I am attempting to dial an Asterisk installation on CentOS 5.4, using Asterisk v1.6.2.12 from the digium RPM repository. I’ve verified using tcpdump that inbound SIP UDP packets are received at the eth0 interface of this machine, but no response is sent from asterisk. Connecting to Asterisk with the option “-vvvvvvvvr” shows no output when the SIP packet arrives. Contrast this with the output from an out-of-the-box Asterisk installation on my Ubuntu server where I show the following debug line:
[Oct 12 16:18:01] NOTICE[19663] chan_sip.c: Call from '' to extension 'test' rejected because extension not found.
Here is the configuration from my non-working Asterisk installation (commented out portions not included):
sip.conf
[general]
context=default ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
tcpenable=no ; Enable server for incoming TCP connections (default is no)
tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
srvlookup=yes
[basic-options](!) ; a template
dtmfmode=rfc2833
context=from-office
type=friend
[natted-phone](!,basic-options) ; another template inheriting basic-options
nat=yes
directmedia=no
host=dynamic
[public-phone](!,basic-options) ; another template inheriting basic-options
nat=no
directmedia=yes
[my-codecs](!) ; a template for my preferred codecs
disallow=all
allow=ilbc
allow=g729
allow=gsm
allow=g723
allow=ulaw
[ulaw-phone](!) ; and another one for ulaw-only
disallow=all
allow=ulaw
extensions.conf:
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
CONSOLE=Console/dsp
IAXINFO=guest
TRUNK=Zap/g2
TRUNKMSD=1
[default]
; E1C extensions
; load all server-generated extension snippets
#include /opt/resources/phone/asterisk/extensions/*.conf
; test
exten => test,1,Answer
exten => test,2,Set(trips = 1)
exten => test,3,Wait,1
exten => test,4,Playback(hello-world)
exten => test,5,GotoIf($[${trips} = 5]?6:7)
exten => test,6,Goto(9999,1)
exten => test,7,Set(trips=$[${trips} + 1])
exten => test,8,Wait,2
exten => test,9,Playback(beep)
exten => test,10,Goto(test,5)
exten => 9999,1,Playback(demo-thanks)
exten => 9999,2,Hangup()
I am using the pjsua command line sip client to make the SIP call, which works fine against my other (Ubuntu) asterisk installation. Also, when attempting to start pjsua on my CentOS machine (with the non-working Asterisk) it fails to bind to port 5060 - telling me that asterisk did indeed startup and bind to that port. Any ideas?