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
Also, any other Server/PBX which support AMR, AMR-WB recommended are
welcome. Best regards,
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…
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…
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
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
http://downloads.asterisk.org/pub/telephony/asterisk/releases 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,…
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?