Kevin P. Fleming recently posted an entry on the Asterisk Wiki about Asterisk SCF that goes as follows:… the Asterisk SCF SIP components will provide all the mechanisms required to implement authentication of incoming requests, but they wont prov..
Ive noticed on our system the sound files have to be in an exact format for Asterisk to play them. Bit Rate: 128kbps Audio sample size: 16 bit Channels: 1(mono) Audio Sample rate: 8kHz Audio format: PCM I actually downloaded a program and remixed ..
someone have been installed Asterisk (Trixbox) on VirtualBox which is installed on a Linux host (Ubuntu server 10.04 specifically). I want to know if it is convenient or not, and the reaseons if i should on shouldnt do it. Thanks in ..
Hey all I wanted to get some input on what you all think is the best way to lookup database data from asterisk dial plan. This is a two fold question. 1. I am using fun_odbc to pull settings and values back and it works good but is there a better w..
Has anyone heard (or read) about an existing or emerging standard targeting the following feature : 1. a SIP handset receives an incoming call 2. this handset starts ringing 3. then it receives an update asking to autoanswer the ringing call. This feat..
Using Asterisk 18.104.22.168
We are now starting to use *call transfer (patching) function.*
Call flow is as ..
list, I use Asterisk with one sipgate.co.uk trunk. Asterisk connects to sipgate.co.uk as a sip agent/client (with register => statement in sip.conf). If I restrict the number of ports used in rtp.conf (to 10000-10005 for example) – will that affect ..