[Freeswitch-users] Help! FS Unable to Handle Incoming Calls Through SIP Gateway

[Yasuro]

Peter and David, thanks for trying to help out.

I did wonder about that too, Peter, but it seems to me that my VoIP 
adapter at 192.168.11.250 sends out an RTCP receiver report right before 
it ends a call. This is shown in the trace of packet exchange between 
AsteriskWin32 and the VoIP adapter 
<http://pastebin.freeswitch.org/14457>. In case you did not see my 
original posting, AsteriskWin32 does do what I want to achieve, i.e., 
automatically answers incoming calls to my DID number.

I have FreeSWITCH at 192.168.11.11 register with the VoIP adapter, which 
is a SIP gateway provided by my SIP service provider. When I call FS's 
extension number from another softphone, also on the home LAN, which is 
also registered with the same VoIP adapter, FS answers automatically as 
I expect it to. The RTP steam is established between FS and the 
softphone directly. This part is different from incoming calls' case, 
where the VoIP adapter seems to act as a proxy and tries to establish an 
RTP stream between itself and FS.

The VoIP adapter sends an ACK back to FS when FS accepts an INVITE with 
an OK. I'd think everything is hunky dory up to that point. What I do 
not get is why the VoIP adapter decides to end the call immediately 
after (and sends an RTCP receiver report). I am guessing it is because 
there is something wrong with the RTP communication that is supposed to 
follow.

I am thinking it is either because: A. the VoIP adapter somehow couldn't 
send RTP packets, or B. the VoIP adapter didn't like the RTP packets 
that it received from FS. I don't think A. is the case. The firewall of 
the PC on which FS runs was turned off during testing and the VoIP 
adapter has no problem sending RTP packets to AsteriskWin32 on the same 
PC (I did not run FS and Asterisk simultaneously). Since I could not 
find anything particularly odd about RTP packets sent by FS, I have no 
idea if B. is a possibility.

By the way, my home LAN is set up in a weird way (please see 
http://i139.photobucket.com/albums/q302/starplatina/For%20Blogs/HomeLANSetup-1.jpg). 
Double NAT layers are understandably a concern, but since both the VoIP 
adapter and FS (and the softphone too) are accessible on the inner layer 
of NAT, this should not be a problem... I would think (Correct me if I 
am wrong). I do not need to and thus am not trying to access FS from the 
Internet.

I would welcome any input. Thanks again for your help!

Yasuro



Peter Steinbach wrote (11/15/2010 9:01 AM):
> Thanks David,
>
> missed the "C".
>
> But anyway, I am wondering why .250 sends a BYE right after reception of
> "Destination unreachable (Port unreachable)".
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20101115/1b8d4c4c/attachment.html