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Dan Perik

2005-Apr-05 19:54 UTC

[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@gs1-FXO") in new
stack
    -- Called 5429411@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

Wai Wu

2005-Apr-06 06:46 UTC

head link

[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@ntm.org]
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 -> ) * -> 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@gs1-FXO") in new
stack
    -- Called 5429411@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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Andrejus Stavickis

2005-Apr-06 13:15 UTC

head link

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

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@lists.digium.com
[mailto:asterisk-users-bounces@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@ntm.org] 
	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 -> ) * -> 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@gs1-FXO")
in new 
	stack 
	    -- Called 5429411@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 
	_______________________________________________ 
	Asterisk-Users mailing list 
	Asterisk-Users@lists.digium.com 
	http://lists.digium.com/mailman/listinfo/asterisk-users 
	To UNSUBSCRIBE or update options visit: 
	   http://lists.digium.com/mailman/listinfo/asterisk-users 

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Dan Perik

2005-Apr-09 13:40 UTC

head link

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

Pardon my answering myself (and for the long post).  But I do have it
sort of working, and I come back with information on the GS HT-488, as
well as questions related to SIP / DTMF issues.

The GS HT-488 acts as a PSTN pass through device for 4 rings.  If the
phone attached to the FXS port hasn't picked up by 4 rings, it will by
default "answer", and you're at an internal (*) dial tone.  You
can also
configure the HT-488 to dial a specific extention, which it will then do
instead of dropping you at an internal dial tone.  From there you can
obviously do what ever you want with the call.  (It would be nice if you
could configure and/or disable the # rings before it switches over to
VoIP.  Maybe that will be something they will add to a firmware update
someday.) 

For dialing out, you set up an extention for the FXO port, and dial
that.  It will ring once, and then present you with the PSTN line, dial
tone and all.   From there you (should be) are able to dial out. 

Now, here is my problem and question.  Both the FXS and FXO ports are
set up to use SIP INFO for DTMF.   You would think that when you have
dialed the FXO port, and are at the PSTN dial tone, the HT-488 will
translate the SIP DTMF INFO passed through to the FXO port as audible
DTMF on the PSTN line.  This is not the case.  So I really can't make
outgoing calls yet.  Now, I can change the FXS line to send DTMF in
audio, which works, but I figure that sending DTMF in audio is not
ideal.  So I'm trying to "translate" the SIP DTMF INFO to DTMF
in-audio.  I've tried a few combinations of SipDTMFMode(inband) (trying
to do a DTMF style translation, I guess), and
Dial(SIP/gs1-FXO,10,D(<PSTNnumber>) ), but can't get it to work.  

Should I just suck it up and keep the FXS port using DTMF in-audio, or
is there a way to get SIP DTMF INFO translated to DTMF tones in audio in
the Dial settings for the FXO extension?

Thanks!
Dan

Dan Perik wrote:
>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.
><snip>
>  
>

asterisk users - Apr 2005 - Grandstream HandyTone-488, * -> FXO problems

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

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

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

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