provided by Star2Billing S.L.

To whoever can suggest any ideas,

I cam across this problem and trying for hours to fix but no luck.

Here's the situation

I have asterisk 1.4.18 installed and I have a SIP provider called "provider1"
[provider1]
type=peer
host=sip.xxxxxx.com
username=0xxxxx
secret=xxxxx
qualify=yes
dtmfmode=RFC2833
canreinvite=no
sendrpid=yes
nat=yes
insecure=port,invite
disallow=all
allow=g729

When I dial out using the above trunk directly it works just fine.
e.g.
[dialout]
exten => _X.,1,Dial(SIP/provider1/${EXTEN})

When I dial using a2billing.php the two calls connected but there is just no voice/sound on both sides.
[a2billing]
exten => _X.,n,DeadAGI(a2billing.php|2)

Any ideas why?

The really strange thing is that, if I put this provider1 trunck on a different asterisk server going through the exact same provider1 via A2B and it worked.
So here are the scenarios
1) [my asterisk box 1] ==>Provider1 (this works)
2) [my asterisk box 1] ==>A2B==>Provider1 (calls connects but no sound)
3) [my asterisk box 1] ==>A2B==>[exact same Provider1 on asterisk box 2] (this works)

And I have tried three different providers and tried all the codecs availbe and still the same problems. Going through A2B the phone rings and connects but no sound.

Help please .

I'm guessing this is a codec or NAT issue. You should analyse the output of the Asterisk console with 'set verbose 15' whilst placing a good and bad call. If that yields no clues try again after enabling 'sip debug'. voip-info wiki will undoubtedly provide assistance with your understanding of this and other issues.
When debugging VoIP problems it's often quicker to fire up Wireshark and analyse the call legs from the captured packets directly, rather than piecing together SIP dialogues from multiple Asterisk consoles. You will need to apply an understanding of IP, CIDR, SIP, SDP and RTP to gain any benefit from this method.

Finally, after hours of testing... a fix was found

As soon as I unload the module ip_nat_sip from the kernel. Whola! I can hear the sounds now!

Excellent news! Well diagnosed!
Of course, voip-info were not going to be easily defeated.
Is there anything Asterisk related that site doesn't document? Except A2B of course.