SIP trunking problem: no path to translate

Hello All,
I have the following scenario :

I am running the asterisk server version 1.8 on my Ubuntu 12.04 laptop 32 bit.

I want to make a sip trunking in my asterisk server to call to indian land line and mobile phone. I have found out the Voip provider in india who provide cheap rates to call to India. My asterisk server is behind my Fritzbox router. (I am located in Germany). I have two zoiper soft phones which are registered to my asterisk server on the local network. I can register the indian voip provider sip trunk in my asterisk server, but when I want to call through one of my Zoiper soft phone to an indian number by dialing 919830911247 (They have told me to dial in the following format) I get the following error on the asterisk console :
WARNING[4961]: channel.c:5794 ast_channel_make_compatible_helper: No path to translate from SIP/ISevenSupport-00000014 to SIP/789-00000013

In Zoiper the call stops immediately.
But when I try to call directly from my Zoiper soft phone without asterisk using this voip provider I can call the indian number.

Here is my sip.conf and extension.conf


sip.conf

[code][general]
context=unauthenticated
allowguest=yes
srvlookup=yes
bindport=5061
udpbindaddr=0.0.0.0
tcpenable=no
transport=UDP
localnet=192.168.1.0/255.255.255.0
externhost=astvoip.uenetz.de
insecure=invite
nat=force_rport,comedia
dtmfmode=auto
disallow=all
allow=gsm
allow = g729
allow = g723
allow=ulaw
allow=alaw
;maxexpiry=36000
;defaultexpiry=600
;tos=0x04
registerattempts=0
registertimeout=1
externrefresh=300
canreinvite=no
qualify=yes
;directmedia=no
register=XXXX:YYYYY@X.X.X.X

[ISevenSupport]
fromuser = XXXX
fromdomain = astvoip.uenetz.de
secret=7548127
disallow = all
allow = g729
allow = g723
allow = ulaw
allow = alaw
type=peer
port=5060
host =X.X.X.X
qualify = yes
nat = yes

home-phone
type=friend
context=RoyNetworks
host=dynamic
;nat=force_rport,co178media
;nat=yes

123
secret=xxxxxxx

456
secret=xxxxxxxx

789
secret=xxxxxxxxx[/code]


Extension.conf

[code][general]

[telefone]

[external]

exten => _91XXXXXXXXXX,1,Dial(SIP/ISevenSupport/${EXTEN})

[RoyNetworks]
include => external
exten => 123,1,Dial(SIP/123)
exten => 456,1,Dial(SIP/456)
exten => 789,1,Dial(SIP/789)[/code]


The voip provider in india is using the g729 and g723 codecs.

My Zoiper softphone does not use g729 or g723 codecs, in spite of that I can call to the indian number directly from the zoiper without routed through asterisk. Therefore I donot think it is a codecs problem. But I am not quite sure.
Hope to get a reply soon.
Roy

You need to buy a licence for the G.729 codec, or restrict the Indian trunk to only use free codecs.

Also:

allowguest is normally a bad idea.

If insecure=invite works in the general section, you have a dangerously insecure system that is very vulnerable to toll fraud, especially when you combine it with your use of type=friend. type=peer should work and should make the system a lot less easy to attack.

canreinvite is deprecated.

You have a mix of deprecated and near equivalent non-deprecated nat= settings.