[Asterisk-Users] TO header field bug using asterisk

[usmankhan]

Hi,

  I have setup and configured asterisk server using SIP. I defined 
two users 2000 and 2001 in the sip.conf file. The extension 
defintions from sip.conf are shown below:
 
[2000]
type=friend
username=2000
secret=hi
host=dynamic
context=from-sip
mailbox=100
 
[2001]
type=friend
username=2001
secret=hi
host=dynamic
context=from-sip
mailbox=101
 
I made entries in extensions.conf, shown below, to place calls to 
2000 and 2001

[from-sip]
exten=>2000,1,Dial(SIP/2000,20)
exten=>2000,2,Voicemail(u2000)
exten=>2000,102,Voicemail(b2000)
exten=>2000,103,Hangup

exten=>2001,1,Dial(SIP/2001,20)
exten=>2001,2,Voicemail(u2001)
exten=>2001,102,Voicemail(b2001)
exten=>2001,103,Hangup

 Initially I tested out the server by registering SJPhone user 
agents and successfully placing calls between them.

 Next I replaced the SJPhones with our VOIP gateways. Everytime I dialed
either extension I always got the unavailable IVR message. I tried looking
deeper into the problem and took ethereal traces and was able to 
isolate the problem. For some reason asterisk has problems in 
rewriting the TO header field when it forwards the INVITE request to 
the callee. This is what the TO header field looks like when it is 
sent by the caller to asterisk
(192.168.0.44 is the IP address of asterisk):

To: <sip:2000 at 192.168.0.44:5060>

and this is what it looks like when it is forwarded to the callee by 
asterisk (192.168.0.243 is the IP address of the callee).

To: <sip:192.168.0.243>

Since the URI does not contain the user part the callee replies with 
404 not found and the call fails. I have thought hard, compared 
signaling traces but cant really make out how to make my gateways 
work, seems like an asterisk bug. Any ideas? I would really appreciate any
help in this regard.

Regards,
Danish