asterisk users - Mar 2005 - Asterisk Quicknet FWD Problem - no path to translate from Phone/phone0 to SIP

I cant seem to be able to figure this out.  As much as I can tell it is
a codec problem.

I can dial out to 612@fwd.pulver.com  and the "Call Me" test there
rings
my phone.  However when the callee endpoint answers, there is a failure
to translate:

Outgoing Call for 612
612 is not a local user
    -- Called 612@fwdpulvercom
No path to translate from SIP/fwdpulvercom-dd5a(2) to Phone/phone0(1)
(Provisional) Stopping retransmission (but retaining packet) on
'1bf802c37038448b795a4dc8300e0627@12.218.223.74' Request 102: Found
(Provisional) Stopping retransmission (but retaining packet) on
'1bf802c37038448b795a4dc8300e0627@12.218.223.74' Request 102: Found
    -- SIP/fwdpulvercom-dd5a is ringing
Unable to handle indication 3 for 'Phone/phone0'
Scheduled a registration timeout # 100
Acked pending invite 102
Stopping retransmission on
'1bf802c37038448b795a4dc8300e0627@12.218.223.74' of Request 102: Found
build_route: Record-Route hop:
<sip:612@69.90.155.70;ftag=as3d6e380d;lr=on>
build_route: Contact hop: <sip:612@69.90.168.13:5028>
    -- SIP/fwdpulvercom-dd5a answered Phone/phone0
No path to translate from Phone/phone0(1) to SIP/fwdpulvercom-dd5a(2)
Had to drop call because I couldn't make Phone/phone0 compatible with
SIP/fwdpulvercom-dd5a
update_user_counter(612) - decrement outUse counter

I have a Quicknet Lite ISA card.

my phone.conf contains:

mode=dialtone
;format=slinear
format=g723.1
echocancel=medium
silencesupression=yes
device => /dev/phone0

my sip.conf contains:

context=default                 ; Default context for incoming calls
port=5060                       ; UDP Port to bind to (SIP standard port
is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds
to all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound
calls
disallow=all
allow=gsm
allow=ulaw
allow=alaw
maxexpirey=180
defaultexpirey=160
tos=reliability
 
register => 6XXXX:mypasswd@fwd.pulver.com
 

[fwdout]
type=friend
username=6XXXX
secret=mypasswd
host=fwd.pulver.com
 

[fwdin]
type=peer
host=fwd.pulver.com
context=default
nat=yes
canreinvite=no


my extensions.conf contains:

[globals]
CONSOLE=Phone/phone0
 

[default]
exten => _XXX,1,Dial(SIP/${EXTEN}@fwdout)
exten => s,1,Dial(Phone/phone0)


Is it possible to call FWD using the Quicknet card?

Thanks for any help

cmisip wrote:
> No path to translate from SIP/fwdpulvercom-dd5a(2) to Phone/phone0(1)
I don't know why the above message is printing codec numnbers, rather 
than names. *shrug*

"show codecs" will tell you what codec number are what codec name.

It appears that your Phone/phone0 is using G723.1.  Looks likes one of 
the newbie problems of using allow=all or bandwidth=low.  DON'T DO THAT!

Use disallow=all and then allow= lines for the one or more codecs that 
you actually want to use.

Asterisk does not fully support G723.1.  "fully" means
"transcode".

--Eric


-- 
Always do right. This will gratify some people and astonish the rest.
Mark Twain