<?xml version="1.0" encoding="iso-8859-1" ?>
<?xml-stylesheet title="XSL_formatting" type="text/xsl" href="/images/lists/rssstyle2.xsl"?>
<rss version="2.0">
<channel>
<title>Cisco | VOIP</title>
<description>Mailing List Archive by Gossamer Threads</description>
<link>http://www.gossamer-threads.com/lists/cisco/voip/</link>
<language>en-us</language>
<copyright>(c) Gossamer Threads Inc. All rights reserved.</copyright>
<lastBuildDate>22 Nov  2009 16:44:28 -0800</lastBuildDate>
<ttl>120</ttl>
<image>
<title>Gossamer Threads | Cisco | VOIP</title>
<width>75</width>
<height>23</height>
<link>http://www.gossamer-threads.com/lists/cisco/voip/</link>
<url>http://www.gossamer-threads.com/images/lists/rss_logo.jpg</url>
</image>
<item>
<title>Re: IPIPGW</title>
<description>thanks for your answers I think Nicks answer is the one am looking for since i am doing H323-H323 no need for DSPs. On Sun, Nov 22, 2009 at 11:22 PM,</description>
<pubDate>22 Nov  2009 13:45:57 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121079</link>
</item><item>
<title>Re: IPIPGW</title>
<description>More importantly - if you&amp;#039;re not transcoding, you don&amp;#039;t need the PVDM at all for an IPIPGW. Be sure that we&amp;#039;re talking about a SIP-SIP, SIP-H323, or</description>
<pubDate>22 Nov  2009 13:22:30 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121077</link>
</item><item>
<title>Re: Configuring ASYNC NM in 2811</title>
<description>Found what I needed. http://www.cisco.com/en/US/products/hw/routers/ps282/products_tech_note09186a008035b051.shtml AccessServer#show line  Tty Lin</description>
<pubDate>22 Nov  2009 12:40:12 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121076</link>
</item><item>
<title>Re: IPIPGW</title>
<description>PVDM2-64 will handle 32 G729 medium complexity calls. PVDM2-32 will handle 16. PVDM2-16 will handle 8. You get the idea.  On Nov 22, 2009, at 12:2</description>
<pubDate>22 Nov  2009 12:37:35 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121074</link>
</item><item>
<title>Configuring ASYNC NM in 2811</title>
<description>I have an ASYNC 16A/S network module in a 2811 and I&amp;#039;m not understanding how to associate the &amp;#039;line&amp;#039; with the ip host command so I may use the 2811 to</description>
<pubDate>22 Nov  2009 12:34:38 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121075</link>
</item><item>
<title>Re: IPIPGW</title>
<description>Cisco DSP Calculator at:   http://www.cisco.com/cgi-bin/Support/DSP/cisco_prodsel.pl   .. is Your friend.      A   From: cisco-voip-bounces</description>
<pubDate>22 Nov  2009 12:30:03 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121078</link>
</item><item>
<title>Re: IPIPGW</title>
<description>One is enough. Greetings Marcin From: cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net] On Behalf Of Ali El Moussaoui S</description>
<pubDate>22 Nov  2009 12:16:07 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121072</link>
</item><item>
<title>IPIPGW</title>
<description>Hello Group, I am a silent watcher and this is my first post. I am planning a setup for an IPIPGW. The requirements state that the gteway must handle</description>
<pubDate>22 Nov  2009 11:25:50 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121071</link>
</item><item>
<title>security concern regarding cups</title>
<description>Dears,  We have a security concern regarding cups. When CUPS querying LDAP the integration account is sending the user name and password in plain te</description>
<pubDate>22 Nov  2009 09:28:51 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121068</link>
</item><item>
<title>Re: VWIC2-2MFT-T1/E1 for non-voice data ?</title>
<description>Don&amp;#039;t forget to set the card type as t1.. http://www.cisco.com/en/US/products/hw/modules/ps2617/products_configuration_example09186a008052c920.shtml</description>
<pubDate>21 Nov  2009 16:49:08 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121057</link>
</item><item>
<title>Re: Self-Signed Certificates on CallManager</title>
<description>Great explanation. thanks!  -----Original Message----- From: Sean Walberg Sent: Sat 11/21/2009 4:13 PM To: Carter, Bill Cc: cisco-voip@puck.nether.n</description>
<pubDate>21 Nov  2009 14:22:42 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121055</link>
</item><item>
<title>Re: Self-Signed Certificates on CallManager</title>
<description>An SSL certificate says that the signer has verified that the subject of the certificate is who they claim to be. So when I register a certificate for</description>
<pubDate>21 Nov  2009 14:13:24 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121054</link>
</item><item>
<title>CME Shared Line with SCCP and SIP</title>
<description>I want to configure an ephone-dn that is shared on two phones. Can one phone be SIP and the other SCCP? ____________________________________________</description>
<pubDate>21 Nov  2009 13:54:31 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121053</link>
</item><item>
<title>Self-Signed Certificates on CallManager</title>
<description>I don&amp;#039;t know much about certificates and CA....I understand web sites etc. that use SSL have registered their certificates with a CA. When we install</description>
<pubDate>21 Nov  2009 13:52:38 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121052</link>
</item><item>
<title>Outbound call with PRI VG - Delayed setup</title>
<description>Strange situation. Installed a new 2851 with a single PRI, 12.4.24T2.  When I call a PSTN number with an AA I hear &amp;quot;for calling XYC company&amp;quot;. If I c</description>
<pubDate>21 Nov  2009 13:37:54 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121051</link>
</item><item>
<title>Re: unity connection 7.1 disable tts</title>
<description>I suppose that, if you are adding a local user with all credentials ( First and Last name ) TTS will also read this, but only requirement when adding</description>
<pubDate>21 Nov  2009 10:01:30 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121048</link>
</item><item>
<title>Re: unity connection 7.1 disable tts</title>
<description>That&amp;#039;s interesting you say that. I don&amp;#039;t have a solution, but I thought I would add that I will have to check on our system for local accounts and whe</description>
<pubDate>21 Nov  2009 09:50:21 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121047</link>
</item><item>
<title>Re: 7937, CUPC and VT advantage camera</title>
<description>I also wanted to deploy the web cam and by advantage with the  conferance station but found out it was unsupported what is the best  reasonably pric</description>
<pubDate>21 Nov  2009 07:48:01 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121046</link>
</item><item>
<title>Re: CME with SCCP and SIP</title>
<description>Hi I&amp;#039;ve managed to put both systems (SIP ad SCCP ) on one router and make it work co-resident, but I didn&amp;#039;t try to use any extra user features, and m</description>
<pubDate>21 Nov  2009 05:13:52 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121045</link>
</item><item>
<title>unity connection 7.1 disable tts</title>
<description>Hi all Recently we have deployed CUCMBE 7 to our customer with english language, they didn&amp;#039;t wont to localize it ( international company where nativ</description>
<pubDate>21 Nov  2009 05:07:13 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121044</link>
</item><item>
<title>Re: 7937, CUPC and VT advantage camera</title>
<description>Hi Ryan, Thanks fro your reply. I need to have a video enabled at both ends so just to rely on CUPC softphone with Video, is there a way to stream l</description>
<pubDate>21 Nov  2009 03:23:20 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121043</link>
</item><item>
<title>Re: Caller ID changing to inbound dialpeer</title>
<description>What you need to do to fix this is the first dial-peer in your configuration just needs to handle the inbound calls with no destination-pattern on it.</description>
<pubDate>20 Nov  2009 23:30:10 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121041</link>
</item><item>
<title>Re: CME with SCCP and SIP</title>
<description>I don&amp;#039;t think this is going to work. You would have better luck with two different DNs (or at least a secondary number / alias for one of them) and u</description>
<pubDate>20 Nov  2009 20:06:30 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121037</link>
</item><item>
<title>Re: sub: Polycom rmx1000</title>
<description>The name of that site is www.vtctalk.com. thanks for the suggestion :-)  On Sat, Nov 21, 2009 at 9:04 AM, venkata sashank &amp;lt;reachsashank@gmail.com&amp;gt;wro</description>
<pubDate>20 Nov  2009 19:38:56 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121036</link>
</item><item>
<title>Re: sub: Polycom rmx1000</title>
<description>That site is on sale dude :-) On Sat, Nov 21, 2009 at 1:13 AM, Philip Walenta &amp;lt;pwalenta@wi.rr.com&amp;gt; wrote: &amp;gt; This really isn&#039;t the right forum to di</description>
<pubDate>20 Nov  2009 19:34:43 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121035</link>
</item>
</channel>
</rss>
