This is a continuation of my previous post about the +39 ENUM DNS zone.

Thanks to the powerful of the RIPE Database you can see all historical changes on a RIPE-assigned domain. Lets see what happen on the domain 9.3.e164.arpa.

List all changes on the domain object ...

ENUM is a DNS based algorithm used to store data associated with E.164 numbers (aka global phone numbers). Using ENUM you can associate your email or Skype or SIP contact along with your phone number.

So, registering your contacts in the ENUM tree you'll make yourself reachable trough ...

One of the most powerful iptables match is the u32 module.
Using the u32 match you are able to write a firewall rule matching against a single bit of a network packet.

If you wanna play with u32 module you have to be prepared to deal with a syntax very ...

SIPp is a real SIP swiss army knife. You can create various SIP scenario and test your devices and environments.

I never found a SIPp scenario implementing an UAs (User Agent server) registrar with authentication.

Following you can find this scenario:

The SIPp XML scenario file: sipp_register_auth_uas.xml

<?xml version ...

Here you can find a port for Asterisk trunk (rev. 377802) of this patch: http://lists.digium.com/pipermail/asterisk-dev/2011-September/051262.html.
This patch permits to set the codec of an outgoing multicast RTP stream via the variable MULTICAST_RTP_CODEC:

exten => 999,1,Answer()  
exten => 999,n,Set(MULTICAST_RTP_CODEC=alaw ...

Starting from Asterisk 1.8 you can send multicast rtp streams using the MulticastRTP channel driver. There is an open issue that breaks outgoing RTP if the source channel doesn't contains timing informations (Eg. playing an audio file with Dial(MulticastRTP/basic/239.255.255.245:5555,,A(my-announce ...

This little piece of extensions.conf implements a callback from a failed transfer (attendant and unattendant/bind)

exten => _1XX,1,Dial(SIP/${EXTEN},20,tT)
exten => _1XX,n,GotoIf($[ "a${BLINDTRANSFER}" = "a" ]?TransferFailed)
exten => _1XX,n,Set(CALLER=${CUT(BLINDTRANSFER,-,1)});
exten => _1XX,n,Goto(CallBack)
exten => _1XX,n(TransferFailed ...

Using directmedia=update or (canreinvite=update in old-sytle configuration) in chan_sip.conf Asterisk sends SIP UPDATE request to refresh SIP session.

If this UPDATE requests refers to an inactive call the phone reply with SIP/2.0 481 Call Leg/Transaction Does Not Exist.

In this situation Asterisk continue in ...

Asterisk use peername as username during SIP inbound digest authentication.
This patch add authuser parameter in SIP peer definition and use authuser in digest authentication:

Peer definition in /etc/asterisk/sip.conf

...
[pietro](sip-client-base)
authuser=MyUsername
secret=XXXXX
qualify=yes
nat=yes
...

Peer definition in my SIP client (Twinkle)

REGISTER ...