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<DIV dir=ltr align=left><SPAN class=627135421-30052006><FONT face=Verdana
size=2>I have:</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=627135421-30052006><FONT face=Verdana
size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=627135421-30052006><FONT face=Verdana
size=2>disallow=all<BR>allow=ulaw<BR>allow=alaw<BR></FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=627135421-30052006><FONT face=Verdana
size=2>I don't have the allow=gsm. What is that for?</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=627135421-30052006><FONT face=Verdana
size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><FONT face=Verdana><FONT size=2><SPAN
class=627135421-30052006>George</SPAN><SPAN
class=627135421-30052006></DIV></SPAN></FONT></FONT><BR>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR tabIndex=-1>
<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>Marco
Mouta<BR><B>Sent:</B> Tuesday, May 30, 2006 4:17 PM<BR><B>To:</B> Asterisk Users
Mailing List - Non-Commercial Discussion<BR><B>Subject:</B> Re: [Asterisk-Users]
No sound?? HELP<BR></FONT><BR></DIV>
<DIV></DIV>check [general] section of your
/etc/asterisk/sip.conf<BR><BR>disallow=all<BR>allow=alaw<BR>allow=ulaw<BR>allow=gsm
<BR><BR>This codecs depends on of your SIP provider as well as activation in
your SIPphone<BR><BR>
<DIV><SPAN class=gmail_quote>On 5/30/06, <B class=gmail_sendername>George A.
Roberts IV</B> <<A
href="mailto:groberts@interjuncture.com">groberts@interjuncture.com</A>>
wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">
<DIV>
<DIV>
<DIV><SPAN><FONT face=Verdana size=2>I just put in a new <A
onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:Asterisk@Home" target=_blank>Asterisk@Home</A> 2.8 system.
Trunk is connected via SIP to ViaTalk.</FONT></SPAN></DIV>
<DIV><SPAN><FONT face=Verdana size=2></FONT></SPAN> </DIV>
<DIV><SPAN><FONT face=Verdana size=2>I had an older <A
onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:Asterisk@Home" target=_blank>Asterisk@Home</A> system up and
running that was working fine and I replicated settings over to the new
box. When I call 7777 from an internal SIP extension I can hear the IVR
menu just fine. However, when I call from a POTS phone to our number and
it comes in via ViaTalk over SIP the call connects but I do not get any
sound. I'm sure it's a setting or something I missed, but I'm not sure
what it is. Anyone have any ideas?</FONT></SPAN></DIV>
<DIV><SPAN><FONT face=Verdana size=2></FONT></SPAN> </DIV>
<DIV><SPAN><FONT face=Verdana
size=2>George</FONT></SPAN></DIV></DIV></DIV><BR>_______________________________________________<BR>--Bandwidth
and Colocation provided by <A
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