asterisk users - Jan 2004 - Cisco FXO as PSTN gateway (updated request for assistance)

I have been compiling information on this configuration onto the Wiki:
http://voip-info.org/wiki-Asterisk+cisco+FXO

I can call out to the PSTN just fine, but inbound calls all appear in my 
default [bogon-calls] context, not in [pstn-incoming]

Can anyone help me locate why?
(Config files are on the Wiki)

I have done a packet sniff & decoded using Ethereal-0.10.0, but this
doesn't tell me a great deal - I just see the rejection message:
y.y.y.y	x.x.x.x	INVITE sip:1234@x.x.x.x:5060
x.x.x.x	y.y.y.y	Status: 100 Trying
x.x.x.x	y.y.y.y	Status: 503 Service Unavailable
y.y.y.y	x.x.x.x	Request: ACK sip:1234@x.x.x.x:5060

(resent as retested with 0.7.1 & the addition of autocreatepeer=yes)

Thanks a lot,
Fran.

Fran Boon wrote:
> I have been compiling information on this configuration onto the Wiki:
> http://voip-info.org/wiki-Asterisk+cisco+FXO
> 
> I can call out to the PSTN just fine, but inbound calls all appear in my 
> default [bogon-calls] context, not in [pstn-incoming]
Hmmm.
As I understand it, the Cisco is not registred with Asterisk as a peer.

Could you please mail a SIP DEBUG output of an incoming INVITE from the
Cisco to Asterisk?

/O

http://www.intel.com/software/products/ipp/samples_table.htm#

Has anyone taken a look at the value of these sample libraries for use in 
Asterisk. 

cameron.

On Thu, 2004-01-15 at 18:27, Cameron Palmer wrote:
> http://www.intel.com/software/products/ipp/samples_table.htm#
> 
> Has anyone taken a look at the value of these sample libraries for use in 
> Asterisk. 
They might be useful, but you still have to license the codecs from the
patent holder, the licenses for those patents to not come with the
software