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