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<DIV dir=ltr align=left><SPAN class=991150420-06042005><FONT face=Arial
color=#0000ff size=2>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. </FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=991150420-06042005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=991150420-06042005><FONT face=Arial
color=#0000ff size=2>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.</FONT></SPAN></DIV><!-- Converted from text/plain format --><BR>
<P><FONT size=2>Sincerely,<BR><BR>--Andy<BR>x6722<BR><BR>"Outsourcing is akin to
making a skyscraper taller by taking material from its lower floors."<BR>--Byron
Katz</FONT> </P>
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<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>Wai
Wu<BR><B>Sent:</B> Wednesday, April 06, 2005 9:47 AM<BR><B>To:</B> 'Asterisk
Users Mailing List - Non-Commercial Discussion'<BR><B>Subject:</B> RE:
[Asterisk-Users] Grandstream HandyTone-488, * -> FXO
problems<BR></FONT><BR></DIV>
<DIV></DIV>
<P><FONT size=2>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.</FONT></P>
<P><FONT size=2>-----Original Message-----</FONT> <BR><FONT size=2>From: Dan
Perik [<A href="mailto:dan_perik@ntm.org">mailto:dan_perik@ntm.org</A>]</FONT>
<BR><FONT size=2>Sent: Tuesday, April 05, 2005 10:55 PM</FONT> <BR><FONT
size=2>To: asterisk-users@lists.digium.com</FONT> <BR><FONT size=2>Subject:
[Asterisk-Users] Grandstream HandyTone-488, * -> FXO problems</FONT>
</P><BR>
<P><FONT size=2>I just got my shiny new Grandstream HandyTone-488 today.
My goal is to</FONT> <BR><FONT size=2>use it to allow incoming/outgoing calls
to PSTN using my normal ole'</FONT> <BR><FONT size=2>phone as usual. I
will be switching over to using BroadVoice as my main</FONT> <BR><FONT
size=2>phone #, but want that to be as seemless of a switchover as
possible</FONT> <BR><FONT size=2>(for the wife and kids, and for people
needing to call us).</FONT> </P>
<P><FONT size=2>I've got the following working:</FONT> </P>
<P><FONT size=2>FXS -> * ( and then -> BroadVoice )</FONT> <BR><FONT
size=2>( BroadVoice -> ) * -> FXS</FONT> <BR><FONT size=2>FXO -> * (
and then -> FXS )</FONT> </P>
<P><FONT size=2>I don't have this working:</FONT> <BR><FONT size=2>( FXS ->
) * -> FXO</FONT> </P>
<P><FONT size=2>In other words, I can't seem to call out on my PSTN line from
Asterisk.</FONT> </P>
<P><FONT size=2>Here's a snippet from sip.conf:</FONT> <BR><FONT
size=2>[gs1-FXO]</FONT> <BR><FONT size=2>type=friend</FONT> <BR><FONT
size=2>context=default</FONT> <BR><FONT size=2>host=dynamic</FONT> <BR><FONT
size=2>username=gs1-FXO</FONT> <BR><FONT size=2>secret=<mysecret></FONT>
<BR><FONT size=2>nat=no</FONT> <BR><FONT size=2>canreinvite=yes</FONT>
<BR><FONT size=2>dtmfmode=info</FONT> <BR><FONT size=2>incominglimit=1</FONT>
<BR><FONT size=2>disallow=all</FONT> <BR><FONT size=2>allow=ulaw</FONT>
<BR><FONT size=2>allow=alaw</FONT> <BR><FONT size=2>allow=g723.1</FONT>
<BR><FONT size=2>allow=g729</FONT> </P>
<P><FONT size=2>Here's a snippet from extensions.conf:</FONT> <BR><FONT
size=2>[gs1-fxo-out]</FONT> <BR><FONT size=2>exten =>
_8.,1,Dial(SIP/${EXTEN:1}@gs1-FXO)</FONT> </P>
<P><FONT size=2>So when I dial, say 85429411, I would expect it to dial
5429411 out on</FONT> <BR><FONT size=2>the PSTN line. I end up not getting any
tone or other audio out of the</FONT> <BR><FONT size=2>handset. But,
using another phone directly connected to the PSTN, I</FONT> <BR><FONT
size=2>find that the Grandstream has taken the line off hook, but not
dialed</FONT> <BR><FONT size=2>any digits. I get this in my * log when I
dial 85429411.</FONT> </P>
<P><FONT size=2> -- Executing Dial("SIP/gs1-FXS-9041",
"SIP/5429411@gs1-FXO") in new</FONT> <BR><FONT size=2>stack</FONT> <BR><FONT
size=2> -- Called 5429411@gs1-FXO</FONT> <BR><FONT
size=2> -- SIP/gs1-FXO-877b is ringing</FONT> <BR><FONT
size=2> -- SIP/gs1-FXO-877b answered SIP/gs1-FXS-9041</FONT>
<BR><FONT size=2> -- Attempting native bridge of
SIP/gs1-FXS-9041 and SIP/gs1-FXO-877b</FONT> <BR><FONT size=2> == Spawn
extension (outgoing-ok, 85429411, 1) exited non-zero on</FONT> <BR><FONT
size=2>'SIP/gs1-FXS-9041'</FONT> </P>
<P><FONT size=2>I know the Handy-Tone 488 is a new device, so there may be
some quirks</FONT> <BR><FONT size=2>to it. But I would think it _should_
work.</FONT> </P>
<P><FONT size=2>Any suggestions?</FONT> </P>
<P><FONT size=2>Thanks!</FONT> <BR><FONT size=2>Dan</FONT> <BR><FONT
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