Re: [Asterisk-Users] Asterisk Quicknet FWD Problem - no path to translate from Phone/phone0 to SIP

[Eric Wieling]

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
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[Asterisk-Users] Asterisk Quicknet FWD Problem - no path to translate from Phone/phone0 to SIP

[cmisip]

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 [EMAIL PROTECTED]  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 [EMAIL PROTECTED]
No path to translate from SIP/fwdpulvercom-dd5a(2) to Phone/phone0(1)
(Provisional) Stopping retransmission (but retaining packet) on
'[EMAIL PROTECTED]' Request 102: Found
(Provisional) Stopping retransmission (but retaining packet) on
'[EMAIL PROTECTED]' 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
'[EMAIL PROTECTED]' of Request 102: Found
build_route: Record-Route hop:

build_route: Contact hop: 
-- 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 => 6:[EMAIL PROTECTED]
 

[fwdout]
type=friend
username=6
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/[EMAIL PROTECTED])
exten => s,1,Dial(Phone/phone0)


Is it possible to call FWD using the Quicknet card?

Thanks for any help



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