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		<item>
		<title>IOS Gateway Configuration for SIP-Proxy Redundancy</title>
		<link>http://www.voiptechblogs.com/?p=301</link>
		<comments>http://www.voiptechblogs.com/?p=301#comments</comments>
		<pubDate>Tue, 15 Nov 2011 09:53:55 +0000</pubDate>
		<dc:creator>Shafqut Hamid</dc:creator>
				<category><![CDATA[GWGK]]></category>
		<category><![CDATA[Cisco]]></category>
		<category><![CDATA[Failover]]></category>
		<category><![CDATA[Redundancy]]></category>
		<category><![CDATA[VoIP]]></category>

		<guid isPermaLink="false">http://www.voiptechblogs.com/?p=301</guid>
		<description><![CDATA[With Cisco IOS gateways, dial-peers are used to match phone numbers, and the destination can be a SIP Proxy Server, DNS SRV, or IP address. The following example shows a Cisco IOS gateway configuration to send calls to a SIP Proxy Server using the SIP Proxy&#8217;s IP address. sip-ua sip-server ipv4:10.4.1.100:5060 dial-peer voice 1000 voip [...]]]></description>
			<content:encoded><![CDATA[<p>With Cisco IOS gateways, dial-peers are used to match phone numbers, and  the destination can be a SIP Proxy Server, DNS SRV, or IP address. The  following example shows a Cisco IOS gateway configuration to send calls  to a SIP Proxy Server using the SIP Proxy&#8217;s IP address.</p>
<blockquote>
<p style="padding-left: 30px;">sip-ua<br />
sip-server ipv4:10.4.1.100:5060<br />
dial-peer voice 1000 voip<br />
session target sip-server<br />
&#8230;<span id="more-301"></span></p>
</blockquote>
<p>The sip-server command on the dial-peer tells the Cisco IOS gateway to use the globally defined<br />
sip-server  that is configured under the sip-ua settings. In order to configure  multiple SIP Proxies for redundancy, you can change the IP address to a  DNS SRV record, as shown in the following example. The DNS SRV record  allows a single DNS name to be mapped to multiple servers.</p>
<blockquote>
<p style="padding-left: 30px;">sip-ua<br />
sip-server dns:cvp.cisco.com<br />
dial-peer voice 1000 voip<br />
session target sip-server<br />
&#8230;</p>
</blockquote>
<p>Alternatively,  you can configure multiple dial-peers to point directly at multiple SIP  Proxy servers, as shown in the following example. This configuration  allows you to specify IP addresses instead of relyingon DNS.</p>
<blockquote>
<p style="padding-left: 30px;">dial-peer voice 1000 voip<br />
session target ipv4:10.4.1.100<br />
preference 1<br />
&#8230;<br />
dial-peer voice 1000 voip<br />
session target ipv4:10.4.1.101<br />
preference 1<br />
&#8230;</p>
</blockquote>
<p>In  the preceding examples, the calls are sent to the SIP Proxy server for  dial plan resolution and callrouting. If there are multiple Unified CVP  Call Servers, the SIP Proxy server would be configured with multiple  routes for load balancing and redundancy.</p>
<p>DNS SRV records allow an administrator to configure redundancy and load balancing with finer<br />
granularity  than with DNS round-robin redundancy and load balancing. A DNS SRV  record allows you to define which hosts should be used for a particular  service (the service in this case is SIP), and it allowsyou to define  the load-balancing characteristics among those hosts. In the  following  example, the<br />
redundancy provided by the three dial-peers configured  above is replaced with a single dial-peer using a DNS SRV record. Note  that a DNS server is required in order to do the DNS lookups.</p>
<blockquote>
<p style="padding-left: 30px;">ip name-server 10.4.33.200<br />
dial-peer voice 1000 voip<br />
session target dns:cvp.cisco.com</p>
</blockquote>
<p>With  Cisco IOS gateways, it is possible to define DNS SRV records  statically, similar to static host records. This capability allows you  to simplify dial-peer configuration while also providing DNS SRV load  balancing and redundancy. The downside of this method is that, if the  SRV record needs to be changed, it must be changed on each gateway  instead of on a centralized DNS server. The following example shows the  configuration of static SRV records for SIP services handled by  cvp.cisco.com, and the SIP SRV records for cvp.cisco.com are configured  to load-balance across three servers.</p>
<blockquote>
<p style="padding-left: 30px;">ip host cvp4cc2.cisco.com 10.4.33.132<br />
ip host cvp4cc3.cisco.com 10.4.33.133<br />
ip host cvp4cc1.cisco.com 10.4.33.131<br />
(SRV records for SIP/TCP)<br />
ip host _sip._tcp.cvp.cisco.com srv 1 50 5060 cvp4cc3.cisco.com<br />
ip host _sip._tcp.cvp.cisco.com srv 1 50 5060 cvp4cc2.cisco.com<br />
ip host _sip._tcp.cvp.cisco.com srv 1 50 5060 cvp4cc1.cisco.com<br />
(SRV records for SIP/UDP)<br />
ip host _sip._udp.cvp.cisco.com srv 1 50 5060 cvp4cc3.cisco.com<br />
ip host _sip._udp.cvp.cisco.com srv 1 50 5060 cvp4cc2.cisco.com<br />
ip host _sip._udp.cvp.cisco.com srv 1 50 5060 cvp4cc1.cisco.com</p>
</blockquote>
<p>Cisco  highly recommends the use of UDP instead of TCP for SIP signaling. TCP  stack timeout delays can cause significant delays to the caller during  failures.</p>
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		<item>
		<title>CME-Topic 26: MOH Configuration</title>
		<link>http://www.voiptechblogs.com/?p=281</link>
		<comments>http://www.voiptechblogs.com/?p=281#comments</comments>
		<pubDate>Thu, 17 Feb 2011 08:56:57 +0000</pubDate>
		<dc:creator>Shafqut Hamid</dc:creator>
				<category><![CDATA[CME]]></category>
		<category><![CDATA[CCIE Voice]]></category>
		<category><![CDATA[Cisco]]></category>
		<category><![CDATA[Cisco Call Manager Express]]></category>
		<category><![CDATA[CUCME]]></category>
		<category><![CDATA[MOH]]></category>
		<category><![CDATA[Music On Hold]]></category>
		<category><![CDATA[UC]]></category>

		<guid isPermaLink="false">http://www.voiptechblogs.com/?p=281</guid>
		<description><![CDATA[MOH is an audio stream which can be played to PSTN and VoIP G.711 or G.729 callers who are placed on hold by phones in a Cisco Unified CME system. This audio stream is intended to reassure callers that they are still connected to their calls. When the phone receiving MOH is part of a system that [...]]]></description>
			<content:encoded><![CDATA[<p>MOH is an audio stream which can be played to PSTN and VoIP G.711 or  G.729  callers who are placed on hold by phones in a Cisco Unified CME  system.  This audio stream is intended to reassure callers that they are  still  connected to their calls.</p>
<p>When the phone receiving MOH is  part of a system that uses a G.729  codec, transcoding is required  between G.711 and G.729. The G.711 MOH  must be translated to G.729.</p>
<blockquote><p><strong>Note:</strong> Because of compression, MOH using  G.729 is of significantly  lower fidelity than MOH using G.711.</p></blockquote>
<p>Below are some MOH resources</p>
<ul>
<li><strong>Flash Memory:</strong> MOH can be played from local stored files in flash memory without the requirement of external source<span id="more-281"></span></li>
<li><strong>Live Feed:</strong> The multicast audio stream has minimal delay for local IP phones. The  MOH stream for PSTN callers is delayed by a few seconds. If the live  feed audio input fails, callers on hold hear silence.</li>
<li><strong>Live Feed and Flash Memory:</strong> This is the preferred method if you want to use Live Feed, so that if Live Feed fails then users still have MOH from local flash memory</li>
</ul>
<p>Below is very basic configuration for MOH</p>
<blockquote>
<pre>CME(config-telephony)# moh {filename.wav | filename.au}
CME(config-telephony)# multicast moh 239.1.1.30 port 2000
</pre>
</blockquote>
<p>Cisco Unified CME 8.0 MOH enhancement allows you to create MOH groups and assign ephone extension numbers to these MOH groups to receive different media streams. Callers to the extension numbers configured under the MOH groups can listen to different MOH media streams when they are placed on hold. Following precedence rules are applicable when an ephone caller is placed on hold:</p>
<ul>
<li><strong>MOH group</strong> defined for internal calls takes highest precedence</li>
<li><strong>MOH group</strong> defined in ephone-dn takes the second highest precedence</li>
<li><strong>MOH group</strong> defined in ephone-dn-template takes precedence if MOH group is not defined in ephone-dn or internal call.</li>
<li>Extension numbers defined in a MOH-group has the least precedence</li>
<li>Phones not associated with any MOH groups default to the MOH parameters defined in the moh command under telephony-service configuration mode.</li>
</ul>
<blockquote>
<pre>router# show voice moh-group
telephony-service
  moh alaska.wav
  Moh multicast 239.1.1.1 port 16384 route 10.1.4.31 10.1.1.2

voice moh-group 1
  description this moh group is for sales
  moh flash:/hello.au
  multicast moh 239.1.1.1 port 16386 route 239.1.1.3 239.1.1.3
  extension-range 1000 to 1999
  extension-range 2000 to 2999
  extension-range 3000 to 3999
  extension-range A1000 to A1999

voice moh-group 2
  description (not configured)
  moh flash1:/minuet.au
  multicast moh 239.23.4.10 port 2000
  extension-range 7000 to 7999</pre>
</blockquote>
]]></content:encoded>
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		</item>
		<item>
		<title>CME-Topic 25: Automatic Line Selection</title>
		<link>http://www.voiptechblogs.com/?p=270</link>
		<comments>http://www.voiptechblogs.com/?p=270#comments</comments>
		<pubDate>Thu, 17 Feb 2011 07:34:22 +0000</pubDate>
		<dc:creator>Shafqut Hamid</dc:creator>
				<category><![CDATA[CME]]></category>
		<category><![CDATA[Auto Line Selection]]></category>
		<category><![CDATA[CCIE Voice]]></category>
		<category><![CDATA[Cisco]]></category>
		<category><![CDATA[Cisco Call Manager Express]]></category>
		<category><![CDATA[CUCME]]></category>
		<category><![CDATA[UC]]></category>

		<guid isPermaLink="false">http://www.voiptechblogs.com/?p=270</guid>
		<description><![CDATA[This feature is very useful on multi-line IP phones where lifting the handset automatically selects the first ringing line on the phone or, if no line is ringing, selects the first available idle line for outgoing calls. This is the default behavior for all multi-line IP phones. But in some situations you require that a [...]]]></description>
			<content:encoded><![CDATA[<p>This feature is very useful on multi-line IP phones where lifting the handset automatically selects the first ringing line on the phone or, if no line is ringing, selects the first available idle line for outgoing calls. This is the default behavior for all multi-line IP phones. But in some situations you require that a specific line button should be explicitly pressed to select an outgoing line or to answer an incoming call. In Cisco CME 3.0 and later, you have the flexibility to assign the type of line selection that each IP phone uses.</p>
<p>The Automatic Line Selection feature allows you to specify, on a per-phone basis, the line that is selected when you pick up a phone handset. Any of the following behaviors can be assigned on a per-phone basis:</p>
<ul>
<li><strong>Automatic line selection</strong> Picking up the handset      answers the first ringing line or, if no line is ringing, selects the first      idle line. Use the <strong>auto-line </strong>command with no keyword or argument.      This is the default. <span id="more-270"></span><strong> </strong></li>
<li><strong>Manual line selection</strong> (no automatic line      selection) Pressing the Answer soft key answers the first ringing line,      and pressing a line button selects a line for an outgoing call. Picking up      the handset does not answer calls or provide dial tone.</li>
<li> Use the <strong>no auto-line </strong>command.<strong> </strong>Automatic line selection for incoming calls only. Picking up the      handset answers the first ringing line, but if no line is ringing, it does      not select an idle line for an outgoing call. Pressing a line button      selects a line for an outgoing call.</li>
<li>Use the <strong>auto-line incoming </strong>command. Automatic line      selection for outgoing calls only. Picking up the handset for an outgoing      call selects the line associated with the <em>button-number </em>argument.      If a button number is specified and the line associated with that button      is unavailable (because it is a shared line in use on another phone), no      dial tone is heard when the handset is lifted. You must press an available      line button to make an outgoing call. Incoming calls must be answered by      pressing the Answer soft key or pressing a ringing line button.</li>
<li>Use the <strong>auto-line </strong>command with the <em>button-number </em>argument.      <strong> </strong>Automatic line selection for incoming and outgoing calls. Pressing      the Answer soft key or picking up the handset answers an incoming call on      the line associated with the specified button. Picking up the handset for      outgoing calls selects the line associated with the specified button.</li>
<li>Use the <strong>auto-line </strong>command with the <em>button-number </em>argument      and <strong>answer-incoming </strong>keyword.</li>
</ul>
<blockquote>
<pre>ephone 5
  auto-line 5 answer-incoming</pre>
</blockquote>
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		</item>
		<item>
		<title>CME-Topic 24: After Hours Call Blocking</title>
		<link>http://www.voiptechblogs.com/?p=265</link>
		<comments>http://www.voiptechblogs.com/?p=265#comments</comments>
		<pubDate>Thu, 17 Feb 2011 07:15:14 +0000</pubDate>
		<dc:creator>Shafqut Hamid</dc:creator>
				<category><![CDATA[CME]]></category>
		<category><![CDATA[After Hours Call Blocking]]></category>
		<category><![CDATA[CCIE Voice]]></category>
		<category><![CDATA[Cisco]]></category>
		<category><![CDATA[Cisco Call Manager Express]]></category>
		<category><![CDATA[CUCME]]></category>
		<category><![CDATA[UC]]></category>

		<guid isPermaLink="false">http://www.voiptechblogs.com/?p=265</guid>
		<description><![CDATA[After hours call blocking blocks specific number for a specific time period Configured from telephony service mode Phones can be exampled (always or through pins) 24-7 block patterns can never be allowed After-hours call blocking has three major steps of configuration: Define days and/or hours of the day that your company considers off-hours. Specify patterns [...]]]></description>
			<content:encoded><![CDATA[<ul>
<li>After hours call blocking blocks specific number <a href='http://atlantic-drugs.net/products/viagra.htm'>for</a> a specific      time period</li>
<li>Configured from telephony service mode</li>
<li>Phones can be exampled (always or through pins)</li>
<li>24-7 block patterns can never be allowed</li>
</ul>
<p>After-hours call blocking has three major steps of configuration:</p>
<ol>
<li> Define days and/or hours of the day that your company      considers off-hours.</li>
<li> Specify patterns that you      would like to block during the times specified in Step 1.</li>
<li>Create exemptions to the policy, if needed. <span id="more-265"></span></li>
</ol>
<blockquote>
<pre>CME(config-telephony)# after-hours date dec 25 00:00 00:00
CME(config-telephony)# after-hours day fri 17:00 8:00
CME(config-telephony)# after-hours day thu 17:00 8:00
CME(config-telephony)# after-hours day wed 17:00 8:00
CME(config-telephony)# after-hours day tue 17:00 8:00
CME(config-telephony)# after-hours day mon 17:00 8:00
CME(config)# telephony-service</pre>
</blockquote>
<blockquote>
<pre>CME(config-telephony)# after-hours block pattern 1 91..........
CME(config-telephony)# after-hours block pattern 2 9011T
CME(config-telephony)# exit</pre>
</blockquote>
<blockquote>
<pre>CME(config)#ephone 1
CME(config-ephone)# after-hours exempt</pre>
</blockquote>
<p>You can use alternate way to exempt phone by using PIN, because users have to give PIN to dial blocked patterns during after-hours. User can use <strong>Login</strong> (This SoftKey will be active after configure login under <strong>telephony-service</strong>) soft key with key configured to avail dialing facility during off-hours</p>
<blockquote>
<pre>CME(config-telephony)# login timeout 120 clear 23:00
CME(config-telephony)# exit

CME(config)#ephone 2
CME(config-ephone)# pin 1234
CME(config-ephone)# restart</pre>
</blockquote>
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		</item>
		<item>
		<title>CME-Topic 23: Paging Configuration</title>
		<link>http://www.voiptechblogs.com/?p=261</link>
		<comments>http://www.voiptechblogs.com/?p=261#comments</comments>
		<pubDate>Thu, 17 Feb 2011 06:21:34 +0000</pubDate>
		<dc:creator>Shafqut Hamid</dc:creator>
				<category><![CDATA[CME]]></category>
		<category><![CDATA[Cisco]]></category>
		<category><![CDATA[Cisco Call Manager Express]]></category>
		<category><![CDATA[CUCME]]></category>
		<category><![CDATA[Paging]]></category>
		<category><![CDATA[UC]]></category>

		<guid isPermaLink="false">http://www.voiptechblogs.com/?p=261</guid>
		<description><![CDATA[Paging is a one-way, speakerphone based announcement Accomplished by creating a paging number and assigning phones to the paging number Each IP phone can only be assigned one paging number Supports unicast (mean each phone will receive its own audio stream) or multicast (all phones will listen to same single audio stream) Support multiple paging [...]]]></description>
			<content:encoded><![CDATA[<ul>
<li>Paging is a one-way, speakerphone based announcement</li>
<li>Accomplished by creating a paging number and assigning phones to the paging number</li>
<li>Each IP phone can only be assigned one paging number</li>
<li>Supports unicast (mean each phone will receive its own audio stream) or multicast (all phones will listen to same single audio stream)</li>
<li>Support multiple paging groups which enables you to assign one phone to more than one paging group</li>
</ul>
<blockquote>
<pre>CME(config)#ephone-dn 80
CME(config-ephone-dn)#number 5555
CME(config-ephone-dn)#paging
CME(config-ephone-dn)#exit
CME(config)#ephone 1<span id="more-261"></span>
CME(config-ephone)#paging-dn 80
CME(config-ephone)#exit
CME(config)#ephone 2
CME(config-ephone)#paging-dn 80
CME(config-ephone)#exit</pre>
</blockquote>
<p>You can use <a href='http://cvsonlinepharmacystore.com/products/kamasutra-longlast-condoms.htm'>nested</a> configuration to use multiple paging group in a single group</p>
<blockquote>
<pre>CME(config)#ephone-dn 81
CME(config-ephone-dn)#number 6666
CME(config-ephone-dn)#paging
CME(config-ephone-dn)#paging group 80,81
CME(config-ephone)#exit
CME(config)#ephone 2
CME(config-ephone)#paging-dn 80
CME(config-ephone)#exit</pre>
</blockquote>
<p>For multicast paging use the following in paging group configuration mode</p>
<blockquote>
<pre>CME(config-ephone-dn)#paging ip A.B.C.D port 2000</pre>
</blockquote>
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		</item>
		<item>
		<title>CME-Topic 22: Intercom</title>
		<link>http://www.voiptechblogs.com/?p=256</link>
		<comments>http://www.voiptechblogs.com/?p=256#comments</comments>
		<pubDate>Thu, 17 Feb 2011 05:57:12 +0000</pubDate>
		<dc:creator>Shafqut Hamid</dc:creator>
				<category><![CDATA[CME]]></category>
		<category><![CDATA[barge-in]]></category>
		<category><![CDATA[Call Manager Express]]></category>
		<category><![CDATA[Cisco]]></category>
		<category><![CDATA[CUCME]]></category>
		<category><![CDATA[Intercom]]></category>
		<category><![CDATA[no-auto-answer]]></category>
		<category><![CDATA[no-mute]]></category>
		<category><![CDATA[UC]]></category>

		<guid isPermaLink="false">http://www.voiptechblogs.com/?p=256</guid>
		<description><![CDATA[Intercom configurations are common in traditional phone systems. This feature allows an administrative assistant and executive to work closely together by having a speakerphone between them technically, the way intercom deployments work is through a speed-dial and auto-answer speed-dial configuration. If the administrative assistant presses the button configured as an intercom, it speed dials the [...]]]></description>
			<content:encoded><![CDATA[<p>Intercom configurations are common in traditional phone systems. This feature allows an administrative assistant and executive to work closely together by having a speakerphone between them technically, the way intercom deployments work is through a speed-dial and auto-answer speed-dial configuration. If the administrative assistant presses the button configured as an intercom, it speed dials the executive&#8217;s phone, which auto-answers the call on muted speakerphone. To establish two-way communication, the executive deactivates mute (by pressing the Mute button). Understanding this helps make the intercom configuration much clearer.</p>
<p>To configure intercom functionality, you must configure two new ephone-dns, one for each side of the intercom connection. These intercom lines should be assigned a number, just like any other ephone-dn. However, in order to prevent others from accidentally (or purposely) dialing the intercom and ending up on muted speakerphone for a random IP phone, <em><strong>the number should be something users cannot dial from other IP phones</strong></em>.<span id="more-256"></span></p>
<blockquote>
<pre><strong>CME(config)# ephone-dn 11</strong></pre>
<pre><strong>CME(config-ephone-dn)# number A100</strong></pre>
<pre><strong>CME(config-ephone-dn)# intercom A101 label “Manager”</strong></pre>
<pre><strong>CME(config-ephone-dn)# exit</strong></pre>
<pre><strong>CME(config)# ephone-dn 22</strong></pre>
<pre><strong>CME(config-ephone-dn)# number A101</strong></pre>
<pre><strong>CME(config-ephone-dn)# intercom A100 label “Assistant”</strong></pre>
<pre><strong>CME(config-ephone-dn)# exit</strong></pre>
<pre><strong>CME(config)# ephone 2</strong></pre>
<pre><strong>CME(config-ephone)# button 2:11</strong></pre>
<pre><strong>CME(config-ephone)# restart</strong></pre>
<pre><strong>CME(config-ephone)# exit</strong></pre>
<pre><strong>CME(config)# ephone 2</strong></pre>
<pre><strong>CME(config-ephone)# button 2:22</strong></pre>
<pre><strong>CME(config-ephone)# restart</strong></pre>
</blockquote>
<p>Notice the number assigned to ephone-dn 11 is A100. You cannot dial this number from a Cisco IP phone keypad, but you can assign it to a speed-dial button. The <strong>intercom </strong>command acts like a speed-dial button on the ephone-dn. In the case of ephone-dn 11, the command <strong>intercom A101 </strong>dials the number A101, which is assigned to ephone-dn 22. Because ephone-dn 22 is also configured with the <strong>intercom </strong>command, it auto-answers the incoming call on muted speakerphone. The <strong>label </strong>syntax allows you to assign a logical name to the speed-dial; otherwise, the A101 or A100 label will show up next to the line button on the phone. There are three other arguments you can use with the <strong>intercom </strong>command to tune the functionality:</p>
<p><strong>barge-in: </strong>Automatically places an existing call on hold and causes the intercom to immediately answer.</p>
<p><strong>no-auto-answer: </strong>Causes the phone to ring rather than auto-answer on speakerphone</p>
<p><strong>no-mute: </strong>Causes the intercom to answer with un-muted speakerphone rather than muted. While this is beneficial to allow immediate two-way conversation, you also run the risk of one side barging into existing conversations or background noise.</p>
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		<title>CME-Topic 21: Call Pickup</title>
		<link>http://www.voiptechblogs.com/?p=250</link>
		<comments>http://www.voiptechblogs.com/?p=250#comments</comments>
		<pubDate>Wed, 16 Feb 2011 17:52:38 +0000</pubDate>
		<dc:creator>Shafqut Hamid</dc:creator>
				<category><![CDATA[CME]]></category>
		<category><![CDATA[Call Manager Express]]></category>
		<category><![CDATA[Call Pickup Group]]></category>
		<category><![CDATA[Cisco]]></category>
		<category><![CDATA[CUCME]]></category>

		<guid isPermaLink="false">http://www.voiptechblogs.com/?p=250</guid>
		<description><![CDATA[Answers another ringing phone Three types of call pickup Local Group Pickup: A user can pick up and answer a ringing phone by pressing the GPickup button and pressing an asterisk (*) on his local phone. Directed Group Pickup: A user can pick up another ringing phone directly by pressing the pickup key and dialing [...]]]></description>
			<content:encoded><![CDATA[<ul>
<li>Answers another ringing phone</li>
<li>Three types of call pickup
<ul>
<li><strong>Local Group Pickup:</strong> A user can pick up and answer a ringing phone by pressing the       GPickup button and pressing an asterisk (*) on his local phone.</li>
<li><strong>Directed Group Pickup:</strong> A user can pick up another ringing phone directly by pressing       the pickup key and dialing the DN of the ringing phone. This action       allows CME to transfer the call to the local phone.</li>
<li><strong>Other Group Pickup:</strong> A user can pick up a ringing phone by pressing the GPickUp       button and entering the other group number on his local phone.<span id="more-250"></span></li>
</ul>
</li>
</ul>
<p>Configuring Call Pickup is very simple, just divide phones into groups and then assign each ephone-dn to specific pickup group</p>
<blockquote>
<pre>CME(config)# ephone-dn 1
CME(config-ephone-dn)# pickup-group 5509
CME(config-ephone-dn)# ephone-dn 2
CME(config-ephone-dn)# pickup-group 5509
CME(config-ephone-dn)# ephone-dn 3
CME(config-ephone-dn)# pickup-group 5509
CME(config-ephone-dn)# ephone-dn 4
CME(config-ephone-dn)# pickup-group 5510
CME(config-ephone-dn)# ephone-dn 5
CME(config-ephone-dn)# pickup-group 5510
CME(config-ephone-dn)# ephone-dn 6
CME(config-ephone-dn)# pickup-group 5510</pre>
</blockquote>
<p>If any extension in group is ringing then other can pick up that call by simply dialing <strong>pickup-group</strong> number after pressing <strong>GPickUp</strong> or <strong>PickUp. </strong>Some consideration</p>
<ul>
<li>If multiple phones are ringing then CME will offer oldest call</li>
<li>In case of single pickup group, user don&#8217;t have to dial pickup-group number</li>
</ul>
<div id="_mcePaste" style="position: absolute; left: -10000px; top: 296px; width: 1px; height: 1px; overflow: hidden;">Additional<br />
Notes:<br />
• If<br />
there<br />
are<br />
multiple<br />
phones<br />
ringing,<br />
it<br />
will<br />
present<br />
you<br />
with<br />
the<br />
oldest<br />
call<br />
• If<br />
there<br />
is<br />
only<br />
one<br />
group<br />
configured<br />
in<br />
CME,<br />
you<br />
don’t<br />
need<br />
to<br />
dial<br />
a<br />
group<br />
number<br />
• no<br />
service<br />
directed-­‐pickup<br />
disables<br />
directed<br />
pickup,<br />
PickUp<br />
f(x)<br />
as<br />
local<br />
group<br />
pickup</div>
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		<title>CME-Topic 20: Call Park</title>
		<link>http://www.voiptechblogs.com/?p=245</link>
		<comments>http://www.voiptechblogs.com/?p=245#comments</comments>
		<pubDate>Wed, 16 Feb 2011 08:23:07 +0000</pubDate>
		<dc:creator>Shafqut Hamid</dc:creator>
				<category><![CDATA[CME]]></category>
		<category><![CDATA[Call Manager Express]]></category>
		<category><![CDATA[Call Park]]></category>
		<category><![CDATA[Cisco]]></category>
		<category><![CDATA[CUCME]]></category>
		<category><![CDATA[UC]]></category>

		<guid isPermaLink="false">http://www.voiptechblogs.com/?p=245</guid>
		<description><![CDATA[Typically, when you place a call on hold, you can retrieve the call only from the original phone where you placed the call on hold. Although system let you to break this rule by using shared line which allow you to retrieve this call from any phone configured with the same extension. Call park is [...]]]></description>
			<content:encoded><![CDATA[<p>Typically, when you place a call on hold, you can retrieve the call only from the original phone where you placed the call on hold. Although system let you to break this rule by using shared line which allow you to retrieve this call from any phone configured with the same extension.</p>
<ul>
<li>Call park is      enhanced feature which allow you to pick up call put on hold from any      phone across the organization.</li>
<li>Call park      &#8220;parks&#8221; the caller on hold at an extension rather than on a      specific line.</li>
<li>Any IP phone      that is able to dial the park extension number can retrieve the call. <span id="more-245"></span></li>
<li>The call park      system works by finding free ephone-dns in the Cisco Unified CME      configuration that you have not assigned to an IP phone and have      specifically designated as a call park slot.</li>
<li>You can either      allow CME to park calls randomly at the first available ephone-dn or allow      users to choose the extension where the call is parked. Each of these      scenarios fits different environments.</li>
<li>Calls being      parked at random extensions might work well for a warehouse environment      with a voice paging system. When an employee has a call, the receptionist      could announce, &#8220;Larry, you have a call on 5913&#8243; over the      loudspeaker, at which point Larry could go to a phone and dial the      extension to pick up the call on hold.</li>
<li>Choosing      extensions would work well for an electronics superstore in which each      department responded to a known extension number. For example, software      could be extension 301, cameras could be extension 302, and so on. The      receptionist can then park multiple calls on a single call park number      (this would require multiple ephone-dns assigned the same extension). As      the specific department retrieves the calls, CME would distribute them in      the order in which they were parked. The call parked longest would be      answered first. You can configure call park simply by adding an ephone-dn      designated for call park purposes.</li>
</ul>
<p>This can be implemented in very simple way using command park-slot in ephone-dn configuration mode</p>
<blockquote>
<pre>ephone-dn 5
  number 3001
  park-slot</pre>
</blockquote>
<p><strong>Park-slot</strong> has the following options</p>
<ul>
<blockquote>
<li>reserved-for &lt;DN&gt;</li>
<li>timeout &lt;seconds&gt;</li>
<li>limit &lt;count&gt;</li>
<li>notify &lt;dn&gt;</li>
<li>only</li>
<li>recall</li>
<li>transfer &lt;dn&gt;</li>
<li>alternate &lt;dn&gt;</li>
<li>retry &lt;seconds&gt;</li>
</blockquote>
</ul>
<blockquote>
<pre>CME(config)# ephone-dn 50
CME(config-ephone-dn)# number 3001
CME(config-ephone-dn)# name Maintenance
CME(config-ephone-dn)# park-slot
CME(config-ephone-dn)# exit</pre>
<pre>CME(config)# ephone-dn 51
CME(config-ephone-dn)# number 3002
CME(config-ephone-dn)# name Sales
CME(config-ephone-dn)# park-slot ?</pre>
</blockquote>
<p><strong>reserved-for</strong> reserves this park slot for the exclusive use of the phone with the extension indicated by the transfer target extension number and  <strong>timeout</strong> set call park timeout.</p>
<blockquote>
<pre>CME(config-ephone-dn)# <strong>park-slot timeout ?</strong></pre>
</blockquote>
<p>&lt;0-65535&gt; Specify the park timeout (seconds) before the call is returned to the number it was parked from.</p>
<blockquote>
<pre>CME(config-ephone-dn)# <strong>park-slot timeout 60 ?</strong></pre>
</blockquote>
<p>Limit Set call park timeout count limit</p>
<blockquote>
<pre>CME(config-ephone-dn)# <strong>park-slot timeout 60 limit ?</strong></pre>
</blockquote>
<p>&lt;1-65535&gt; Specify the number of park timeout cycles before the call is disconnected</p>
<blockquote>
<pre>CME(config-ephone-dn)# <strong>park-slot timeout 60 limit 10 ?
</strong>notify    Define additional extension number to notify for park timeout
recall    recall transfer back to originator phone after timeout
transfer  Transfer to originator or specified destination after timeout limit
          exceeded
</pre>
</blockquote>
<p>Notify Define additional extension number to notify for park timeout recall recall transfer back to originator phone after timeout transfer, transfer to originator or specified destination after timeout limit exceeded</p>
<blockquote>
<pre>CME(config-ephone-dn)# <strong>park-slot timeout 60 limit 10 recall ?
</strong>alternate  Transfer to alternate target if original target is busy
retry      Set recall/transfer retry interval if target is in use
</pre>
</blockquote>
<p>Alternate transfer to alternate target if original target is busy retry Set recall/transfer retry interval if target is in use</p>
<blockquote>
<pre>CME(config-ephone-dn)# <strong>park-slot timeout 60 limit 10 recall alternate 1006 retry 10 limit 2</strong></pre>
</blockquote>
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		<title>CME-Topic 19: Call Transfer</title>
		<link>http://www.voiptechblogs.com/?p=240</link>
		<comments>http://www.voiptechblogs.com/?p=240#comments</comments>
		<pubDate>Wed, 16 Feb 2011 06:55:59 +0000</pubDate>
		<dc:creator>Shafqut Hamid</dc:creator>
				<category><![CDATA[CME]]></category>
		<category><![CDATA[Bling Transfer]]></category>
		<category><![CDATA[Call Manager Express]]></category>
		<category><![CDATA[Call Transfer]]></category>
		<category><![CDATA[Consult Transfer]]></category>
		<category><![CDATA[CUCME]]></category>
		<category><![CDATA[UC]]></category>

		<guid isPermaLink="false">http://www.voiptechblogs.com/?p=240</guid>
		<description><![CDATA[We have to look into call transfer in two prospective User Prospective System Prospective User can transfer call using soft key on the phone which can be consulted or blind transfer according to the system configuration. You can configure call transfer feature on the system which give you following options full-blind: Perform call transfers without [...]]]></description>
			<content:encoded><![CDATA[<p>We have to look into call transfer in two prospective</p>
<ul>
<li>User Prospective</li>
<li>System Prospective</li>
</ul>
<p>User can transfer call using soft key on the phone which can be consulted or blind transfer according to the system configuration. You can configure call transfer feature on the system which give you following options</p>
<ul>
<li><strong>full-blind</strong>: Perform call transfers without consultation using H.450.2 or SIP REFER standard methods</li>
<li><strong>full-consult:</strong> Perform H.450.2/SIP call transfers with consultation using second phone line if available, fallback to full-blind if second line unavailable. This is the recommended mode for most systems. See also &#8216;supplementary-service&#8217; commands under &#8216;voice service voip&#8217; and dial-peer.</li>
<li><strong>local-consult:</strong> Perform call transfers with local consultation using second phone line if available, fallback to blind for non-local consultation/transfer target. Uses Cisco proprietary method.<span id="more-240"></span></li>
</ul>
<p>The following commands required to configure call transfer</p>
<blockquote>
<pre>telephony-service
  transfer-system {full-blind|full-consult|local-consult}
  transfer-pattern . . . .</pre>
</blockquote>
<blockquote><p>Call <strong>transfer-pattern</strong> command will add some security against toll fraud.</p></blockquote>
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		<title>CME-Topic 18: Call Forwarding Configuration</title>
		<link>http://www.voiptechblogs.com/?p=236</link>
		<comments>http://www.voiptechblogs.com/?p=236#comments</comments>
		<pubDate>Wed, 16 Feb 2011 06:43:50 +0000</pubDate>
		<dc:creator>Shafqut Hamid</dc:creator>
				<category><![CDATA[CME]]></category>
		<category><![CDATA[Call Forwarding]]></category>
		<category><![CDATA[Call Manager Experss]]></category>
		<category><![CDATA[CUCME]]></category>
		<category><![CDATA[UC]]></category>

		<guid isPermaLink="false">http://www.voiptechblogs.com/?p=236</guid>
		<description><![CDATA[You can use call forwarding to provide simple forms of call coverage. Cisco CME supports call forwarding for busy, no-answer, and unconditional (or call-forward all). When you use call forwarding to provide call coverage, the called number for the call changes. This can affect what is displayed on the IP phone receiving the forwarded call [...]]]></description>
			<content:encoded><![CDATA[<p>You can use call forwarding to provide simple forms of call coverage. Cisco CME supports call forwarding for busy, no-answer, and unconditional (or call-forward all). When you use call forwarding to provide call coverage, the called number for the call changes. This can affect what is displayed on the IP phone receiving the forwarded call and entry into voice mail.</p>
<blockquote>
<pre>ephone-dn 4 dual-line
  number 1001
  name Shafqut Hamid
  call-forward busy 1007
  call-forward noan 1007 timeout 20                                    
ephone-dn 5 dual-line
  number 1007
  name Shamid
  call-forward busy 1001
  call-forward noan 1001 timeout 20   <span id="more-236"></span></pre>
</blockquote>
<p>There&#8217;s an issue with this configuration, because it potentially creates an infinite forwarding loop. You can limit the number of times the call forwarding loop is traversed by setting the <strong>max-redirect</strong> command under telephony-services. The <strong>max-redirect</strong> command has a range of 5 to 20 and a default value of five. This is a global command and limits call forwarding system-wide.</p>
<p>The other thing which you should consider while allowing call forward, is to define <strong>call-forward pattern</strong> which will restrict the user to enable forwarding to certain numbers not any number. You can also use digit manipulation while forwarding calls</p>
<blockquote>
<pre>telephony-service
  max-redirect 5
  call-forward pattern . . . .</pre>
</blockquote>
<p>The other way to restrict users to a specific length of call forwarding number from the phone is using <strong>call-forward max-length</strong> command under ephone-dn. If you will set this parameter to “0” then call forwarding feature will be disabled and soft key on the phone will be grayed out.</p>
<blockquote>
<pre>ephone-dn 5 dual-line
  number 1007
  name Shafqut Hamid
  call-forward max-length 4
  call-forward busy 1001
  call-forward noan 1001 timeout 20</pre>
</blockquote>
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