Hi<br><br>the asterisk version is 1.4.21.2<br><br>Here is the CLI<br><br> -- Executing [s@incomming:1] Set("Zap/4-1", "DB(lastcaller/zap4)=01942876818") in new stack<br> -- Executing [s@incomming:2] GotoIf("Zap/4-1", "0?s-spoof|1:") in new stack<br>
-- Executing [s@incomming:3] Ringing("Zap/4-1", "") in new stack<br> -- Executing [s@incomming:4] Set("Zap/4-1", "CDR(accountcode)=s") in new stack<br> -- Executing [s@incomming:5] Dial("Zap/4-1", "SIP/105|20|tT") in new stack<br>
-- Called 105<br><br><br>Sip.conf ( with somethings changed)<br>[gerneral]<br>externhost=<a href="http://a.host.to.setup.com">a.host.to.setup.com</a><br>localnet=<a href="http://10.1.1.0/255.255.255.0">10.1.1.0/255.255.255.0</a><br>
nat=yes<br><br><br>[105]<br>callerid=105<br>type=friend<br>username=105<br>host=dynamic<br>context=dialednum<br>secret=red<br>dtmfmode=rfc2833<br>disallow=all<br>allow=alaw<br>insecure=very<br>;mailbox=105@homR<br>qualify=no<br>
nat=yes<br><br><br><div class="gmail_quote">2009/8/7 Danny Nicholas <span dir="ltr"><<a href="mailto:danny@debsinc.com">danny@debsinc.com</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
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<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;">Show us your CLI output. I suspect that
you’re not getting a bridge and/or you’re timing out. Also
sip.conf and user.conf would be helpful as well as Asterisk release.</span></font></p>
<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;"> </span></font></p>
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<p><b><font size="2" face="Tahoma"><span style="font-size: 10pt; font-family: Tahoma; font-weight: bold;">From:</span></font></b><font size="2" face="Tahoma"><span style="font-size: 10pt; font-family: Tahoma;">
<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b><span style="font-weight: bold;">On Behalf Of </span></b>robert boardman<br>
<b><span style="font-weight: bold;">Sent:</span></b> Friday, August 07, 2009 9:01
AM<br>
<b><span style="font-weight: bold;">To:</span></b> Asterisk
Users Mailing List - Non-Commercial Discussion<br>
<b><span style="font-weight: bold;">Subject:</span></b> [asterisk-users] Linksys
SPA922</span></font></p>
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<p><font size="3" face="Times New Roman"><span style="font-size: 12pt;"> </span></font></p>
<p><font size="3" face="Times New Roman"><span style="font-size: 12pt;"><br>
Nearly got an SPA922 phone working behind a NAT,<br>
<br>
the phone registers, and I can dial out and have two way speech, <br>
<br>
on an incoming call the SPA922 rings<br>
<br>
I answer and the SPA922 shows "Anwsering" but never does and the far
end continues ringing until the voicemail answers, <br>
<br>
this then show as a disconnected call on the SPA922<br>
<br>
I'm on the lastest firmware 6.1.5(a)<br>
<br>
Thanks in advance for your help<br>
<br>
Robb</span></font></p>
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