asterisk users - Aug 2004 - Problem dialing out to Free World Dialup

Hi

I am trying to make a call to a Free World Dialup number with the 
following call file:

Channel: SIP/476589@fwd.pulver.com
Callerid: Nagios
MaxRetries: 0
WaitTime: 30
Context: autodialout
Extension: s
Priority: 1

When I put the file in /var/spool/asterisk/outgoing/ directory, the 
X-Lite software phone installed on my other computer rings once then it 
says "Hung up" in the window of the phone.  This is output I get from 
the CLI:

-- Attempting call on SIP/476589@fwd.pulver.com:5060 for s@autodialout:1 
(Retry 1)
Aug 27 09:20:22 WARNING[1116957488]: chan_sip.c:497 retrans_pkt: Maximum 
retries exceeded on call 78679d5410a59d996fde8789069ff099@216.6.202.6 
for seqno 102 (Request)
Aug 27 09:20:22 NOTICE[1200884528]: pbx_spool.c:235 attempt_thread: Call 
failed to go through, reason 1
Aug 27 09:20:28 WARNING[1116957488]: chan_sip.c:497 retrans_pkt: Maximum 
retries exceeded on call 78679d5410a59d996fde8789069ff099@216.6.202.6 
for seqno 102 (Request)

Any help as to why it rings once then stops would be greatly 
appreciated, thanks.

-- 
Andrew Elchuk
Technical Associate
Cronus Technologies
248 - 111 Research Drive
Saskatoon, SK  S7N 2X8
Tel: (306) 652-5798 ext. 112
Fax: (306) 652-5799
Toll Free: 1-877-655-5798
http://www.cronustech.com

On Fri, 27 Aug 2004, Andrew Elchuk wrote:
> -- Attempting call on SIP/476589@fwd.pulver.com:5060 for s@autodialout:1 
> (Retry 1)
> Aug 27 09:20:22 WARNING[1116957488]: chan_sip.c:497 retrans_pkt: Maximum 
> retries exceeded on call 78679d5410a59d996fde8789069ff099@216.6.202.6 
> for seqno 102 (Request)
> Aug 27 09:20:22 NOTICE[1200884528]: pbx_spool.c:235 attempt_thread: Call 
> failed to go through, reason 1
> Aug 27 09:20:28 WARNING[1116957488]: chan_sip.c:497 retrans_pkt: Maximum 
> retries exceeded on call 78679d5410a59d996fde8789069ff099@216.6.202.6 
> for seqno 102 (Request)
> 
> Any help as to why it rings once then stops would be greatly 
> appreciated, thanks.

Usually this sort of stuff is a symptom of connectivity/NAT problems 
between the two SIP endpoints.

Steve