All, Is it possible to specify more than 1 localnet? I know this is an odd question. I have a customer that has multiple sites linked by VPN. Main range is 192.168.33.0/24 and a remote site is 10.1.1.0/24 We want to allow some access to the public..
I am getting several hundred registration attempts on my aserterisk per minute.I have fail2ban installed but its not stopping the attempts.Any suggestions.Whatever they are using is changing theuserid on each attempt. Latest IP: 184.108.40.206 Than..
In 1.4 I used alsa.conf and Dial(Console/Dsp) In 1.8 this is not working (as I had it) . I know there is a new chan_console Id like to try both. What is the correct Dial() for ALSA direct? What is the correct Dial() for chan_console? I thought if chan_a..
I recently came across this email that I wrote in May 2008 …… http://lists.digium.com/pipermail/asterisk-users/2008-May/210887.html Its such a shame that Digium manhandled the project away from the community only to then bury it and not allow..
Im trying to use a couple of old Western Electric type 500 phones (desk model, rotary dial).These phones work fine, as tested with telco lines (they dial, receiver/transmitter works fine, etc). Im running Asterisk 220.127.116.11. I cant get them to dial thro..
All- Recently an Asterisk server we host was hacked and used to route some unauthorized calls.We have since improved our security measures, including installation of fail2ban. The interesting thing is the way in which this was discovered.The unauthori..
All, We have a production system running 18.104.22.168 and yesterday we upgraded it to 1.4.36. Basically we use this system to generate scheduled calls via .call files. Sample .call file used: Channel: local/111111111@context-out WaitTime: 30 CallerId: 333333..
Is there the ability in the Asterisk 1.8 CEL logging to log the SIP
endpoint IP as weell as the medie enpoints IDs?
i setup successfull asterisk version 1.4 + opensips, Opensips is the Registrar Server, Asterisk is the IVR server the call flow IP phone —INVITE 1001—-> opensips —–> ASterisk —-INVITE 5001—>opensips —> Busy|cancel|404..—>asterisk—w..
I have a Netmod ISDN BRI router and from the router I have connected the analog port in Asterisk via FXO card. Two analog lines I have connected to asterisk machine. When both the lines are established, after 31 minutes the call is automatically disconnect..