<?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 17:19:49 -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><item>
<title>CME with SCCP and SIP</title>
<description>My goal is to a ring a SIP and SCCP phone in cme 7.1 at the same time. I know if they both have the same extension, just the sccp phone rings. I was t</description>
<pubDate>20 Nov  2009 19:11:12 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121034</link>
</item><item>
<title>UCCx 5.0 agent from one queue to another</title>
<description>I&amp;#039;m trying to move an agent from one queue to another. assigned the skill, changed the team, the queue is skill based but have also changed the resou</description>
<pubDate>20 Nov  2009 13:49:57 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121033</link>
</item><item>
<title>Re: SIP Issue</title>
<description>Hi Nick, Thx for the reply. You are correct this is exactly what it turned out to be. I would agree that it&amp;#039;s not a bug, per se. However, Cisco TAC c</description>
<pubDate>20 Nov  2009 13:30:01 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121032</link>
</item><item>
<title>Re: Cisco 2811 SIP trunk and FXS port</title>
<description>There are probably 20 things that could cause this, only few of which may be inferred from the configuration. &amp;#039;debug ccsip messages&amp;#039; is much better f</description>
<pubDate>20 Nov  2009 12:54:07 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121031</link>
</item><item>
<title>Re: SIP Issue</title>
<description>What it sounds like is this: Your provider doesn&amp;#039;t support SIP video. This was a feature added into 12.4(20)T which is why things worked before that</description>
<pubDate>20 Nov  2009 12:49:51 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121030</link>
</item><item>
<title>Re: GW POTS to PSTN SIP Trunk</title>
<description>It&amp;#039;s not going to be a lot of configuration. If you&amp;#039;ve already got calls routing, you only need one command: voice service voip  fax protocol pass-</description>
<pubDate>20 Nov  2009 12:43:52 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121029</link>
</item><item>
<title>Re: Number Transformation on Inbound Calls</title>
<description>The best way to do this is with + dialing. Look at the dialplan section of the CM 7 SRND. From: cisco-voip-bounces@puck.nether.net [mailto:cisco-voi</description>
<pubDate>20 Nov  2009 12:20:29 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121028</link>
</item><item>
<title>Number Transformation on Inbound Calls</title>
<description>All - I am looking for an easy way to prefix incoming calls with a 91 so that when a user tries to use the &amp;quot;Missed Calls&amp;quot; or &amp;quot;Received Calls&amp;quot; functio</description>
<pubDate>20 Nov  2009 12:17:04 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121027</link>
</item><item>
<title>Re: CUCMBE 7.1 authentication rules for CUC</title>
<description>OK, it looks like you need to apply that credential policy to each user in CUCM... On Fri, Nov 20, 2009 at 1:15 PM, Jonathan Charles &amp;lt;jonvoip@gmail.c</description>
<pubDate>20 Nov  2009 11:52:01 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121026</link>
</item><item>
<title>Re: sub: Polycom rmx1000</title>
<description>This really isn&amp;#039;t the right forum to discuss completely non-Cisco related devices.   If you want VC help I recommend checking here:   http://www.v</description>
<pubDate>20 Nov  2009 11:43:50 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121025</link>
</item><item>
<title>CUCMBE 7.1 authentication rules for CUC</title>
<description>Trying to force a 4 digit PIN, but the authentication policy is missing from CUC on CUCMBE... I changed the credential policy on CUCM, but CUC is sti</description>
<pubDate>20 Nov  2009 11:15:21 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121024</link>
</item><item>
<title>Re: Caller ID changing to inbound dialpeer</title>
<description>Interesting. I haven&amp;#039;t seen that done before. But if you have done this all along, then I don&amp;#039;t know if that would be the issue. Might be worth a s</description>
<pubDate>20 Nov  2009 10:47:23 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121023</link>
</item><item>
<title>Unity 4.1 on box messaging</title>
<description>Its been a while since I had to check this but of course the situation arose on a Friday. I have a user on a Unity 4.1 voicemail only system. They a</description>
<pubDate>20 Nov  2009 10:25:05 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121022</link>
</item><item>
<title>Re: Caller ID changing to inbound dialpeer</title>
<description>I have always configured the 911 dial peers as my inbound call leg also. Just saves some dialpeers. I guess I can remove all the inbound call leg st</description>
<pubDate>20 Nov  2009 10:22:01 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121021</link>
</item><item>
<title>Re: UCCX - CAD Agent Login Timeout</title>
<description>Just to let you know that the combination of the 3 workarounds is a success: CAD workaround (CSCta49259) UCCX workaround (CSCsv31620) CCM workaround</description>
<pubDate>20 Nov  2009 09:23:32 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121019</link>
</item><item>
<title>Re: Certificates question</title>
<description>Open the Certificate Authority MMC, and look under &amp;#039;certificate templates&amp;#039; and see if it is listed there.    From: Tim Reimers [mailto:treimers@as</description>
<pubDate>20 Nov  2009 09:18:19 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121018</link>
</item><item>
<title>Re: Certificates question</title>
<description>I&amp;#039;m the one that installed the CA services on that box, eg, I am the CA administrator I never disabled any of the certificate profiles. this is how CA</description>
<pubDate>20 Nov  2009 09:12:00 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121017</link>
</item><item>
<title>Re: Certificates question</title>
<description>If the web server certificate profile is not listed, then it was removed by your CA administrator. You will either need to do this via command line o</description>
<pubDate>20 Nov  2009 09:09:25 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121016</link>
</item><item>
<title>Re: Certificates question</title>
<description>I just noticed in this article http://www.cisco.com/en/US/products/sw/secursw/ps2086/products_configura tion_example09186a00804721c3.shtml  that Cisc</description>
<pubDate>20 Nov  2009 09:06:59 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121015</link>
</item><item>
<title>Re: Certificates question</title>
<description>I think you might need a server enterprise edition server running as your CA to generate the right type of cert Matthew Loraditch 1965 Greenspring Dr</description>
<pubDate>20 Nov  2009 09:04:14 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121014</link>
</item><item>
<title>Certificates question</title>
<description>Hi everyone -  I&amp;#039;m having trouble getting a certificate installed for our UCM, using a cert supplied by our domain CA server (not a public CA server)</description>
<pubDate>20 Nov  2009 08:55:43 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121013</link>
</item><item>
<title>Re: Caller ID changing to inbound dialpeer</title>
<description>why do you have service ani_filter on an outbound 9911 dial-peer? why do you have answer-address .... on a 9911 dial-peer? why do you have direct-in</description>
<pubDate>20 Nov  2009 08:38:00 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121012</link>
</item><item>
<title>Re: Failed Unity TUI access</title>
<description>Thx for the reply Jason. We tried that a short while ago... same issue. We found this error within micro traces: SG_API_FAILURE IMsgStore::OpenEntry h</description>
<pubDate>20 Nov  2009 08:06:54 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121007</link>
</item><item>
<title>Caller ID changing to inbound dialpeer</title>
<description>I just switched from a CMM blade to an ISR VGW. Setup everything pretty much the same. IOS is 12.4.24t2 but if a call comes in with OUT caller ID,</description>
<pubDate>20 Nov  2009 08:06:50 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121006</link>
</item><item>
<title>Re: sub: Polycom rmx1000</title>
<description>Thanks for the response. I have rebooted it thrice, still the same issue. One more thing i see that it doesnt have any voice or video ports. could tha</description>
<pubDate>20 Nov  2009 08:02:43 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121005</link>
</item><item>
<title>Re: unsubscribe</title>
<description>Self Service: https://puck.nether.net/mailman/listinfo/cisco-voip &amp;lt;https://puck.nether.net/mailman/listinfo/cisco-voip&amp;gt;RSS feeds are local you your</description>
<pubDate>20 Nov  2009 07:55:55 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121004</link>
</item><item>
<title>Re: sub: Polycom rmx1000</title>
<description>I don&amp;#039;t have these model of units, but I have found Polycom equipment to be very odd at times. Usually just rebooting HDX units takes care of most pr</description>
<pubDate>20 Nov  2009 07:49:59 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121003</link>
</item><item>
<title>sub: Polycom rmx1000</title>
<description>Dear All, Has any one worked on Polycom rmx1000 video conferencing unit. I just have some problem with it. I have a rmx 1000 and 2 v700s for installa</description>
<pubDate>20 Nov  2009 07:42:27 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121002</link>
</item><item>
<title>unsubscribe</title>
<description>Can you please remove my account from this RSS feed? -Thank you Travis Bugh Network Wireless Associate CDW-Advanced Technology Services Travis.Bugh@</description>
<pubDate>20 Nov  2009 07:37:24 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121001</link>
</item><item>
<title>Re: Failed Unity TUI access</title>
<description>Did TAC run the the Bunny Killer (Remove Subscriber Properties Tool) and try to create a new subscriber afterwards?     You can use COBRAS to bac</description>
<pubDate>20 Nov  2009 07:33:58 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/121000</link>
</item><item>
<title>Re: Tandberg Discussion List?</title>
<description>Jeffrey Ollie wrote: &amp;gt; So I know this is semi-offtopic, but does anyone know of a discussion &amp;gt; list similar to this one except it&amp;#039;s about Tandberg VoI</description>
<pubDate>20 Nov  2009 07:04:55 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120999</link>
</item><item>
<title>GW POTS to PSTN SIP Trunk</title>
<description>Hi,    Does anyone have a sample configuration of a POTS FXS Fax going to a PSTN SIP trunk. The SIP trunk provider is Verizon and based on their d</description>
<pubDate>20 Nov  2009 06:55:32 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120998</link>
</item><item>
<title>Tandberg Discussion List?</title>
<description>So I know this is semi-offtopic, but does anyone know of a discussion list similar to this one except it&amp;#039;s about Tandberg VoIP equipment? I&amp;#039;m trying t</description>
<pubDate>20 Nov  2009 06:42:30 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120997</link>
</item><item>
<title>Re: CUPS and UCM Application Dial Rules</title>
<description>Presently in CUPS, if I add an Application Dial Rule to UCM I have to restart the CUP SIP Proxy service in order for the changes to take affect. Does</description>
<pubDate>20 Nov  2009 06:41:49 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120996</link>
</item><item>
<title>Failed Unity TUI access</title>
<description>Folks: We upgraded a client from Unity 4.x to 7.x over the weekend. There&amp;#039;s 2,000 subscribers on this Unity cluster and everyone but one user is enco</description>
<pubDate>20 Nov  2009 06:32:04 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120995</link>
</item><item>
<title>Re: Masking Caller ID and Translation Patterns</title>
<description>Yeah they do say they can only present as one DDI (DID same thing). Not sure on the &amp;quot;something for their maintainers to resolve&amp;quot; though. Good luck w</description>
<pubDate>20 Nov  2009 05:06:35 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120989</link>
</item><item>
<title>Re: Masking Caller ID and Translation Patterns</title>
<description>Thanks Chris. This is the response I received. I don&amp;#039;t quite understand so I&amp;#039;ll probably have to call them. &amp;quot;The only services we offer are presenta</description>
<pubDate>20 Nov  2009 05:01:44 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120988</link>
</item><item>
<title>Re: Masking Caller ID and Translation Patterns</title>
<description>If its BT then they sometimes Mask all outgoing calls to the Main DID you own. Its takes some time but they do eventually admit this and can change i</description>
<pubDate>20 Nov  2009 04:34:43 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120987</link>
</item><item>
<title>Re: SIP Issue</title>
<description>After looking at the SDP header I found the difference between a good and bad call was video capabilities. Seems there is a TAC bug regarding the issu</description>
<pubDate>19 Nov  2009 22:53:36 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120978</link>
</item><item>
<title>Re: 7937, CUPC and VT advantage camera</title>
<description>CUPC can do video by itself in softphone mode, no CUVA needed. CUVA does require the PC to be connected directly behind the phone because the app has</description>
<pubDate>19 Nov  2009 18:39:27 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120972</link>
</item><item>
<title>7937, CUPC and VT advantage camera</title>
<description>Hi, I have UCM 7.02a, a CUP server version 7.0(4), and 7937 conference phone and I need to deploy video call to other Cisco 7942G phone with VT camer</description>
<pubDate>19 Nov  2009 16:40:54 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120971</link>
</item><item>
<title>TAPS will not load UCCX 7.0.1SR4 and UCM 7.1.3</title>
<description>UCCX 7.0.1SR4 and UCM 7.1.3 installed the TAPS plugin from UCM.  Has anyone got this working? Using the TAPS.AAR to load the applicaiton. The AAR ap</description>
<pubDate>19 Nov  2009 14:30:31 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120969</link>
</item><item>
<title>voip integration software</title>
<description>Has anyone used voipintegration software to deploy IP Phone background image in large environment? Iâ€™ve tested with demo copy and it worked fine de</description>
<pubDate>19 Nov  2009 13:18:03 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120970</link>
</item><item>
<title>Re: Masking Caller ID and Translation Patterns</title>
<description>That&amp;#039;s definitely good to know. It&amp;#039;s a UK Telco so who knows what rules they have. I&amp;#039;ve written the email but haven&amp;#039;t heard back. I&amp;#039;ll let you know</description>
<pubDate>19 Nov  2009 11:40:46 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120965</link>
</item><item>
<title>Re: Fwd: Convert Wav to RAW file for ringtones</title>
<description>Anyone know where this setting is in Audacity v1.3.9? Vincent Loschiavo Director of Consulting Services DATACORP 8200 N.W. 41st Street, Suite 130 Mia</description>
<pubDate>19 Nov  2009 11:18:03 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120964</link>
</item><item>
<title>Re: cdrs</title>
<description>calls with &amp;#039;abnormal disconnects&amp;#039; should always be logged. /Wes On Thursday, November 19, 2009 1:46:41 PM, Scott Voll &amp;lt;svoll.voip@gmail.com&amp;gt; wrote:</description>
<pubDate>19 Nov  2009 11:01:00 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120963</link>
</item><item>
<title>Re: cdrs</title>
<description>Dial it yourself, and you&amp;#039;ll see the same type of CDR from your phone.  What is the disconnect cause on that CDR? Unallocated/unassigned number shou</description>
<pubDate>19 Nov  2009 10:53:16 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120962</link>
</item><item>
<title>Re: cdrs</title>
<description>we have 0 duration setup. and see 0 duration calls. I was just trying to find out why the blank ones. sounds like per your email below that it was</description>
<pubDate>19 Nov  2009 10:46:41 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120961</link>
</item><item>
<title>Re: Masking Caller ID and Translation Patterns</title>
<description>We had to sign paperwork with Verizon promising &amp;quot;to solemnly swear&amp;quot; not to commit fraud, or no good.  (Quotes added for comic relief) On Wed, Nov 18</description>
<pubDate>19 Nov  2009 10:41:35 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120960</link>
</item><item>
<title>Re: Fwd: Convert Wav to RAW file for ringtones</title>
<description>Thanks. -----Original Message----- From: cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net] On Behalf Of Peter Pauly Sent</description>
<pubDate>19 Nov  2009 10:37:09 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120959</link>
</item><item>
<title>Re: cdrs</title>
<description>Because the call never connected. Even with 0 duration CDRs disabled any call that results in reorder to a phone will still generate a CDR. -Ryan O</description>
<pubDate>19 Nov  2009 10:36:03 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120958</link>
</item><item>
<title>cdrs</title>
<description>why would a CDR are a blank duration on a call? I&amp;#039;m doing CDR records minute by minute on a CM 6.1 cluster. Thanks Scott</description>
<pubDate>19 Nov  2009 10:31:12 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120957</link>
</item><item>
<title>Cisco 2811 SIP trunk and FXS port</title>
<description>Hi,    I have a remote site with a 2811 router connected to a PSTN via SIP trunk. I have a requirement to connect a fax to an FXS/DID port on the</description>
<pubDate>19 Nov  2009 10:29:23 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120956</link>
</item><item>
<title>Re: FW: CUCM Route List</title>
<description>The behavior should be in the route list component and should not rely on the device the call is routing through so I&amp;#039;d expect it to behave the same i</description>
<pubDate>19 Nov  2009 10:27:36 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120955</link>
</item><item>
<title>SIP Issue</title>
<description>This is a little off-topic but to me still relevant in my learning process. I&amp;#039;m not looking for someone to solve this specific problem, in and of itse</description>
<pubDate>19 Nov  2009 10:00:45 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120949</link>
</item><item>
<title>Re: FW: CUCM Route List</title>
<description>George,         I have the same verbiage on a CUCM 7.1 cluster but apparently the behavior also applies to on cluster scenarios. Ryan may kno</description>
<pubDate>19 Nov  2009 09:48:02 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120948</link>
</item><item>
<title>Re: FW: CUCM Route List</title>
<description>Hey Mark/Ryan,    This is good to know. I have CCM 3.3 cluster which has h.323 gateways and I&amp;#039;d like to set this parameter. I believe the paramet</description>
<pubDate>19 Nov  2009 09:33:04 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120947</link>
</item><item>
<title>Fwd: Convert Wav to RAW file for ringtones</title>
<description>---------- Forwarded message ---------- From: Peter Pauly &amp;lt;ppauly@gmail.com&amp;gt; Date: Thu, Nov 19, 2009 at 12:17 PM Subject: Re: [cisco-voip] Convert Wav</description>
<pubDate>19 Nov  2009 09:17:49 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120946</link>
</item><item>
<title>Re: VWIC2-2MFT-T1/E1 for non-voice data ?</title>
<description>Thanks guys. This is my first new gen card for data T1s. Scott On Thu, Nov 19, 2009 at 8:41 AM, Matthew Loraditch &amp;lt; MLoraditch@heliontechnologies.c</description>
<pubDate>19 Nov  2009 09:12:54 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120945</link>
</item><item>
<title>Re: Last Year&amp;#039;s Cisco Live/Networkers Catalog/Schedule</title>
<description>Here the list of sessions that were recorded for the 2009 event. I don&amp;#039;t think its the entire list of sessions but it will probably suit your needs.</description>
<pubDate>19 Nov  2009 08:57:55 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120944</link>
</item><item>
<title>Re: Mobile Voice Access - Problem</title>
<description>can you post the relevant config from the router what version of CUCM what version of IOS and so on On Thu, Nov 19, 2009 at 9:22 AM, Nenad Lazarevic</description>
<pubDate>19 Nov  2009 08:43:28 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120943</link>
</item><item>
<title>Re: VWIC2-2MFT-T1/E1 for non-voice data ?</title>
<description>controller T1 0/0/0 channel-group 0 timeslots 1-24 Configure the controllers as such. You will then get serial0/0/0:0 and that acts just like a regul</description>
<pubDate>19 Nov  2009 08:41:48 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120942</link>
</item><item>
<title>Re: VWIC2-2MFT-T1/E1 for non-voice data ?</title>
<description>controller T1 X/X/X channel-group 0 timeslots 1-24  That should do it for you.  Brian  On Thu, Nov 19, 2009 at 10:23 AM, Scott Voll &amp;lt;svoll.voip@gm</description>
<pubDate>19 Nov  2009 08:40:49 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120941</link>
</item><item>
<title>VWIC2-2MFT-T1/E1 for non-voice data ?</title>
<description>Can I use a VWIC2-2MFT-T1/E1 for just a point to point data T1? if so, how do I get the serial interfaces setup? I can add card type, but then I get</description>
<pubDate>19 Nov  2009 08:23:11 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120940</link>
</item><item>
<title>Re: FW: CUCM Route List</title>
<description>Thanks Sean. It&amp;#039;s good information to know that this will occur between two H.323 GW&amp;#039;s as well. I also appreciate the troubleshooting tip. I modifi</description>
<pubDate>19 Nov  2009 08:00:25 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120932</link>
</item><item>
<title>Re: FW: CUCM Route List</title>
<description>Just a note on the H.323 and temporary failure... I ran into this while testing failover between two H.323 GWs. If you run &amp;quot;show dial-peer voice summ</description>
<pubDate>19 Nov  2009 07:58:17 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120931</link>
</item><item>
<title>Re: InterCluster Disconnects</title>
<description>H.245 interruptions are usually caused by one of several things: 1. a race condition inside ccm.exe processing of the required signals 2. firewalls a</description>
<pubDate>19 Nov  2009 07:50:03 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120929</link>
</item><item>
<title>Re: FW: CUCM Route List</title>
<description>Thanks Ryan, that was the fix. I overlooked that setting. I was focusing under H.323 Clusterwide Parameters. Mark Marquez Unified Communications Sp</description>
<pubDate>19 Nov  2009 07:49:24 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120930</link>
</item><item>
<title>CIPC can register but can&amp;#039;t get DN after CCM 4.2(3) upgrade</title>
<description>Dear All, I just finished CCM upgrade from 4.02(a) to 4.2(3), everything looked good during the upgrade. But after it&amp;#039;s done, my CIPC can&amp;#039;t get DN su</description>
<pubDate>19 Nov  2009 07:32:15 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120925</link>
</item><item>
<title>Re: InterCluster Disconnects</title>
<description>Wes, Thanks for that little tidbit. Which, of course, leads to the logical question - Any ideas on what is causing that to be interrupted? Latency a</description>
<pubDate>19 Nov  2009 07:29:35 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120924</link>
</item><item>
<title>Mobile Voice Access - Problem</title>
<description>I have configure Mobile Voice Access on CCM 7.1.2 and my H323 GW,     PSTN phone-----&amp;gt;FXO-H323 GW.-----&amp;gt;Voice mobile access number -------&amp;gt;CCM   but w</description>
<pubDate>19 Nov  2009 07:22:28 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120939</link>
</item><item>
<title>Re: InterCluster Disconnects</title>
<description>hold/unhold triggers h.245 communication over the ICT to redirect the media. If this is failing in your script it will likely fail for users perfor</description>
<pubDate>19 Nov  2009 06:55:54 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120923</link>
</item><item>
<title>Re: Wallboard Apps</title>
<description>We use Inova for our UCCX wallboards. It has a whole bunch of input plugins for other systems, you may want to check with them to find out if it can p</description>
<pubDate>19 Nov  2009 06:46:31 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120922</link>
</item><item>
<title>Re: Query for DN Active/Inactive</title>
<description>Yep - That&amp;#039;s why I was initially looking for a query for inactive/active directory number. We try to avoid this by assigning them to CTI RP&amp;#039;s and givi</description>
<pubDate>19 Nov  2009 06:38:57 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120921</link>
</item><item>
<title>Re: IVR issue</title>
<description>...another thing. If you use the GUI to record a custom greeting, it will not automatically select &amp;quot;play user recording&amp;quot; (or whatever it says), unlike</description>
<pubDate>19 Nov  2009 06:33:56 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/120920</link>
</item>
</channel>
</rss>
