* You are viewing the archive for June 14th, 2012

Does Asterisk support AMR and AMR-WB

Hi all, I have a project for the 3G related, AMR and AMR-WB support.

I’m using the client develop suite from the PortSIP(http://www.portsip.com),
as their said
support the AMR, AMR-WB with RFC4867.

Now I have to setup a SIP server/SIP PBX in our Lab for test, does the
support these codecs and RFC4867 ? If no, there has any plugin to support
this ?

Also, any other Server/PBX which support AMR, AMR-WB recommended are

Best regards,

Digium IP Phones – Teleworker Capability?

We couldn’t see anything about this on the Digium site, but maybe
someone here can comment?

Do the new Digium phones provide good “teleworker” functionality?

The benchmark we’re comparing against is the capabilities of Mitel
3300 IP systems with Mitel 5330 IP phones (running their proprietary
MINET protocol), specifically:

a. A Mitel phone can be easily configured for teleworker mode (select
TW mode and the IP of the gateway server). The phone reboots and it
is ready to be used (once the Mitel border gateway is set to recognize
the unit’s ID, based on its MAC address, printed on the label on the
back of the phone). If the phone gets reallocated back to a directly
connected office environment, a simple reset procedure brings it back.

b. You can plug in the phone virtually anywhere. It has a built-in
tunnelling mechanism providing end-to-end encryption and is very
tolerant of the network configuration, routers, NAT, etc.

c. If the link between the phone and the gateway goes down, the phone
will restore itself gracefully and automatically once the network
function resumes. Absolutely hassle-free to the user.

d. Users can be configured to have hot-desk functionality. The phone
has a default extension assigned, but the user can be set up so that
they can “log in” to their normal office extension number from
wherever they are. Their office phone is automatically logged-out and
goes to its default extension when you log in to a teleworker phone
(you don’t have to log out from it first). Your phone buttons,
display settings, voicemail WMI and access, (everything) move to this
new phone, and you can work from your home office, on the road, etc.,
and inbound and outbound calls work just like you were there in the
office (callerid, etc).

These four features would be a big selling point for us to consider
moving our organization from Mitel to Digium/Asterisk/Switchvox.

How much of this can be done with Asterisk/Switchvox and, say, the
Digium D70 phone with dynamic button display?

Thanks for all comments!

Skinny Channel Driver Remote Crash Vulnerability

A previously developed patch dealt with a denial of service attack exploitable in the Skinny channel driver that occurred when certain messages are sent after a previously registered station sends an Off Hook message. Unresolved in that patch is an issue in the Asterisk 10 releases, wherein, if a Station Key Pad Button Message is processed after an Off Hook message, the channel driver will inappropriately dereference a Null pointer.

Similar to the problem solved with the previous patch, a remote attacker with a valid SCCP ID can use this vulnerability by closing a connection to the Asterisk server when a station is in the “Off Hook” call state and crash the server.

Now the presence of a device for a line is checked in the appropriate channel callbacks, preventing the crash.

you can download the latest Asterisk packages in the download section, as usual.

Stay tunned for more security updates.


I have an Asterisk (v10.2.0) running and bound to address “::”. I think
this way he listens and answers to requests send to the IPv4 and IPv6
address (haven’t check that with IPv6 yet). What I want to achieve is,
that he handles signaling via IPv4, but RTP via IPv6.
In my setup, I have a user agent (Dualstack) that generates an INVITE
and sends it out via IPv6. In the SDP part the user agent expects the
RTP traffic on its IPv6 address as well. Between the user agent and the
Asterisk, I have a proxy that handles the signaling part, and translates
from IPv6 to IPv4 and vice versa. The Asterisk accepts that request
(IPv4) and does everything well, except that in his SDP offer, he
inserts his IPv4 address (I think that’s because he received the request
via IPv4).
The result of this is:
The user agent sends RTP traffic via IPv4 to the Asterisk.
The Asterisk sends RTP traffic via IPv6 to the user agent.

Signaling: UA < ---- (IPv6) ----> Proxy < ---- (IPv4) ----> Asterisk
RTP: UA ——————- (IPv4) ——————–> Asterisk
UA < ----------------- (IPv6) ---------------------- Asterisk

Does anybody know how I can achieve that Asterisk does input his IPv6
address in the SDP offer and uses that for incoming RTP, if he sees,
that the user agent also uses an IPv6 address in his SDP offer?
Maybe there is an easy way and I’ve just overseen a configuration
option. Or do I have to patch the sources?

Thanks in advance!

Asterisk 10.5.1 Now Available (Security Release)

The Asterisk Development Team has announced a security release for Asterisk 10.
This security release is released as version 10.5.1.

The release is available for immediate download at

The release of Asterisk 10.5.1 resolves the following issue:

* A remotely exploitable crash vulnerability was found in the Skinny (SCCP)
Channel driver. When an SCCP client sends an Off Hook message, followed by
a Key Pad Button Message, a structure that was previously set to NULL is
dereferenced. This allows remote authenticated connections the ability to
cause a crash in the server, denying services to legitimate users.

This issue and its resolution is described in the security advisory.

For more information about the details of this vulnerability, please read
security advisory AST-2012-009, which was released at the same time as this

For a full list of changes in the current releases, please see the ChangeLog:


The security advisory is available at:

* http://downloads.asterisk.org/pub/security/AST-2012-009.pdf

Thank you for your continued support of Asterisk!

asterisk with ss7 voice broadcast


Asterisk under > 90% load of SS7 calls can only withstand the voice
broadcasting for 30 minutes. After around 30 minutes, it stops receiving
any call hits via AMI. No errors are reported. Giving it a minute’s rest
makes it work for another 30 minutes.

Can anyone hint to what may be causing this?