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

[Andrejus Stavickis]

Title: RE: [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: [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 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:[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 -> ) * -> 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:[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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