Due to high volume of outbound calls you might be asked to alternate the outbound traffic by your termination provider.

Basically, what what you need to do here is: Having 2 destinations (Dst_A, Dst_B), check/set a variable in the AstDB. Then if the variable is 1, set it to 2 and route via Dst_A; otherwise, set it to 1 and route via Dst_B.

This code snippet might be what  you are looking for:

Maybe in a future you might want to think in another approach to this task, but in the meantime this might come in handy.

We are currently looking for a PHP/MySQL senior developer in the Dominican Republic. We are offering a full contract with excellent salary.

Requirements:

• OOP PHP
• MySQL knowledge (you don’t have to be a ninja but you must have a good level)
• jQuery
• Must be efficient and proactive, oriented to results.
• Must live in Dominican Republic (Santo Domingo)

Desirable:

• HTML/CSS knowledge
• Linux CLI knowledge (this is a great plus)

If you know would like to apply just send me an email (see the ‘About Me‘ section).

Note: This is not a Digium job vacancy neither it’s related with them.

System exposed to the Internet are target for malicious users that try to discover security holes and take advantage of them. Of course, Asterisk servers are not the exception and usually get involved into heavy attacks.

If your Asterisk box is facing the Internet and you are being constantly flooded with break-in attempts, you might want to follow these steps in order to secure your Asterisk Server:

• If a system does not have to connect to your Asterisk box, bock it, either using a hardware firewall or software firewall.
  The point is, “that which is not explicitly allowed is prohibited”.  As Paul Russell said:   “block everything, then open up holes as necessary”. Block every VoIP request by default (e.g SIP REGISTER packages), then open your box to the system that really should connect to your server.Normally, attackers tend to go away if the target is blocking them, otherwise, they could just sit there and keep going and going. They might not succeed but the bandwidth consumption will be a problem for you.
• If you do not need a service running in your Asterisk box, do not enable it, whether or not you have blocked access to it.
• Use Fail2Ban in order to prevent brute force attacks.
• Someone points out that these types of attacks happen on servers that fit a digital signature. You might want to change your signature and change default ports.
• Use a VoIP blacklist project in order to know which rank of IP’s to block.
• Do not just only worry about the bad guy you stopped, you should also worry about the bad guy that might already be inside (a trojan in the M$ Winblows box with the softphone) and that now has credentials that will not be flagged by Fail2Ban.You might want to implement some sort of monitoring/alert system based on outbound call volume, simultaneous initiated calls, out of average call duration, etc.
• I really should not be mentioning it (because it is obvious) but, Ok, here I go: use strong passwords, I mean, really strong passwords.
• And last, never, -repeat-: never, by any means, forget about your ‘default‘ context in your config files (specially in extensions.*). You can empty them if you want, but whatever you do, make sure you don’t leave a security hole there.

These steps will not guarantee you a 100% full-bulletProof Asterisk box, but they will definitively help you out (our your client) in avoiding a $30K phone bill.

The news of Digium IP Phones have got a big number of people giving their opinion on the subject. Some of the inquiries are about the manufacturing process, others even question the very reasons for Digium to launch phones (after all, they are entering to a very tight and busy phone market).

The fact that Digium itself is manufacturing and designing the phones brings good news for us, developers. The Digium’s team was ordered to create the easiest phone to install, integrate, provision and use. After the general availability date of these phones comes (April 2012), we will see if the engineers delivered on that goal.

Advantages:

• The new phones are easier to auto-discover and configure them.
•  The application an the phone now will be more integrated as the phones have been thought in a way that they can access all sorts of user status and system information.
• Custom developer’s application will now be able to run directly in the phone thanks to an application engine that has been included.

In those places with good broadband connection service available, like O2, this opens a new realm of possibilities. Now the tools for a complete VoIP solution have been provided with the introduction of this Astersisk-specifical oriented phones.

Could you even imagine what could a community of more than 80,000 users and developers do now that the maximum flexibility ever thought will be in both sides of the VoIP solution?. Even at home (for those who are hobbyist) you could set your own VoIP solution and adapt it to fit your needs.

This is the beginning of a new era of innovative solutions to rise. If you would like to introduce yourself in the world of VoIP, don’t miss some future articles where I’ll be covering the installation details and “obscure” (not really) steps about how to set your own VoIP solution at home.

The Asterisk Development Team is pleased to announce the release of Asterisk 1.8.9.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.9.0 resolves several issues reported by the community and would have not been possible without your participation.

Thank you!

The following is a sample of the issues resolved in this release:

* AST-2012-001: prevent crash when an SDP offer is received with an encrypted video stream when support for video is disabled and res_srtp is loaded. (closes issue ASTERISK-19202)
Reported by: Catalin Sanda

* Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop. Failing
to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop
causes the loop to exit prematurely. This causes a variety of negative side
effects, depending on when the loop exits. This patch handles the frame by
essentially swallowing the frame in the local loop, as the current channel
drivers expect the RTP bridge to handle the frame, and, in the case of the
local bridge loop, no additional action is necessary.
(closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested
by: Matt Jordan

* Fix timing source dependency issues with MOH. Prior to this patch,
res_musiconhold existed at the same module priority level as the timing
sources that it depends on. This would cause a problem when music on
hold was reloaded, as the timing source could be changed after
res_musiconhold was processed. This patch adds a new module priority
level, AST_MODPRI_TIMING, that the various timing modules are now loaded
at. This now occurs before loading other resource modules, such
that the timing source is guaranteed to be set prior to resolving
the timing source dependencies.
(closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H,
Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont
Patched by elguero

* Fix RTP reference leak. If a blind transfer were initiated using a
REFER without a prior reINVITE to place the call on hold, AND if Asterisk
were sending RTCP reports, then there was a reference leak for the
RTP instance of the transferrer.
(closes issue ASTERISK-19192) Reported by: Tyuta Vitali

* Fix blind transfers from failing if an ‘h’ extension
is present. This prevents the ‘h’ extension from being run on the
transferee channel when it is transferred via a native transfer
mechanism such as SIP REFER. (closes issue ASTERISK-19173) Reported
by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by
Mark Michelson (license 5049)

* Restore call progress code for analog ports. Extracting sig_analog
from chan_dahdi lost call progress detection functionality. Fix
analog ports from considering a call answered immediately after
dialing has completed if the callprogress option is enabled.
(closes issue ASTERISK-18841)
Reported by: Richard Miller Patched by Richard Miller

* Fix regression that ‘rtp/rtcp set debup ip’ only works when a port
was also specified.
(closes issue ASTERISK-18693) Reported by: Davide Dal Reviewed by:
Walter Doekes

For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.0

Thank you for your continued support of Asterisk!