(udp/5060, udp/2727, among others). One way to tell for sure would be to run 'lsof -i' which would show you the process associated with the port. As far as the call not reaching asterisk or being a firewall issue, one way to tell might be to start a tcpdump just prior to making the incoming call. Something like this: tcpdump -n 'port 5060' That would show the connection attempt. But if it's not showing up on the console then chances are that it's not even reaching the asterisk server to begin with. Steve On Sun, Jul 13, 2008 at 12:03:29AM +0100, Chris Rowson wrote:> Hi, this is my first post to the list, but I have tried to search > elsewhere for a solution, and have had a read of 'Asterisk - The > Future of Telephony'. So you could say that I have at least tried to > RTFM as it were! > > I've configured a couple of Asterisk instances on both Debian and > CentOS based VPS's, and got them working fine. However, I recently > installed a copy of Astlinux and installed on a WRAP board and I'm > totally stuck! > > I'm using sipgate.co.uk for incoming calls, but when I make a test > call from the PSTN, the call just dies without connecting to my > Astlinux box. (I'm monitoring asterisk console via 'asterisk -rvvvvv' > and see nothing). > > I wondered if it might be a problem with Asterisk not listening > properly, or perhaps a problem with my home firewall. Would anyone be > kind enough to advise me as to where I may have gone wrong? > > Thanks, Chris. > > My sip.conf looks like this: > > ------------------------------------------------------------------------------ > [general] > context = default ;default context for incoming calls > bindport = 5060 > bindaddr = 0.0.0.0 > srvlookup = yes > disallow=all ;disallow all codecs > allow=alaw ;except alaw (1st pref) > allow=ulaw ;and ulaw (second pref) > > register => 277****:********t at sipgate.co.uk/277**** > > [sipgate] ;sipgate sip in on 01482 77**** > type=peer > context=from-pots > fromuser=277**** > username=277**** > authuser=277**** > secret=*********** > host=sipgate.co.uk > fromdomain=sipgate.co.uk > dtmfmode=inband > insecure=very > canreinvite=no > disallow=all > allow=alaw > allow=ulaw > nat=yes > qualify=yes > ------------------------------------------------------------------------------------- > My extensions.conf looks like this: > > ------------------------------------------------------------------------------------- > [general] > static=yes > writeprotect=np > autofallthrough=yes > clearglobalvars=no > priorityjumping=no > > [from-pots] > exten => s,1,Answer() > exten => s,n,Wait(3) > exten => s,n,Playback(tt-weasels) > exten => s,n,Hangup() > -------------------------------------------------------------------------------------- > > and netstat looks like this > > -------------------------------------------------------------------------------------- > Active Internet connections (only servers) > Proto Recv-Q Send-Q Local Address Foreign Address State > tcp 0 0 *:www *:* LISTEN > tcp 0 0 *:ftp *:* LISTEN > tcp 0 0 *:ssh *:* LISTEN > tcp 0 0 *:https *:* LISTEN > udp 0 0 *:1025 *:* > udp 0 0 *:1026 *:* > udp 0 0 *:1027 *:* > udp 0 0 *:1028 *:* > udp 0 0 *:1029 *:* > udp 0 0 *:1030 *:* > udp 0 0 *:1031 *:* > udp 0 0 *:1032 *:* > udp 0 0 *:2727 *:* > udp 0 0 *:4520 *:* > udp 0 0 *:5060 *:* > udp 0 0 *:tftp *:* > udp 0 0 *:4569 *:* > udp 0 0 *:5353 *:* > udp 0 0 *:5353 *:* > udp 0 0 *:5353 *:* > udp 0 0 *:5353 *:* > udp 0 0 *:5353 *:* > udp 0 0 *:5353 *:* > udp 0 0 *:5353 *:* > udp 0 0 *:5353 *:* > udp 0 0 *:ntp *:* > ----------------------------------------------------------------------------------------- > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users