search for: johnml

I've been playing around with asterisk for the past few days. One thing which came to mind which could be a useful addition to the meetme functionality would be the possibility to specify a moderator password in meetme.conf. (A moderator in the sense that music is heard until the moderator opens the call and then the call is disconnected when they hangup - options wx). At the moment it

I have successfully configured asterisk to make outgoing calls over FWD, but cannot receive incoming calls. The console shows no messages, even though an XTEN client on the same network has no problems receiving incoming calls. This is the relevant part of sip.conf [general] ..... register => 688426:xxxxxxxx@fwd.pulver.com/6000 [fwd.pulver.com] type=friend username=688426 fromuser=688426

I'm looking at experimenting with asterisk with an ISDN BRI and ISDN phones (since I have these already). I saw that the Billion card was cheap and could be used in either TE or NT modes. I have the following question which I couldn't answer by reading through the manual. Maybe someone has experience of using this card and can help me out. when using in NT mode does the card require

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've have installed a new Asterisk 1.4.15 system after having previously used a 1.2 CVS head (from 10 Sep 2005). Both systems are pentiums though the newer one is actually a slower processor. On the new system, playback of gsm files is noticeably poorer (voice quality is flakely) on any connected phone (sip or isdn, internal or external).

I've now setup SIP for: - internal softphones - registering with external providers (like FWD) for making calls - receiving calls from theese providers For the latter step, it was necessary to forward ports from my NAT to the asterisk server: 5060 + range of ports mentioned in rtp.conf. I was just wondering about how to make this setup as secure as possible. Here's what I've done so