[Asterisk-Users] interoperablity problems between Asterisk and VOIP gateway

[usmankhan]

Hi,

 I have setup and configured asterisk server using SIP. I defined two users
2000 and 2001 inthe 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. I would really appreciate
any help.

Regards,
Danish