Chris Rowson
2008-Jul-12 23:03 UTC
[asterisk-users] Incoming call does not reach asterisk.
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 *:* -----------------------------------------------------------------------------------------
Hello,
On Sun, 13 Jul 2008, 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: > > register => 277****:********t at sipgate.co.uk/277****You should use ngrep when making a call to see what is happening on the wire. You don't mention whether or not you can make outbound calls so I will ask now, can you make outbound calls? What do you see on a sip show peer 277**** is your line registered. I'm unsure about Astlinux but if you've seen em one you've seen em all, is iptables running on the machine itself (iptables -L), is the device connected properly, can it reach other places say Google. Are you doing NAT if so, did you configure a STUN server, did you specific NAT in sip.conf. In a terminal on the configured box - run the following: ngrep -d YOUR_ETHERNET_CARD 227**** udp port 5060 Place a call, what do you see, if nothing comes through (these are SIP messages by the way) then its not hitting your machine period, whether its a firewall, ACL on a router, doesn't make a difference, its not hitting the box, you have to troubleshoot from there. Starting point, throw the box in a DMZ, with the same ngrep command, place another call, technically you should see some messages hitting the machine. Have you contacted your provider, are they doing any kind of IP address filtering, SIP filtering. -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo SGFA #579 (FW+VPN v4.1) SGFE #574 (FW+VPN v4.1) CEH/CNDA, CHFI "Experience hath shewn, that even under the best forms (of government) those entrusted with power have, in time, and by slow operations, perverted it into tyranny." Thomas Jefferson wget -qO - www.infiltrated.net/sig|perl http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x3AC173DB