[Asterisk-Users] Is somthing broken?

[WipeOut]

Hi,

I updated my live server yesterday(after testing on my Dev server first, 
all works on the Dev server)..

Here is the setup..

SIP_UA---[NAT]---Asterisk1---PSTN(chan_capi)

The SIP_UA is able to recieve calls from the server with no problems.. 
Initiated from the PSTN or my Dev Asterisk box which is connected to 
Asterisk1 with IAX..

When the SIP_UA tries to make calls out via the PSTN or to Voicemail on 
Asterisk1 or another extention there is no sound..

The definition in sip.conf is fairly standard(included below)..

This config has been working fine for months.. the last update was about 
1 month ago so sometime between then and now it seems that SIP has 
changed and so stopped working..

Hopefully this can be solved quickly becasue it is a problem..

Later..


Definition from sip.conf
[2014]
context=users
type=friend
secret=magic
nat=yes
canreinvite=no
dtmfmode=info           ; Grandstream
host=dynamic
mailbox=2014            ; Mailbox for message waiting indicator