[Asterisk-Users] Grandstream HandyTone-488, * -> FXO problems

[Andrejus Stavickis]

Well, the x100p is not always good either. If we forget that it only
support 600 ohm impedance, the proper example would be the problem i
have and not being able to overcome is tremendous echo on the VOIP phone
when i make a call to pstn. after 2 months of trying i had to quit using
it. 
 
The issue i have is that no matter what i do i never receive the output
from Asterisk saying somethig else, than "Echo Cancellation: 0 taps
unless TDM bridged, currently OFF" in responce to the command "zap show
channel 1". this is the ONLY card in the pc, does not share IRQ or IO.
It does not matter what i put in config files what echo cancellation i
use, it just never ever goes to something like "currently ON". I've read
a lot about echo problem on the pstn <-> voip but none of the solution
are working for me.


Sincerely,

--Andy
x6722

"Outsourcing is akin to making a skyscraper taller by taking material
from its lower floors."
--Byron Katz 

 


________________________________

	From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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 problems
	
	

	You 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:dan_perik at ntm.org] 
	Sent: Tuesday, April 05, 2005 10:55 PM 
	To: asterisk-users at 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 -> ) * -> FXO 

	In 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=g729 

	Here's a snippet from extensions.conf: 
	[gs1-fxo-out] 
	exten => _8.,1,Dial(SIP/${EXTEN:1}@gs1-FXO) 

	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/5429411 at gs1-FXO")
in new 
	stack 
	    -- Called 5429411 at gs1-FXO 
	    -- 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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