment directly from the message (without opening in Notepad) to the PBX. Your license should then work fine.

Exit code 158 means that HAAst is unable to find the service/executable file needed to control Asterisk. Since you installed Asterisk from source, I assume you are not using any configuration generator (eg: Elastix) as well. If that’s the case then the ‘distribution’ setting in the [asterisk] stanza of the haast.conf file should be set to 2 (Digium Asterisk). Confirm that setting – then proceed to the steps below if the error remains.

Next ensure that you installed an Asterisk service file appropriate for your Linux distribution. Most recent Linux distributions have switched to systemd, which requires you use an asterisk.service file. If you are using an older Linux distribution then you require a SysV / initd style service file.

While Asterisk includes a sample SysV style service file for Asterisk, it might not include a systemd style service file just yet (as of Dec 2016). You can find one by searching the Digium support forums, but here’s an example you can use:


[Unit]
Description=Asterisk PBX and telephony daemon
Documentation=man:asterisk(8)
Wants=network.target
After=network.target
 
[Service]
Type=simple
User=asterisk
Group=asterisk
ExecStart=/usr/bin/asterisk -f -C /etc/asterisk/asterisk.conf
ExecStop=/usr/bin/asterisk -rx ‘core stop now’
ExecReload=/usr/bin/asterisk -rx ‘core reload’
 
[Install]
WantedBy=multi-user.target


Save the above as ‘asterisk.service’ and place it in /etc/systemd/system/ and set file permissions to 755. Don’t add restart directives (like you might find on the internet) since HAAst should be responsible for restarting Asterisk when necessary. You will also have to tell Linux about the new service file with the ‘systemctl daemon-reload’ command (or similar depending on your Linux distro).

If you want to use the SysV / initd style service file that’s ok too, but you’ll need to tell HAAst which style of service file you are using. To do so contact Telium support and we will provide you with details specific to your Linux distribution. (But in general if your Linux distro uses systemd, stick with systemd service files).

The problem is that you have a space character after the backslash on line 5 (based on analyzing the config file sent). If you add any character after a backslash it means ‘escape’ as opposed to ‘continuation’. So SecAst stops reading your excludes at line 5.

The solution is to ensure you have no characters following the backslash character.

Yes. No problem!

Many call centers are keeping their primary PBX’s on site (for performance and other reasons), and using the cloud for their backup PBX’s. HAAst will run on Amazon Web Services (AWS) + Amazon Elastic Cloud 2 (EC2), Microsoft Azure, and more, to create your primary PBX, backup PBX, or both. You can even place the HAAst operational database in Amazon Relational Data Services (RDS) or equivalent with both peers writing performance and operating data to the shared database. (The HAAst web GUI allows you to view combined and individual PBX reports in many areas). We do not recommend placing Asterisk configuration databases in RDS or equivalent – but this should not be a problem as they are fairly static and small.

Unlike other open source and closed source commercial HA products, HAAst does not use any LAN based protocols (NFS/DRBD/hearbeat/etc). HAAst is designed around only WAN based protocols, primarily it’s own PeerLink protocol. HAAst can accommodate large swings in network latency as would be seen in links between in-house servers and the cloud. HAAst (Commercial Unlimited edition) even learns these variations and adapts to the changes, so you never have a false positive fail-over of the cluster. This approach allows customers to even place their PBX instances in different Amazon Recovery Zone’s, and in different Regions.

With AWS, or any cloud provider, you will want to use their static network hardware option; Amazon calls it Elastic Network Interfaces (ENI). As of August 2016 Azure (ARM and ASM) support static network hardware (MAC address).

Telium’s call continuity feature (available in HAAst OEM edition) maintains all SIP/RTP traffic data in the event of a failover. When HAAst transitions from one cluster node to another, HAAst reconstructs the calls in/out of Asterisk exposing identical IP/port numbers to the outside world. The allows calls to resume without any user agents realizing a cluster failover has taken place. HAAst can also initiate events with Asterisk to resume call recordings, log data, etc. ensuring business requirements can be satisfied as well.

If you cannot use Telium’s call continuity capability you can still create a simplistic workaround. First of all it’s important to understand the limitations of the SIP protocol (regardless of HA product), and then design around those limitations. The SIP protocol does not allow for the midpoints (e.g. PBX) or endpoints (e.g. phone) to be seized without the midpoints/endpoints participating in SIP negotiation. This means that failure of a midpoint or endpoint results in failure of the call(s) since SIP communications cease. A future version of SIP may support seizing channels, but the IETF has not indicated any such upcoming functionality.

We know of at least one product on the market that claims a proprietary / ‘patented’ way to keep SIP calls up, but all they do is introduce a SIP proxy in front of the PBX which tries to re-establish the missing leg of a dropped call. These products don’t have much commercial success because they introduce a new single point of failure in front of the cluster (so if the proxy dies the entire cluster dies). In the case of power outage or network outage these proxy solutions do nothing and all calls drop. Telium has worked with vendors to design solutions for emergency call centers, hospitals, large commercial call centers, etc. and they all frown on placing a single point of failure in front of the cluster. An on-PBX solution (as offered by Telium) is the preferred solution.

Even large proprietary PBX vendors don’t support salvaging SIP calls in case of a complete PBX failure. They may offer “HA” options which transition calls in progress within or between PBX’s so long as the core of the PBX is still functional (so for example the PBX could withstand a media processor failure). However, they cannot salvage SIP calls in the event of a complete and immediate disconnect of the PBX (e.g. power failure, data center network failure, etc).

If you really want to try to salvage calls in progress (sometimes called ‘call continuity’ or ‘call survival’) with a do-it-yourself solution the best way to do so is at the ITSP/trunk provider. You can request that your carrier redirect any calls in progress which terminate without a SIP BYE command be redirected to a backup number/trunk. If your ITSP/Carrier will not do so, you can create your own SIP proxy but it should be collocated at your ITSP/carrier’s POP. Placing such a proxy on your premises is worthless (your cluster will appear to perform well in simplistic tests, but fail in real world outage scenarios), and is no different than the solutions mentioned above. You can try to use direct media with user agents (phones) that don’t drop a call if the SIP connection is unresponsive (eg; to registration) but this depends heavily on details of your implementation.

We would like to emphasize that Telium discourages adding a black box device or proxy in front of a cluster. However, if you still want to go down the DIY path you can create an open source version of the above (not recommended) product as described here: https://telium.io/topic/patented-call-survival-add-on/ or you could design a solution which attempts to keep RTP up without SIP (we have a few posts on that topic).

• This reply was modified 4 years, 8 months ago by WebMaster.
• This reply was modified 4 years, 8 months ago by WebMaster.
• This reply was modified 4 years, 8 months ago by WebMaster.

As of Dec 11th 2016 we now officially support Debian 8 LTS. Look for the Debian option on the download page.

Please note that we will create custom builds of our products for any major distribution / architecture for users of the commercial editions. Custom builds are created at no charge (for major platforms) and for-fee (for rare/unusual platforms).

If you’ve never used Linux before, then installing any third party product onto Linux can be a challenge. Creating a Linux system from a CD/ISO is easy: click next>next>finish. So unfortunately that might artificially raise your confidence. Linux is not the most friendly system (compared to Windows).

Although our products target large commercial installations, we realize that a some of these organizations experiment with Asterisk using one of the free configuration generators (like Issabel(TM), FreePBX(TM), Elastix(TM), PIAF(TM), etc). For the convenience of these users we can create a virtual machine with Linux & FreePBX & Asterisk & HAAst & SecAst preinstalled. This should help Windows admins get up and running quickly.

If you want to install our products on an existing Linux PBX, then we’re here to help. We include support with all product sales – but I suspect you will exhaust the included support incidents quickly. I would recommend you purchase support assistance along with your software purchase. For a straight forward setup Telium can perform the entire installation and configuration with/for you in 4 hours, assuming this is a very simple cluster design. (But not setup your users/devices/dialplan etc in Asterisk).

We have also helped customers create prebuilt software images to their specifications. (This is useful if you plan to deploy PBX’s to numerous sites.) We can create bare metal images in Ghost, dd, and G4L formats to suit your needs, which you can use to deploy new systems at your convenience.