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<DIV><FONT face=Arial size=2>Does everything work fine now? I am still having
problems with </FONT><FONT face=Arial size=2>SayUnixTime. Voicemailmain2 works
though. The one simple AGI script I wrote doesn't do anything. Asterisk starts
playing and the grandstream just rings. Both work fine on other channels/sip
phones.</FONT></DIV>
<DIV><FONT face=Arial size=2> Thanks,</FONT></DIV>
<DIV><FONT face=Arial size=2> Will</FONT></DIV>
<DIV><FONT face=Arial size=2> </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<BLOCKQUOTE dir=ltr
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=wim.venneman@skynet.be href="mailto:wim.venneman@skynet.be">Wim
Venneman</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Friday, November 07, 2003 1:46
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Asterisk-Users] Grandstream
problem</DIV>
<DIV><BR></DIV>
<DIV><FONT face=Arial size=2>Thanks William,</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>Works fine now.</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>Wim</FONT></DIV>
<BLOCKQUOTE dir=ltr
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=wcarlson@w0ss.com href="mailto:wcarlson@w0ss.com">William
Carlson</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Thursday, November 06, 2003 9:43
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Asterisk-Users]
Grandstream problem</DIV>
<DIV><BR></DIV>
<DIV><FONT face=Arial size=2>try </FONT></DIV>
<DIV><FONT face=Arial size=2>disallow=all</FONT></DIV>
<DIV><FONT face=Arial size=2>allow=ulaw</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>under the general section of
sip.conf</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>that half fixes it for me calls between phones
work but talking to asterisk has some problems.</FONT></DIV>
<BLOCKQUOTE dir=ltr
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=wim.venneman@skynet.be href="mailto:wim.venneman@skynet.be">Wim
Venneman</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Thursday, November 06, 2003
2:29 PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> [Asterisk-Users] Grandstream
problem</DIV>
<DIV><BR></DIV>
<DIV><FONT face=Arial size=2>Hi,</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>I installed Asterisk an all works
fine exept for Grandstream.</FONT></DIV>
<DIV><FONT face=Arial size=2>When I call with a softphone (ex X-ten) to a
Grandstream (BudgetTone-100), I can make a conversation. = ok</FONT></DIV>
<DIV><FONT face=Arial size=2>When I call to a softphone with a
Grandstream I can pich up the call with the softphone but the Grandstream
keeps ringing like on the other site you didn't pick up the phone.(even if
you do so)</FONT></DIV>
<DIV><FONT face=Arial size=2>It's the same when I call between two
Grandstream phone's. Call from phone1 to phone 2, I pick up phone2 and
afther 3 seconds I get congestion tone from both phone's.</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>Info from command *CLI></FONT></DIV>
<DIV><FONT face=Arial size=2>-- Executing Dial("SIP/phone2-a030a",
"sip/phone1") in new stack</FONT></DIV>
<DIV><FONT face=Arial size=2>-- Called phone1</FONT></DIV>
<DIV><FONT face=Arial size=2>-- SIP/phone1-663a is ringing</FONT></DIV>
<DIV><FONT face=Arial size=2>-- SIP/phone1-663a answered
SIP/phone2-a030a</FONT></DIV>
<DIV><FONT face=Arial size=2>-- Attempting native bridge of
SIP/phone2-a030a and SIP/phone1-663a</FONT></DIV>
<DIV><FONT face=Arial size=2>== Spawn extension (sip, 1,1) exited
non-zero on 'SIP/phone2-a030a'</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>and I get congestion</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>Can anyone give me a direction to solve my
problem?</FONT></DIV>
<DIV><FONT face=Arial size=2>Thanks in advance,</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>Wim</FONT></DIV>
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