Ext Calls: "All Circuits Busy" Int:OK

Any help with this would be hugely appreciated. I’ve been searching and experimenting for days!

I’m running Trixbox on two SIP trunks with the same provider. When we put the providers login details into our VOIP box/router (Gateway NB9), it lets us call out just fine.

We can make internal calls between extensions just fine.

  • The router allows G729, Ulaw and Alaw
  • Our VSP allows: G729 and GSM
  • Our Trixbox allows: G729 (2 licenses), Ulaw, Alaw, GSM etc.
  • Our Softphone (xlite) allows Ulaw, Alaw, GSM but NOT g729

If we try and connect directly to our VSP we get “Not acceptable here”, presumably because there is no single codec that can go from Xlite to the Router to our VSP.

If we try and get the Router (Netcomm NB9) to log into asterisk directly, it registers, but on calling always just returns “All Circuits Busy. Please try again later.” I would have thought that in this case, calling through VOIP box (NB9) to asterisk to the router (also NB9) to the VSP all would have just used G729.

I have my firewall / NAT temporarily turned off.

My sip.conf is:

[quote]; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn’t, try adding “nat=1” to each peer definition to
; solve translation problems.

[general]

bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc
allow=g729
; If you need to answer unauthenticated calls, you should change this
; next line to ‘from-trunk’, rather than ‘from-sip-external’.
; You’ll know this is happening if when you call in you get a message
; saying "The number you have dialed is not in service. Please check the
; number and try again."
context = from-trunk ; Send unknown SIP callers to this context
callerid = Unknown
tos=0x68

; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
[/quote]

my sip_additional.conf is:

[quote]; do not edit this file, this is an auto-generated file by freepbx
; all modifications must be done from the web gui
register=2877746:XXXXXXXXXX@platypus.ecn.net.au/2877746
register=4316941:XXXXXXXXXX@platypus.ecn.net.au/4316941

[101]
type=friend
secret=XXXXXXXXX
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=yes
mailbox=101@default
host=dynamic
dtmfmode=rfc2833
dial=SIP/101
context=from-internal
canreinvite=no
callerid=device <101>

[102]
type=friend
secret=XXXXXXXXX
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=yes
mailbox=102@default
host=dynamic
dtmfmode=rfc2833
dial=SIP/102
context=from-internal
canreinvite=no
callerid=device <102>

[2877746]
username=2877746
type=user
secret=XXXXXXXXX
qualify=no
nat=1
fromuser=2877746
context=from-trunk
canreinvite=no

[4316941]
username=4316941
type=user
secret=XXXXXXXXX
qualify=no
fromuser=4316941
context=from-trunk
canreinvite=no

[globechat1]
username=2877746
type=peer
secret=XXXXXXXXX
nat=1
insecure=very
host=platypus.ecn.net.au
disallow=all
canreinvite=no
canredirect=no
allow=g729
allow=gsm

[globechat2]
username=4316941
type=peer
secret=XXXXXXXXX
insecure=very
host=platypus.ecn.net.au
disallow=all
canreinvite=no
canredirect=no
allow=g729
allow=gsm

[/quote]

My extensions.conf is:

Obviously the XXXXXXXXX’s a placeholders for my passwords.[/quote]