[Users] inbound call failed

[Klaus Darilion]

unplug wrote:
>>From the log, it showed lookup(): 'account name' Not found in usrloc. 
> I think it is the NAT problem, so I use stun for the case below.

Does the INVITE from the GW does have a proper formated request URI? The 
  called phone number must be mapped to the username before doing 
lookup("location"). E.g. you can use aliases to map phone numbers to 
user names lookup("aliases"); (or use the aliasdb module)

regards
klaus


> 
> Telephone (A)
>   |
> PSTN
>   |
> G/W
>   |
> openser
>   |
> NAT -- IP phone(B) -- STUN
>   + ----- IP phone(C) ------+
> 
> Below is the result:
> A to B/C is ok
> B/C to A is ok
> However, there is no sound when B to C or vice versa.  What reason
> will cause no sound between B and C?  Is the the reason from the
> NAT/STUN?
> 
> 
> On 12/12/05, Klaus Darilion <klaus.mailinglists at pernau.at> wrote:
> 
>>Use ngrep to watch for incoming SIP requests on the SIP proxy.
>>
>>Take a look at the logfiles on the gateway.
>>
>>klaus
>>
>>unplug wrote:
>>
>>>Below is the common configuration of the network.
>>>
>>>Telephone -- PSTN -- G/W -- openser -- softphone (eg windows messenger)
>>>
>>>I can make call from softphone to Telephone.  However, it is failed to
>>>make call from Telephone to softphone.  I wonder why it happened and
>>>any reference to trace the problem.  Anyone have such experience?
>>>
>>>_______________________________________________
>>>Users mailing list
>>>Users at openser.org
>>>http://openser.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>>
> 
>