sipgate.co.uk inbound help

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:-

Code:

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 1. Device:   Asterisk PBX 1.6.0.21    IP: sip:1234567@my.external.ip.address

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 http://www.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

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]