Sincerely,
--Andy
x6722
"Outsourcing is akin to
making a skyscraper taller by taking material from its lower floors."
--Byron
Katz
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu
Sent: Wednesday, April 06, 2005 9:47 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Grandstream HandyTone-488, * -> FXO problemsYou can stop trying. They still have problem with the firmware concerning the FXO port. If you really want to make a call from * out the PSTN, I suggest you to get a x100p. They are selling it on ebay for $6.99, and I have 4 of those in my * box.
-----Original Message-----
From: Dan Perik [mailto:[EMAIL PROTECTED]]
Sent: Tuesday, April 05, 2005 10:55 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Grandstream HandyTone-488, * -> FXO problems
I just got my shiny new Grandstream HandyTone-488 today. My goal is to
use it to allow incoming/outgoing calls to PSTN using my normal ole'
phone as usual. I will be switching over to using BroadVoice as my main
phone #, but want that to be as seemless of a switchover as possible
(for the wife and kids, and for people needing to call us).I've got the following working:
FXS -> * ( and then -> BroadVoice )
( BroadVoice -> ) * -> FXS
FXO -> * ( and then -> FXS )I don't have this working:
( FXS -> ) * -> FXOIn other words, I can't seem to call out on my PSTN line from Asterisk.
Here's a snippet from sip.conf:
[gs1-FXO]
type=friend
context=default
host=dynamic
username=gs1-FXO
secret=<mysecret>
nat=no
canreinvite=yes
dtmfmode=info
incominglimit=1
disallow=all
allow=ulaw
allow=alaw
allow=g723.1
allow=g729Here's a snippet from extensions.conf:
[gs1-fxo-out]
exten => _8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])So when I dial, say 85429411, I would expect it to dial 5429411 out on
the PSTN line. I end up not getting any tone or other audio out of the
handset. But, using another phone directly connected to the PSTN, I
find that the Grandstream has taken the line off hook, but not dialed
any digits. I get this in my * log when I dial 85429411.-- Executing Dial("SIP/gs1-FXS-9041", "SIP/[EMAIL PROTECTED]") in new
stack
-- Called [EMAIL PROTECTED]
-- SIP/gs1-FXO-877b is ringing
-- SIP/gs1-FXO-877b answered SIP/gs1-FXS-9041
-- Attempting native bridge of SIP/gs1-FXS-9041 and SIP/gs1-FXO-877b
== Spawn extension (outgoing-ok, 85429411, 1) exited non-zero on
'SIP/gs1-FXS-9041'I know the Handy-Tone 488 is a new device, so there may be some quirks
to it. But I would think it _should_ work.Any suggestions?
Thanks!
Dan
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