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<title>Cisco | VOIP</title>
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<lastBuildDate>12 Feb  2012 16:55:52 -0800</lastBuildDate>
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<title>Gossamer Threads | Cisco | VOIP</title>
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<link>http://www.gossamer-threads.com/lists/cisco/voip/</link>
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<item>
<title>Re: Finding the MCS Chassis Model via CUCM Admin</title>
<description>I believe the new Platform Administration Web Services(PAWS) will provide this: http://developer.cisco.com/paws/API_Reference/ME-HardwareModelService</description>
<pubDate>12 Feb  2012 07:45:57 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157856</link>
</item><item>
<title>Re: CUCM 8.5 and SIP options ping</title>
<description>Hello Ryan Thank you for the reply and sorry for the late follow-up. Looked over the information and I haven&amp;#039;t been able to find the information I w</description>
<pubDate>11 Feb  2012 17:00:48 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157854</link>
</item><item>
<title>Re: Finding the MCS Chassis Model via CUCM Admin</title>
<description>On the OS Platform Admin screen (not the main admin screen) many commands available from the CLI are there. You can do a show cluster but I think what</description>
<pubDate>11 Feb  2012 14:32:46 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157853</link>
</item><item>
<title>Re: Finding the MCS Chassis Model via CUCM Admin</title>
<description>Hi Florian, Yes, that does give me the model number but is there a different way either by the GUI or AXL/SQL? Thx, Chris On Sat, Feb 11, 2012 at</description>
<pubDate>11 Feb  2012 13:30:35 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157851</link>
</item><item>
<title>Re: Finding the MCS Chassis Model via CUCM Admin</title>
<description>Hy, u mean something like a &amp;#039;show hardware&amp;#039; on cli? -- Florian Kroessbacher Am 11.02.2012 um 22:10 schrieb Chris Lee &amp;lt;chris@variphy.com&amp;gt;: &amp;gt; Hi Voi</description>
<pubDate>11 Feb  2012 13:14:38 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157850</link>
</item><item>
<title>Finding the MCS Chassis Model via CUCM Admin</title>
<description>Hi Voice Guru&amp;#039;s - Is it possible to find the MCS chassis model from GUI of CUCM Administrator or is there a AXL or SQL command to fetch that info? R</description>
<pubDate>11 Feb  2012 13:08:03 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157849</link>
</item><item>
<title>Re: Searching for a feature</title>
<description>On 02/10/2012 01:41 PM, Bernhard Albler wrote: &amp;gt; &amp;gt; So if you are not dependent on the Voicemail button (or can train the &amp;gt; user to just dial the vm p</description>
<pubDate>10 Feb  2012 14:56:33 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157843</link>
</item><item>
<title>Re: 8.5(1) .zip download version of IP Phone SDK matching 8.5(1) Release Notes?</title>
<description>Hi Jason, Although the IP Phone SDK doc is updated for 8.5, I don&amp;#039;t think anything is new/different, hence the reason the only download is for 7.1. T</description>
<pubDate>10 Feb  2012 14:51:55 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157842</link>
</item><item>
<title>8.5(1) .zip download version of IP Phone SDK matching 8.5(1) Release Notes?</title>
<description>8.5(1) has a IP Phone SDK per release notes below http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/all_models/xsi/8_5_1/ipphoneservicessdk.html</description>
<pubDate>10 Feb  2012 12:53:48 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157835</link>
</item><item>
<title>Re: Unity Connection Upload a greeting</title>
<description>Think I got it I saw a reference to Java 6 Update 7 in one of the threads. So I loaded this (ontop of Update 30) Java still reports that it&amp;#039;s Updat</description>
<pubDate>10 Feb  2012 12:46:53 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157834</link>
</item><item>
<title>Re: Got Ice Cream Sandwhich still no Anyconnect for Android in Market</title>
<description>I&amp;#039;m always a fan of rooting :-) http://forum.xda-developers.com/forumdisplay.php?s=faa209de5bb19f396fc5cbba7527db26&amp;amp;f=1414 -mike From: Jason Aarons</description>
<pubDate>10 Feb  2012 12:40:20 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157833</link>
</item><item>
<title>Re: Unity Connection Upload a greeting</title>
<description>to confirm it&amp;#039;s not a recording codec issue, trying downloading an existing greeting and uploading it again to see if it&amp;#039;s in fact your PC.  just a t</description>
<pubDate>10 Feb  2012 12:18:19 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157826</link>
</item><item>
<title>Re: Unity Connection Upload a greeting</title>
<description>It does not play the correct greeting after I log out and go back in and use the media master bar. Actually, it only plays the correct greeting in me</description>
<pubDate>10 Feb  2012 12:06:14 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157825</link>
</item><item>
<title>Re: Unity Connection Upload a greeting</title>
<description>key thing - does it play the correct greeting after you log out and go back in and use the media master bar again?  if so, then check which greetings</description>
<pubDate>10 Feb  2012 12:01:20 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157824</link>
</item><item>
<title>Unity Connection Upload a greeting</title>
<description>So I have a new computer. Old one is not accessible. I need to upload a greeting to a call handler in Unity Connection. Greeting is created by a th</description>
<pubDate>10 Feb  2012 11:58:05 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157823</link>
</item><item>
<title>Re: Searching for a feature</title>
<description>Hey there iDivert really just diverts the Call to the Voicemail Pilot of the attached Voicemail Profile.  So if you are not dependent on the Voicema</description>
<pubDate>10 Feb  2012 11:41:48 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157822</link>
</item><item>
<title>Re: Searching for a feature</title>
<description>Hy, i think with the 3rd Party ANDTEK Group (andtek.com) there is a feature that redirects the call to a configurable destination. -- Florian Kroess</description>
<pubDate>10 Feb  2012 11:35:02 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157821</link>
</item><item>
<title>Re: Searching for a feature</title>
<description>Hy, i think with the 3rd Party ANDTEK Group (andtek.com) there is a feature that redirects the call to a configurable destination. -- Florian Kroess</description>
<pubDate>10 Feb  2012 11:01:32 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157820</link>
</item><item>
<title>Searching for a feature</title>
<description>I am looking for a service or feature that behaves like Immediate Divert, but instead sends the call to an assistant or a configurable DN. I am gues</description>
<pubDate>10 Feb  2012 09:44:15 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157813</link>
</item><item>
<title>Re: intercluster trunk over IPSec VPN</title>
<description>I was surprised to find that SLA is not included in the IPBase module of v15, nor in the UC module. You need the Data module.   Sent from my iPhone.</description>
<pubDate>10 Feb  2012 09:32:22 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157812</link>
</item><item>
<title>Re: intercluster trunk over IPSec VPN</title>
<description>Maybe a good place for some IP SLA monitoring. Dennis Heim Senior Engineer (Unified Communications) CDW Advanced Technology Services 10610 9th Place</description>
<pubDate>10 Feb  2012 09:19:06 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157811</link>
</item><item>
<title>Re: Got Ice Cream Sandwhich still no Anyconnect for Android in Market</title>
<description>Good info. Guess it&amp;#039;s time to root and mod? Lol. From: Mike Wilusz (miwilusz) [mailto:miwilusz@cisco.com] Sent: Friday, February 10, 2012 10:42 AM T</description>
<pubDate>10 Feb  2012 09:00:31 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157810</link>
</item><item>
<title>Re: Got Ice Cream Sandwhich still no Anyconnect for	Android in Market</title>
<description>Transformer Prime 5 processor downgrade to Samsung dual-core? No thanks! Did you root the Galaxy? Or via the Market? Where did you get the AnyConnec</description>
<pubDate>10 Feb  2012 08:58:43 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157809</link>
</item><item>
<title>Has anyone heard of ISDN15 ?</title>
<description>A telco in Poland is offering an ISDN15 for a site install. It&amp;#039;s a fraction of the cost of ISDN30, but definitely isn&amp;#039;t provided as an ISDN30 with onl</description>
<pubDate>10 Feb  2012 08:43:45 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157808</link>
</item><item>
<title>Re: Replacing MoH file - What am I missing?</title>
<description>OK...That may have worked after all. Maybe I tested too soon. Thanks for the help! On Fri, Feb 10, 2012 at 11:32 AM, Dave Wolgast &amp;lt;dwolgas1@rocheste</description>
<pubDate>10 Feb  2012 08:36:46 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157807</link>
</item><item>
<title>Re: Replacing MoH file - What am I missing?</title>
<description>James, Thanks...I forgot to say that I did that already, on all 3 nodes. On Fri, Feb 10, 2012 at 11:30 AM, Buchanan, James &amp;lt;jbuchanan@presidio.com&amp;gt;w</description>
<pubDate>10 Feb  2012 08:32:00 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157806</link>
</item><item>
<title>Re: Replacing MoH file - What am I missing?</title>
<description>Restart the Cisco TFTP service on all three servers James Buchanan| UC Technology Manager | Presidio South | Presidio Networked Solutions 12 Cadillac</description>
<pubDate>10 Feb  2012 08:30:30 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157805</link>
</item><item>
<title>Replacing MoH file - What am I missing?</title>
<description>Using CUCM 7.1(3) I uploaded a new MoH file to each node of the cluster (all nodes run MoH Server/IPVMS service) using MoH File Management. CUCM repo</description>
<pubDate>10 Feb  2012 08:19:54 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157804</link>
</item><item>
<title>Re: phones deleted during user garbage collection</title>
<description>Thanks Wes. I need to get better at searching for bugs, I can never find them. Good to know it&amp;#039;s fixed in 8.6(2) since we&amp;#039;re planning to upgrade in th</description>
<pubDate>10 Feb  2012 08:00:09 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157801</link>
</item><item>
<title>Re: Service Paramter FW: Always Display Original Dialed Number</title>
<description>Allowing applications - whether SIP, SCCP, or CTI based - to pass through caller id information is woefully inconsistent.  There is another facet of</description>
<pubDate>10 Feb  2012 07:50:05 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157800</link>
</item><item>
<title>Re: Got Ice Cream Sandwhich still no Anyconnect for Android in Market</title>
<description>We have a GalaxyTab here that we have confirmed working with Anyconnect (much to the user&amp;#039;s joy). Pretty sure it is older that ICS too. Maybe you can</description>
<pubDate>10 Feb  2012 07:46:27 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157799</link>
</item><item>
<title>Re: Got Ice Cream Sandwhich still no Anyconnect for Android in Market</title>
<description>Jason, While SSL VPN client support was added by Google in ICS, there was an error in some of the manufacturer loads that left a critical library out</description>
<pubDate>10 Feb  2012 07:42:05 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157798</link>
</item><item>
<title>Re: intercluster trunk over IPSec VPN</title>
<description>most likely still packet throughput issues. packets may be late to the point of discarded. they would not technically be lost in that case. this woul</description>
<pubDate>10 Feb  2012 07:35:41 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157797</link>
</item><item>
<title>Re: Service Paramter FW: Always Display Original Dialed Number</title>
<description>Yes but say a Secretary calls a manager, it rolls to his voicemail than the screen display won&amp;#039;t update showing voicemail, etc. There are lot of non</description>
<pubDate>10 Feb  2012 06:54:10 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157796</link>
</item><item>
<title>CUPC 8.5.4 client server name at bottom on login</title>
<description>The first time you launch CUPC you have to put the server name in at the bottom (we can&amp;#039;t use DNS SRV as we have multiple CUPS servers per region and</description>
<pubDate>10 Feb  2012 06:34:58 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157795</link>
</item><item>
<title>Got Ice Cream Sandwhich still no Anyconnect for Android in Market</title>
<description>This reminds me of the 64 bit VPN Client for Windows 7, came out way too late. Cisco indicated Andorid 2x/3xdidn&amp;#039;t have the APIs etc, that ICS would</description>
<pubDate>10 Feb  2012 06:18:59 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157794</link>
</item><item>
<title>Re: CUEAC Install</title>
<description>Thanks for the assistance everyone. It helps when the OS is 32-bit and not 64-bit. One of those Layer 1 problems again. From: Matthew Loraditch [ma</description>
<pubDate>10 Feb  2012 06:09:46 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157791</link>
</item><item>
<title>Re: 7925 Wireless Setup</title>
<description>If you have available DLUs the phone will consume them and work. There is no technical need to install any more license files. Legally since you are o</description>
<pubDate>10 Feb  2012 05:40:17 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157790</link>
</item><item>
<title>Re: Cisco VOIP version 8.6 Backup Clients</title>
<description>It&amp;#039;s not possible to use another tool, with no root access you can&amp;#039;t install anything nor could you possibly access everything that needs backing up.</description>
<pubDate>10 Feb  2012 05:36:40 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157789</link>
</item><item>
<title>Re: 7925 Wireless Setup</title>
<description>Thanks Mike. I did have the deployment guide, but sheesh! I started going through it and it is a bit overwhelming. And of course, the standard Cisc</description>
<pubDate>10 Feb  2012 05:30:53 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157788</link>
</item><item>
<title>Weather apps for 896x &amp;amp; 99xx phones</title>
<description>I&amp;#039;ve got a customer that wants Weather alerts on their phones - 896x &amp;amp; 99xx. CCM 8.6. Does anyone have any recommendations on who might do this? Do</description>
<pubDate>10 Feb  2012 05:27:19 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157786</link>
</item><item>
<title>Weather apps for 896x &amp;amp; 99xx phones</title>
<description>I&amp;#039;ve got a customer that wants Weather alerts on their phones - 896x &amp;amp; 99xx. CCM 8.6. Does anyone have any recommendations on who might do this? Le</description>
<pubDate>10 Feb  2012 05:24:04 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157785</link>
</item><item>
<title>Re: CUEAC Install</title>
<description>Depending on what version you are using the install GUI used to be a bit confusing as to what IPs you were supposed to put in where, one field gets t</description>
<pubDate>10 Feb  2012 05:09:46 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157784</link>
</item><item>
<title>Re: CUEAC Install</title>
<description>Usually occurs when the service on the cueac is not running, most times when service is not running the license is invalidated. you have 60 days but t</description>
<pubDate>10 Feb  2012 05:04:08 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157783</link>
</item><item>
<title>Cisco VOIP version 8.6 Backup Clients</title>
<description>Hello,      Is there anyone that is using a dedicated Backup Client to back up their VOIP Servers (Call Manager, IPCC, Unity) rather than using</description>
<pubDate>10 Feb  2012 04:57:18 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157787</link>
</item><item>
<title>Re: CUBE Issues with SIP Reinvite...</title>
<description>On Thu, Feb 9, 2012 at 5:45 PM, Jonathan Charles &amp;lt;jonvoip@gmail.com&amp;gt; wrote: &amp;gt; I am pretty sure SIP to SIP on a cube is not supported, or at least it &amp;gt;</description>
<pubDate>10 Feb  2012 04:37:07 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157782</link>
</item><item>
<title>Re: phones deleted during user garbage collection</title>
<description>I bumped it to severe. Give it 24-48 hours to update in bug toolkit. The version fields also got updated. /wes On Feb 9, 2012, at 10:09 PM, Dennis</description>
<pubDate>10 Feb  2012 04:30:08 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157781</link>
</item><item>
<title>Re: intercluster trunk over IPSec VPN</title>
<description>Dears, thank you all for the excellent support I managed to keep the VPN tunnel up be sending periodic ping but the problem still persist. Bandwidth</description>
<pubDate>10 Feb  2012 03:41:30 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157780</link>
</item><item>
<title>Re: phones deleted during user garbage collection</title>
<description>Moderateâ€¦ if it is deleting phones as part of the GC process, I would call that sev1.  Dennis Heim Senior Engineer (Unified Communications) CDW</description>
<pubDate>09 Feb  2012 19:09:49 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157774</link>
</item><item>
<title>Re: CUCM 8.5 and SIP options ping</title>
<description>The alarms that are used for all SIP options apply to options ping. &amp;gt; Serviceability Alarms &amp;gt; &amp;gt; The following alarms support SIP OPTIONS: &amp;gt; &amp;gt; &amp;#149;SIPTr</description>
<pubDate>09 Feb  2012 18:55:18 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157773</link>
</item><item>
<title>CUCM 8.5 and SIP options ping</title>
<description>Hello  CUCM 8.5 introduced SIP options ping for SIP trunks and my question is how can we see the real-time status of a trunk ? Where can we see if a</description>
<pubDate>09 Feb  2012 16:18:24 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157769</link>
</item><item>
<title>Re: OT Cisco ASA question</title>
<description>Yep this works fine. Just have to set the native VLAN. I had 15 different zones setup this way with no issues. Thanks, Jobe On Feb 9, 2012, at 2:4</description>
<pubDate>09 Feb  2012 15:44:49 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157768</link>
</item><item>
<title>Re: SIP DTMF</title>
<description>What version of CUCM are you running? In 8.5 i have seen problems when this is checked (sometimes no audio) I had to disable it and apply an appropria</description>
<pubDate>09 Feb  2012 14:17:03 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157767</link>
</item><item>
<title>TCL script for announcement !!!</title>
<description>hi group.. does anybody has a TCL script to share which fits into this scenario:   1.    IP phone ( registered to CUCM ) dials a number on PSTN</description>
<pubDate>09 Feb  2012 13:28:58 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157766</link>
</item><item>
<title>Re: phones deleted during user garbage collection</title>
<description>moderate? ayieeeeeeeee!  --- Lelio Fulgenzi, B.A. Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1 (519) 824-4120 x56354 (519</description>
<pubDate>09 Feb  2012 13:24:24 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157765</link>
</item><item>
<title>Re: phones deleted during user garbage collection</title>
<description>CSCtl20458  Phones associated with inactive end users are deleted during GC On Feb 9, 2012, at 2:23 PM, Eric Pedersen wrote: We have Callmanager 8</description>
<pubDate>09 Feb  2012 13:21:37 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157764</link>
</item><item>
<title>Re: SIP DTMF</title>
<description>Fixed it.. I turned use MTP on the trunk and it started to workŠ funny I thought it should be turned off as sending dtmf through a mtp can cause issue</description>
<pubDate>09 Feb  2012 13:05:53 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157763</link>
</item><item>
<title>Re: phones deleted during user garbage collection</title>
<description>I&amp;#039;m not sure I like the idea of anything being deleted just because a userID has been deleted from AD. I can see this leading to problems for sure.</description>
<pubDate>09 Feb  2012 12:59:58 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157762</link>
</item><item>
<title>Re: phones deleted during user garbage collection</title>
<description>i only know this behaviour with remote destination and remote destination profiles. with normal phones i think this isn&amp;#039;t right. what about to contact</description>
<pubDate>09 Feb  2012 12:56:32 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157760</link>
</item><item>
<title>Re: SIP DTMF</title>
<description>Many times this is sloppy dial peers - you may be matching a different dial peer on the CUBE, and one doesn&amp;#039;t have dtmf rtp-nte enabled. Also - certa</description>
<pubDate>09 Feb  2012 12:56:16 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157761</link>
</item><item>
<title>Re: debug MGCP analog</title>
<description>presenting dialtone usually happens when the router is function in h.323 mode and has no connection plar to specify a destination. The router can onl</description>
<pubDate>09 Feb  2012 12:43:18 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157759</link>
</item><item>
<title>Re: SIP DTMF</title>
<description>No the box is not ticked  From: &amp;quot;Matt Slaga (AM)&amp;quot; &amp;lt;matt.slaga@dimensiondata.com&amp;gt; Date: Thu, 9 Feb 2012 15:36:29 -0500 To: Leslie Meade &amp;lt;lmeade@sal</description>
<pubDate>09 Feb  2012 12:39:02 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157758</link>
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<title>Re: SIP DTMF</title>
<description>Also on the ones that are &amp;quot;not working&amp;quot; the numbers from 1 - 4 seam not to work. 5 and about do. So this sounds like the tones are not recognized. I h</description>
<pubDate>09 Feb  2012 12:37:45 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157757</link>
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<title>Re: SIP DTMF</title>
<description>Do you have MTP required selected on your UCM trunk? From: cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net] On Behalf O</description>
<pubDate>09 Feb  2012 12:36:29 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157756</link>
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<title>Re: SIP DTMF</title>
<description>dtmf-relay rtp-nte   On 12-02-09 12:30 PM, &amp;quot;Paul&amp;quot; &amp;lt;asobihoudai@yahoo.com&amp;gt; wrote: &amp;gt;What are you using for DTMF transport? &amp;gt; &amp;gt; &amp;gt;_____________________</description>
<pubDate>09 Feb  2012 12:31:54 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157755</link>
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<title>Re: SIP DTMF</title>
<description>What are you using for DTMF transport?  ________________________________ From: Leslie Meade &amp;lt;lmeade@salientnetworks.com&amp;gt; To: cisco-voip &amp;lt;cisco-voip@p</description>
<pubDate>09 Feb  2012 12:30:57 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157754</link>
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<title>Re: phones deleted during user garbage collection</title>
<description>No, the phones getting deleted are hardware phones with a disabled user as the Owner User ID.  From: Florian Kroessbacher [mailto:florian.kroessbach</description>
<pubDate>09 Feb  2012 12:15:32 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157752</link>
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<title>SIP DTMF</title>
<description>I have an intermit issues with DTMF on my SIP trunk to the providerŠ DTMF seams to be working on some numbers but not on others.. I know that the DTMF</description>
<pubDate>09 Feb  2012 12:02:46 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157751</link>
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<title>Re: OT Cisco ASA question</title>
<description>Thanks to someone that sent me this off list: http://itprofesionals.blogspot.com/2009/08/configuring-cisco-asa-with-8021q-vlan.html  So the the answ</description>
<pubDate>09 Feb  2012 11:45:12 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157750</link>
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<title>Re: debug MGCP analog</title>
<description>Can anyone point me on the right direction on which traces to pull? It&amp;#039;s a 6.1.4 cluster   From: cisco-voip-bounces@puck.nether.net [mailto:cisco-vo</description>
<pubDate>09 Feb  2012 11:40:59 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157749</link>
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<title>OT Cisco ASA question</title>
<description>Sorry to bother the list, but I&amp;#039;m looking for validation of a thought before I go and try to lab it up. Cisco ASA 5550 Is it possible to create an 8</description>
<pubDate>09 Feb  2012 11:40:21 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157748</link>
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<title>Re: CUEAC Install</title>
<description>Mike, You should make sure that AXL is running on the CUCM you are trying to connect too. Try browsing to https://cucm-ipaddress/axl You should be</description>
<pubDate>09 Feb  2012 11:39:14 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157747</link>
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<title>Re: phones deleted during user garbage collection</title>
<description>do you mean remote destinations as their &amp;quot;phones&amp;quot; -- Florian Kroessbacher gmail: florian.kroessbacher@gmail.com Am 09.02.2012 um 20:24 schrieb Eric</description>
<pubDate>09 Feb  2012 11:32:34 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157746</link>
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<title>phones deleted during user garbage collection</title>
<description>We have Callmanager 8 configured for LDAP synchronization with AD. When users are disabled in Active Directory, the nightly garbage collection is rem</description>
<pubDate>09 Feb  2012 11:23:32 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157745</link>
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<title>CUEAC Install</title>
<description>Trying to install CUEAC in a lab environment. I have a Test 8.5 CUCM server up and running. All the services are up. I have followed the Web Admin</description>
<pubDate>09 Feb  2012 10:46:31 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157744</link>
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<title>Re: Off net to off net transfer using Verizon SIP Trunk</title>
<description>There you go. You will need to validate that by allowing the SDP information out, that it does not create other side-affects. Bob Zanett Technical S</description>
<pubDate>09 Feb  2012 10:41:12 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157743</link>
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<title>Re: Service Paramter FW: Always Display Original Dialed Number</title>
<description>Ironically I just reviewed this case: &amp;lt;quote&amp;gt; The following scenario is working as designed: &amp;quot;Always Display Original Dialed Number&amp;quot; in the Service</description>
<pubDate>09 Feb  2012 10:27:01 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157742</link>
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<title>Re: new smartnet option for phones?</title>
<description>All levels of SNT include phone firmware. Hopefully the services finder will be getting corrected soon. -Ryan On Feb 8, 2012, at 4:42 PM, Lelio Ful</description>
<pubDate>09 Feb  2012 10:19:10 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157731</link>
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<title>Re: Off net to off net transfer using Verizon SIP Trunk</title>
<description>Thanks Bob, I have just managed to fix it by adding the the following voice service voip sip  pass-thru content sdp   On 9 February 2012 17:30, B</description>
<pubDate>09 Feb  2012 10:15:27 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157730</link>
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<title>Re: background images limit</title>
<description>Robert, I hate answering a question by asking a question...but what phone models are you using? I don&amp;#039;t know what the limits are for all phone model</description>
<pubDate>09 Feb  2012 09:56:57 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157729</link>
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<title>Re: meetingplace 8.5</title>
<description>Seconded. MP 8.5.2 is what 8.0 should have been to begin with. The only issues with it are when Upgrading from MPX, you basically have to re-impleme</description>
<pubDate>09 Feb  2012 09:45:06 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157728</link>
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<title>Re: CUBE Issues with SIP Reinvite...</title>
<description>You may want to start here: http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb-gw-sipsip.html -nick On Thu, Feb 9, 2012 at 11:45</description>
<pubDate>09 Feb  2012 09:33:24 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157727</link>
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<title>Re: Off net to off net transfer using Verizon SIP Trunk</title>
<description>Nick, Run debugs on the gateway to validate that the transfer (re-invite) is getting to your gateway and how it is handling it.  If CUBE, as I recall</description>
<pubDate>09 Feb  2012 09:30:42 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157726</link>
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<title>Off net to off net transfer using Verizon SIP Trunk</title>
<description>Hi All I have an issue when trying to complete an off net to off net transfer for calls using Verizon SIP trunking, the call arrives on the SIP and t</description>
<pubDate>09 Feb  2012 09:26:21 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157725</link>
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<title>Re: Calls to internal numbers fail</title>
<description>If I remember my port channel correctly, (reference: http://www.cisco.com/en/US/tech/tk389/tk213/technologies_tech_note09186a0080094714.shtml )  The</description>
<pubDate>09 Feb  2012 09:20:33 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157724</link>
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<title>Re: Is it possible to monitor individual b-channels	off of a 66-block terminated PRI?</title>
<description>That certainly is a really cool device! That site has a few nice products.  Thanks for sharing.  --- Lelio Fulgenzi, B.A. Senior Analyst (CCS) *</description>
<pubDate>09 Feb  2012 09:16:05 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157723</link>
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<title>Re: CUBE Issues with SIP Reinvite...</title>
<description>I am not sure about the SIP-to-SIP not working. This was working years ago, as I helped a very large medical firm begin to utilize SIP to telco.  At</description>
<pubDate>09 Feb  2012 09:13:54 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157722</link>
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<title>Re: Is it possible to monitor individual b-channels off of a 66-block terminated PRI?</title>
<description>That&amp;#039;s pretty neat! It still doesn&amp;#039;t resemble *anything* close to what I&amp;#039;ve been described as bridging off of a 66. What about a DS3? I only see a p</description>
<pubDate>09 Feb  2012 08:58:44 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157721</link>
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<title>Re: Calls to internal numbers fail</title>
<description>Interesting. You would have to investigate how traffic was routing through the port channels to determine exactly why it acted as it did. Most likel</description>
<pubDate>09 Feb  2012 08:57:57 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157720</link>
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<title>Re: Is it possible to monitor individual b-channels off of	a 66-block terminated PRI?</title>
<description>Yes... see web site for digital loggers http://digitalloggers.com/T1.html Records all calls directly off the PRI before it ever gets to any terminati</description>
<pubDate>09 Feb  2012 08:49:42 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157719</link>
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<title>Re: CUBE Issues with SIP Reinvite...</title>
<description>I am pretty sure SIP to SIP on a cube is not supported, or at least it doesn&amp;#039;t work. I have tried connecting a SIP trunk from CUCM to the CUBE (and a</description>
<pubDate>09 Feb  2012 08:45:40 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157718</link>
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<title>Service Paramter FW: Always Display Original Dialed Number</title>
<description>From a customer&amp;#039;s internal Cisco 7965 if you dial 12000 (CTI Route Point for our main number) you see 12000 on the screen then your phone display chan</description>
<pubDate>09 Feb  2012 08:43:52 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157717</link>
</item><item>
<title>Re: Upgrade from 8.5.1.13900-5 to 8.6.2.20000-2 failed...</title>
<description>See, I would add a note to the 8.6 upgrade saying this.... when you click on the readme, it just takes you to the CUCM 8.X documentation guide...  Jo</description>
<pubDate>09 Feb  2012 08:43:34 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157716</link>
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<title>Re: Calls to internal numbers fail</title>
<description>It turned out one the members of a port channel was suspended (access on one side, dot1q on the other). When I fixed that the phones were able to call</description>
<pubDate>09 Feb  2012 08:42:37 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157715</link>
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<title>Re: Is it possible to monitor individual b-channels off of a 66-block terminated PRI?</title>
<description>Nope. The B channels are multiplexed onto a T1 or E1 carrier; the only way to do it is with an Adtran or a T-bird.  Jonathan On Thu, Feb 9, 2012 at</description>
<pubDate>09 Feb  2012 08:41:07 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157714</link>
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<title>Is it possible to monitor individual b-channels off of a 66-block terminated PRI?</title>
<description>If it is, it&amp;#039;d be the first time *I&amp;#039;ve* ever heard of it. Some folks I&amp;#039;m working with are suggesting this is the best way to configure call recording</description>
<pubDate>09 Feb  2012 08:36:07 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157713</link>
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<title>Re: intercluster trunk over IPSec VPN</title>
<description>TCP keepalives are only used while a call is active. When no call is active there is no active h323/h225/h245 signaling, tcp session, or udp. The on</description>
<pubDate>09 Feb  2012 06:57:26 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157707</link>
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<title>Re: intercluster trunk over IPSec VPN</title>
<description>No, I don&amp;#039;t think it&amp;#039;s there in 6.1.3 and if it was it would show up in the SIP profile. Incidentally 6.1(3) base is now over 3 years old. You reall</description>
<pubDate>09 Feb  2012 06:55:32 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157705</link>
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<title>Re: MOH location on CM 7</title>
<description>I believe that is correct. IPVMS logs print when they load an MOH files. Turn on your IPVMS traces and collect them for a week. avoid deleting any</description>
<pubDate>09 Feb  2012 06:55:10 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157706</link>
</item><item>
<title>Re: 7925 Wireless Setup</title>
<description>Hi Dave, Here&amp;#039;s the deployment guide: http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7925g/7_0/english/deployment/guide/7925dply.pdf  Page 8 s</description>
<pubDate>09 Feb  2012 06:26:01 -0800</pubDate>
<link>http://www.gossamer-threads.com/lists/cisco/voip/157704</link>
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