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For being specific, I am using asterisk with a Heartbeat active/pasive cluster. There are 2 nodes in the cluster. Let's suppose Asterisk1 Asterisk2. Eveything is well configured in my cluster. When one of the nodes looses internet connection, asterisk service fails or the Asterisk1 is turned off, the asterisk service and the failover IP migrate to the surviving node (Asterisk2).

The problem is if we actually were processing a call when the Asterisk1 fell down asterisk stops the call and I can redial until asterisk service is up in asterisk2 (5 seconds, not a bad time).

But, my question is: Is there a way to make asterisk work like skype when it looses connection in a call? I mean, not stopping the call and try to reconnect the call, and reconnect it when asterisk service is up in Asterisk2?

Cœur
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Carlos Daniel
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3 Answers3

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There are some commercial systems that support such behavour.

If you want do it on non-comercial system there are 2 way:

1) Force call back to all phones with autoanswer flag. Requerment: Guru in asterisk.

2) Use xen and memory mapping/mirror system to maintain on other node vps with same memory state(same running asterisk). Requirment: guru in XEN. See for example this: http://adrianotto.com/2009/11/remus-project-full-memory-mirroring/

Sorry, both methods require guru knowledge level.

Note, if you do sip via openvpn tunnel, very likly you not loose calls inside tunnel if internet go down for upto 20 sec. That is not exactly what you asked, but can work.

arheops
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Since there is no accepted answer after almost 2 years I'll provide one: NO. Here's why.

  1. If you failover from one Asterisk server 1 to Asterisk server 2, then Asterisk server 2 has no idea what calls (i.e. endpoint to endpoing) were in progress. (Even if you share a database of called numbers, use asterisk realtime, etc). If asterisk tried to bring up both legs of the call to the same numbers, these might not be the same endpoints of the call.

  2. Another server cannot resume the SIP TCP session of the other server since it closed with the last server.

  3. The MAC source/destination ports may be identical and your firewall will not know you are trying to continue the same session.

etc.....

If you goal is high availability of phone services take a look at the VoIP Info web site. All the rest (network redundancy, disk redundancy, shared block storage devices, router failover protocol, etc) is a distraction...focus instead on early DETECTION of failures across all trunks/routes/devices involved with providing phone service, and then providing the highest degree of recovery without sharing ANY DEVICES. (Too many HA solutions share a disk, channel bank, etc. that create a single point of failure)

TSG
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Your solution would require a shared database that is updated in realtime on both servers. The database would be managed by an event logger that would keep track of all calls in progress; flagged as LINEUP perhaps. In the event a failure was detected, then all calls that were on the failed server would be flagged as DROPPEDCALL. When your fail-over server spins up and takes over -- using heartbeat monitoring or somesuch -- then the first thing it would do is generate a set of call files of all database records flagged as DROPPPEDCALL. These calls can then be conferenced together.

The hardest part about it is the event monitor, ensuring that you don't miss any RING or HANGUP events, potentially leaving a "ghost" call in the system to be erroneously dialed in a recovery operation.

You likely should also have a mechanism to build your Asterisk config on a "management" machine that then pushes changes out to your farm of call-manager AST boxen. That way any node is replaceable with any other.

What you should likely have is 2 DB servers using replication techniques and Linux High-Availability (LHA) (1). Alternately, DNS round-robin or load-balancing with a "public" IP would do well, too. These machine will likely be light enough load to host your configuration manager as well, with the benefit of getting LHA for "free".

Then, at least N+1 AST Boxen for call handling. N is the number of calls you plan on handling per second divided by 300. The "+1" is your fail-over node. Using node-polling, you can then set up a mechanism where the fail-over node adopts the identity of the failed machine by pulling the correct configuration from the config manager.

If hardware is cheap/free, then 1:1 LHA node redundancy is always an option. However, generally speaking, your failure rate for PC hardware and Asterisk software is fairly lower; 3 or 4 "9s" out of the can. So, really, you're trying to get last bit of distance to the "5th 9".

I hope that gives you some ideas about which way to go. Let me know if you have any questions, and please take the time to "accept" which ever answer does what you need.

(1) http://www.linuxjournal.com/content/ahead-pack-pacemaker-high-availability-stack

MichelV69
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