asterisk users - Aug 2004 - SIP Provider in India/Pakistan/Bengladesh

Hello All,
 
We are looking for a SIP provider teminating calls in India, Pakistan
and Bengladesh.
Any one knows a good one? 
 
Regards,
Cesar
 
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I have a test box setup and I can make outbound calls on the PSTN thru the
diguim card, however I can not make a sip user to sip user call by dialing the
extensions. I am getting the following error.

-- Called cisco7960
    -- Got SIP response 482 "Loop Detected" back from 208.218.14.123
  == No one is available to answer at this time



CLI> sip show peers
Name/username    Host            Dyn Nat ACL Mask             Port     Status

cisco7960/5052   208.218.14.123   D   N      255.255.255.255  5060     OK (1 ms)
garycarr/5011    208.218.14.123   D   N      255.255.255.255  5060     OK (1 ms)


sip.conf statements

register => garycarr@sip.carolina.net/5011
register => cisco7960@sip.carolina.net/5052

[cisco7960]
type=friend
host=dynamic
nat=yes
qualify=200
dtmfmode=rfc2833
canreinvite=no
mailbox=5052
callerid="Cisco 7960"
context=local

[garycarr]
type=friend
host=dynamic
nat=yes
qualify=200
dtmfmode=rfc2833
canreinvite=no
mailbox=5011
callerid="Gary Carr"
context=local

extensions.conf statements

exten => 5011,1,dial(SIP/garycarr,20,tr)
exten => 5052,1,dial(SIP/cisco7960,20,tr)

Is this a possible nat issue? I can make a good call from behind the firewall
doing sip to pstn so it seems 2 way traffic thru the firewall is working.


I am still sifting thru the sip debug info but anyone has any ideas that would
be great.


Gary
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PTA(Pakistan Telecommunication Authority) has recently issued LDI
licences to number of contenders and use VoIP. noone yet has  announced
but very soon someone from them will announce SIP termination in
Pakistan.

can't say anything about India/Bengladesh

-- 
Atif