asterisk users - Nov 2004 - Problems with CISCO, SIP and Asterisk

Hello People,

I'm newbie in * 1.0.1, running a Linux 2.6.7 in a Debian Sarge,
and this is my situation:

 +------------+             +-------------+
 | Sip Server |-------------|CISCO PSTN GW|
 +------------+             +-------------+
          \                       ||
           \                      ||
            \ +----------+        ||
              | Asterisk |========              +----------+

The * and CISCO are authorized by the sip.
The call is coming from PTSN via CISCO to *.
I'm seen the sip debug in the * console, I see that * send
"demo-enterkeywords"
but in this moment the CISCO hangup the connexion.
In the SER server the CISCO send BYE to SER but nothing to *.

If the call is coming from the x-lite authorized by the same SIP server, there
are no problems.

The peer configuration in the CISCO 5XXX is the same that:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cisco%20FXO

My sip.conf is:

[general]
disallow=all
allow=ulaw
port=5060
bindaddr=0.0.0.0
context=from-sip
autocreatepeer=yes

register => XXXXXXXXXXXX:XXXX:XXXXXX@sip.example.com

[666]
context=local-phones
type=friend
user=666
secret=666
auth=md5
host=dynamic
defaultip=192.168.10.167
reinvite=no
canreinvite=no
qualify=1000
callerid="diavolo" <666>
disallow=all
allow=ulaw

My extensions.conf is:

[default]
include => mainmenu
include => lan-phones

[mainmenu]
exten => s,1,Answer
exten => s,2,SetMusicOnHold(default)
exten => s,3,DigitTimeout,15
exten => s,4,ResponseTimeout,35
exten => s,5,Background(demo-enterkeywords)

exten => 1,1,VoicemailMain()
exten => 1,2,Hangup

exten => 2,1,Playback(demo-echotest)
exten => 2,2,Echo
exten => 2,3,Playback(demo-echodone)
exten => 2,4,Goto(mainmenu,s,6)

exten => 3,1,MusicOnHold(default)
exten => 3,2,Goto(mainmenu,s,6)

exten => 4,1,Playback(demo-thanks)
exten => 4,2,Hangup

exten => t,1,Goto(4,1)
exten => i,1,Playback(invalid) 
          
[lan-phones]
exten => 666,1,Dial(SIP/666,20)
exten => 666,2,Voicemail(u666)

[from-sip]
include => mainmenu
include => lan-phones

Asterisk show while running:

linux*CLI> sip show registry
Host                            Username       Refresh State               
sip.example.com:5060            XXXXXXXXXXXX       105 Registered          
linux*CLI> 

could you give me some clue about it? 
Thanks in advance

Charlie
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Hi Carlos,
have you any success in this case.
It's interesting because we trying to connect Cisco 1760 ISDN gateway to
Asterisk: no luck. Immediate disconnect comes after connect messages on
"debug isdn q.931" with cause value "16" (normal call
clearing).
Difference is that our installation is without SIP proxy server.


I fell that without sip proxy our installation won?t work.


Vaidotas