I have recently started with asterisk after using switchvox for a while.
I have asterisk set up on a machine inside NAT. On my draytek router I have opened ports UDP 5060, 5004, 10000-20000 to my server running asterisk.
I have two softphones setup and they register and comminicate with each other.
I have a sipgate.co.uk account setup. In asterisk CLI I can run ‘sip show registry’ and the following is returned:-
Host Username Refresh State Reg.Time
sipgate.co.uk:5060 1234567 105 Registered Thu, 04 Feb 2010 22:11:42
If I log in to my sipgate account I see:-
[code]Status: Online
- Device: Asterisk PBX 1.6.0.21
IP: sip:1234567@my.external.ip.address[/code]
From one of my softphones i can dial my mobile number and that works perfectly.
If I then call my telephone number provided by my sip provider I get a double beep and nothing happens, nothing appears in the cli with vvvvv verbosity.
I have tried many different sip.conf and extensions.conf from so many websites but none work.
My feelings suggest that it’s not the asterisk configurations at fault as nothing is appearing in the logs. See below my current sip.conf and extensions.conf anyway.
Any advice is greatly received before I rip what hair I have left out. I appreciate some of the contexts need changing but these are the last settings I tried after following this link growse.com/projects/setting- … -asterisk/
I have also setup trixbox and have been able to get it all working but wanted more satisfaction of building asterisk from scratch.
sip.conf
[code][general]
qualify=no
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
register => 1234567:PASSWORD@sipgate.co.uk/1234567
externip=my.external.ip.address
localnet=192.168.200.0/255.255.255.0
nat=yes
[sipgate.co.uk]
type=peer
secret=PASSWORD
username=1234567
host=sipgate.co.uk
fromuser=1234567
outgoingproxy=sipgate.co.uk
canreinvite=no
dtmfmode=inband
nat=yes
insecure=very
context=default
[1000]
context=default
type=friend
host=dynamic
secret=1234
mailbox=1000@default
[1001]
context=default
type=friend
host=dynamic
secret=5678
mailbox=1001@default[/code]
extensions.conf
[code][general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
SpeakingClock=2
[default]
exten => 1234567,1,Dial(SIP/1000)
exten => 1234567,2,VoiceMail(1000@default)
exten => 1234567,3,HangUp()
exten => _9.,1,Dial(SIP/${EXTEN:1}@sipgate.co.uk,30,r)
exten => 500,1,Playback(demo-echotest)
exten => 500,n,Echo()
exten => 500,n,Hangup()
exten => 501,1,Answer
exten => 501,n,Playback(tt-weasels)
exten=> 501,n,Hangup()
exten => 1999,1,VoicemailMain
exten => 1999,n,Hangup
exten => 1001,1,Dial(SIP/1001,5)
exten => 1001,n,Voicemail(1001,u)
exten => 1001,n,Hangup()
exten => 1000,1,Dial(SIP/1000,5)
exten => 1000,n,Voicemail(1000,u)
exten => 1000,n,Hangup()
exten => ${SpeakingClock},1,Wait(1)
exten => ${SpeakingClock},2,setvar(FutureTime=$[${EPOCH} + 10])
exten => ${SpeakingClock},3,playback(at-tone-time-exactly)
exten => ${SpeakingClock},4,SayUnixTime(${FutureTime},R)
exten => ${SpeakingClock},5,playback(vm-and)
exten => ${SpeakingClock},6,SayUnixTime(${FutureTime},S)
exten => ${SpeakingClock},7,playback(seconds)
exten => ${SpeakingClock},8,playback(beep)
exten => ${SpeakingClock},9,wait(2)
exten => ${SpeakingClock},10,goto(1)[/code]