these are the steps for MVA and SNR<br><br>Step 1.<br>Go to the UCM administration page<br>-Select the System TAB<br>-Select the Service parameters TAB<br>-For the server field, select the hostname/ip address of the publisher<br>
-For the service field, select Cisco Callamanager<br>-Under Clusterwide Parameters (System-Mobility)<br>-Set the “Enable Enterprise Feature Access” parameter to True<br>-Set the “Enable Mobile Voice Access” parameter to True<br>
-Proceed to Enter the Mobile Voice Access number that you are going to use in our case will be 3511<br>-Set the “Matching Caller ID with Remote Destination” parameter to “complete match” (by selecting this parameter the gateway will look for a “complete match” of digits coming from the PSTN and it will decide if is a complete match of an existing Destination Number then the user will be prompted for it’s CCM user PIN number, if is not a complete match then the user will be prompted to enter its Remote Destination Number followed by the pound sign.<br>
-In the other hand if you set the “Matching Caller ID with Remote Destination” parameter to “Partial Match” then the gateway is suppose to match the number of digits that you decide against an existing Destination number but will ONLY WORK WITH A DEDICATED H.323 GATEWAY and it will not work using a hairpinning set up which is our case.<br>
Note: When setting the “Matching Caller ID with Remote Destination” parameter to “Partial Match” in this setup (a hairpinning set up) the Remote Destination user calling from a device that match a remote destination number will experience a silence when dialing our MVA DID<br>
-Click save.<br>Step 2<br>-Go to the Media Resources TAB<br>-Select the Mobile Voice Access TAB<br>-On the Mobile Voice Access Directory Number field enter 3511<br>-On the Mobile voice Access Partition select a partition that is an accessible partition<br>
-Proceed to select the desire Locale.<br>Step 3<br>-Telnet or SSH to the gateway that you are going to use.<br>-Under config mode proceed to enter the following command<br>!<br>Voice service voip<br>Allow-connections h323 to h323<br>
!<br>Voice class h323 1<br>H225 timeout tcp establish 3<br>!<br>Application<br>Service cmm http://[ip address of your publisher]:8080/ccmivr/pages/IVRMainpage.vxml <br><br><br>Step 4<br>-To verify that the vxml script along with the correct english locale was successfully loaded into our gateway issue the following commands<br>
Yourgateway#sh call appl voic cmm<br>A sign that the vxml script and our English locate is properly configured and loaded will be<br><assign name=”srcdir” expr=”`en_US`”/><br>*if you are seeing the following code<br>
<assign name=”`srcdir`” expr=”`null`”/><br>Check the cmm url characters under application, if your url is incorrect proceed to make the necessary changes with the correct url also, visit the UCM administration page, select the Media Resources TAB , select Mobile Voice Access and click save, this will push out the vxml script to the voice gateway, you will also need to reboot your gateway, in order for the script to show correctly in your Voice Gateway.<br>
Step 5<br>We will now proceed to create our Dial-Peers<br>!<br>dial-peer voice 5311 voip<br>service cmm<br>session target ipv4:[ip address of the publisher]<br>incoming called-number 5311 <------our incoming DID from the PSTN<br>
codec g711ulaw<br>!<br>dial-peer voice 3511 voip<br>preference 1<br>destination-pattern 3511 <------our Mobile Voice Access DN<br>voice-class h323 1<br>session target ipv4:[ip address of the publisher]<br>dtmf-relay h245-alphanumeric<br>
codec g711ulaw<br>no vad<br>!<br>dial-peer voice 3512 voip<br>preference 2<br>destination-pattern 3511 <------our Mobile Voice Access DN<br>voice-class h323 1<br>session target ipv4:[ip address of the subscriber]<br>dtmf-relay h245-alphanumeric<br>
codec g711ulaw<br>no vad<br>!<br>dial-peer voice 5312 voip<br>service cmm<br>session target ipv4:[ip address of the subscriber]<br>incoming called-number 5311 <------our incoming DID from the PSTN<br>codec g711ulaw<br>
!<br>Step 6<br>We will now proceed to create our h323 gateway in our UCM.<br>-Go to the UCM Administration page<br>-Select the Device TAB<br>-Select Gateway<br>-Select Add New<br>-For our gateway Type parameter select H.323 Gateway, click Next<br>
- Enter the MGCP gateway ip address for the Device Name* parameter<br>-Proceed to the Select the desire Device pool<br>-Proceed to select the Correct Media Resource Group List, in my case I use <None><br>-Proceed to select the appropriate Location<br>
-Proceed to check the Media Termination Point Required* field.<br>-Proceed to select the appropriate parameters for the Call routing information – Inbound Calls fields.<br>-For testing and for this example I am using the following parameters<br>
Calling Search Space [Local-CSS]<br>AAR Calling Search Space [Local-CSS]<br>-Select Save. <br><br><br>Step 7<br>-For this example I will be using a Route Group Called “RG-Local” which will contain our H323 gateway IP address .<br>
-For this example I have created a Route List called “RL_MVA_Devices” which have the RG-Local selected as a Route Group.<br>Step 8<br>-We will now proceed to create a route pattern that matches our inbound DID from the PSTN.<br>
-For the route pattern* field we will enter our inbound DID number which for this document is 5311<br>-Proceed to select the appropriate route partition, for this example I have selected an open/default partition <None><br>
-For the Gateway/Route List* Field we will select “RL_MVA_Devices”<br>-Route this pattern<br>-Call Clasification* “OffNet”<br>-Provide Outside Dial Tone<br>-Select Save.<br>Step 9<br>Finally we will need to add our Mobile-Voice-Access_RP partition to the Calling Search Space that we use in our H323 gateway<br>
-Go to the UCM Administration page<br>-Go to the Call Routing Tab<br>-Go to the Class of Control Tab<br>-Select Calling Search Space<br>-Look for the Calling Search Space that you use for your H.323 gateway, in our case I use “Local-CSS”<br>
-Proceed to add the appropriate Mobile Voice Access Partition to the selected partitions for the “Local-CSS”, in my case I will add the Mobile-Voice-Access-RP partition and I will put it on the 3rd level.<br>First my 911_PT<br>
Second my Internal_PT<br>Third it will be our Mobile-Voice-Access_RP partition<br>-Click save.<br>At this point you should be able to dial the DID used to access the MVA, depending on your call manager service parameters configured, Unified Communications Manager /MVA IVR will answer the call and you will be prompted for enter your Remote Destination number followed by the pound sign, then you will be prompted to enter your PIN, once authenticated you will be able to start using the MVA feature at which point you can start dialing as if you were using an internal IP Phone.<br>
While you are logged into MVA and during a call you can use the following default dtmf codes to.<br>Put a call in hold *81<br>Put a call in an exclusive hold *82<br>To resume a call *83<br>To transfer a call *84<br>To conference a party *85<br>
*************************************************************************************<br>Notes: You can Multiple DID’s and Gateways for MVA, what you will need to do is for each DID/ GW pair, create a route pattern matching the did and route that call to its corresponding gateway, each gateway configuration will be similar to the one we configured. The incoming called-number statement has to match the DID local to that Site. The second dial-peer which matches destination-pattern XXXX will be same on all routers.<br>
I have successfully reproduced the same configuration on a 3825 Voice Gateway located in Europe using an E1 with framing NO-CRC4, under mgcp conditions, with IOS version 12.4(11)xw5<br>For troubleshooting you may use<br>CCM SDI and SDL traces, MVA traces<br>
Debug isdn q931, debug voice ccapi inout, debug voice application vxml application, debug voice application vxml all<br>For more information please the following documents<br><a href="http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/6_0_1/ccmfeat/fsmobmgr.pdf">http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/6_0_1/ccmfeat/fsmobmgr.pdf</a><br>
<a href="http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/6_0_1/ccmfeat/fsmobmgr.html#wp1144476">http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/6_0_1/ccmfeat/fsmobmgr.html#wp1144476</a> <br><br><div class="gmail_quote">
2009/7/9 Moataz Mamdouh <span dir="ltr"><<a href="mailto:moataz_mmdh@yahoo.com">moataz_mmdh@yahoo.com</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<table border="0" cellpadding="0" cellspacing="0"><tbody><tr><td style="font-family: inherit; font-style: inherit; font-variant: inherit; font-weight: inherit; font-size: inherit; line-height: inherit; font-size-adjust: inherit; font-stretch: inherit;" valign="top">
<p style="margin-bottom: 0.0001pt; line-height: normal;"><span style="font-size: 12pt; font-family: "Times New Roman","serif";">Dial-peer voice
1000 pots</span></p>
<p style="margin-bottom: 0.0001pt; line-height: normal;"><span style="font-size: 12pt; font-family: "Times New Roman","serif";">incoming
called-number 202</span></p>
<p style="margin-bottom: 0.0001pt; line-height: normal;"><span style="font-size: 12pt; font-family: "Times New Roman","serif";">service ccm</span></p>
<p style="margin-bottom: 0.0001pt; line-height: normal;"><span style="font-size: 12pt; font-family: "Times New Roman","serif";">no digit-strip</span></p>
<p style="margin-bottom: 0.0001pt; line-height: normal;"><span style="font-size: 12pt; font-family: "Times New Roman","serif";">direct-inward-dial</span></p>
<p style="margin-bottom: 0.0001pt; line-height: normal;"><span style="font-size: 12pt; font-family: "Times New Roman","serif";"> </span></p>
<p style="margin-bottom: 0.0001pt; line-height: normal;"><span style="font-size: 12pt; font-family: "Times New Roman","serif";">dial-peer voice
2000 voip</span></p>
<p style="margin-bottom: 0.0001pt; line-height: normal;"><span style="font-size: 12pt; font-family: "Times New Roman","serif";">destination-pattern
202</span></p>
<p style="margin-bottom: 0.0001pt; line-height: normal;"><span style="font-size: 12pt; font-family: "Times New Roman","serif";">session target
ipv4: 192.168.1.100</span></p>
<p style="margin-bottom: 0.0001pt; line-height: normal;"><span style="font-size: 12pt; font-family: "Times New Roman","serif";">codec g711ulaw</span></p>
<p style="margin-bottom: 0.0001pt; line-height: normal;"><span style="font-size: 12pt; font-family: "Times New Roman","serif";">no vad</span></p>
<br><br>--- On <b>Thu, 7/9/09, Moataz Mamdouh <i><<a href="mailto:moataz_mmdh@yahoo.com" target="_blank">moataz_mmdh@yahoo.com</a>></i></b> wrote:<br><blockquote style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;">
<br>From: Moataz Mamdouh <<a href="mailto:moataz_mmdh@yahoo.com" target="_blank">moataz_mmdh@yahoo.com</a>><div class="im"><br>Subject: RE: [cisco-voip] Mobile Voice Access<br></div>To: "Bob Fronk" <<a href="mailto:bob@btrfronk.com" target="_blank">bob@btrfronk.com</a>><br>
Cc: "CISCO VOIP LIST" <<a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a>><br>Date: Thursday, July 9, 2009, 2:09 PM<div><div></div><div class="h5"><br><br><div><table border="0" cellpadding="0" cellspacing="0">
<tbody><tr><td style="font-family: inherit; font-style: inherit; font-variant: inherit; font-weight: inherit; font-size: inherit; line-height: inherit; font-size-adjust: inherit; font-stretch: inherit;" valign="top">Sample Dial Peer Configuration:<br>
<br><br>--- On <b>Thu, 7/9/09, Bob Fronk <i><<a href="mailto:bob@btrfronk.com" target="_blank">bob@btrfronk.com</a>></i></b> wrote:<br><blockquote style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;">
<br>From: Bob Fronk <<a href="mailto:bob@btrfronk.com" target="_blank">bob@btrfronk.com</a>><br>Subject: RE: [cisco-voip] Mobile Voice Access<br>To: "Moataz Mamdouh" <<a href="mailto:moataz_mmdh@yahoo.com" target="_blank">moataz_mmdh@yahoo.com</a>><br>
Date: Thursday, July 9, 2009, 8:32 AM<br><br><div>
<div>
<p><span style="color: rgb(31, 73, 125);">I
have the two dial peers configured one for the incoming DID and one for the
destination.</span></p>
<p><span style="color: rgb(31, 73, 125);"> </span></p>
<p><span style="color: rgb(31, 73, 125);">Do
you have a sample of the dial peers to see if I did them correctly?</span></p>
<p><span style="color: rgb(31, 73, 125);"> </span></p>
<p><b><span style="color: rgb(31, 73, 125);">--</span></b></p>
<p><b><span style="color: rgb(31, 73, 125);">Bob</span></b><span style="font-size: 11pt; color: rgb(31, 73, 125);"></span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p><span style="color: rgb(31, 73, 125);"> </span></p>
<div style="border-style: solid none none; border-color: rgb(181, 196, 223); border-width: 1pt medium medium; padding: 3pt 0in 0in;">
<p><b><span style="font-size: 10pt;">From:</span></b><span style="font-size: 10pt;"> Moataz Mamdouh
[mailto:<a href="mailto:moataz_mmdh@yahoo.com" target="_blank">moataz_mmdh@yahoo.com</a>] <br>
<b>Sent:</b> Thursday, July 09, 2009 8:27 AM<br>
<b>To:</b> Bob Fronk<br>
<b>Subject:</b> Re: [cisco-voip] Mobile Voice Access</span></p>
</div>
<p> </p>
<table border="0" cellpadding="0" cellspacing="0">
<tbody><tr>
<td style="padding: 0in;" valign="top">
<p>check the 2 dial-peer configuration<br>
one have to be matched with destination-pattern and the other must match
using incoming called-number<br>
<br>
--- On <b>Thu, 7/9/09, Bob Fronk <i><<a href="mailto:bob@btrfronk.com" target="_blank">bob@btrfronk.com</a>></i></b> wrote:</p>
<p style="margin-bottom: 12pt;"><br>
From: Bob Fronk <<a href="mailto:bob@btrfronk.com" target="_blank">bob@btrfronk.com</a>><br>
Subject: Re: [cisco-voip] Mobile Voice Access<br>
To: "Moataz Mamdouh" <<a href="mailto:moataz_mmdh@yahoo.com" target="_blank">moataz_mmdh@yahoo.com</a>><br>
Date: Thursday, July 9, 2009, 6:20 AM</p>
<div>
<p>Yes, downloaded the XML and able to use show to see it on
the router. I am not able to call the application internally.</p>
<p>Bob Fronk<br>
<a href="mailto:bob@btrfronk.com" target="_blank">bob@btrfronk.com</a><br>
Sent from my Blackberry.<br>
<br>
Sent via Blackberry</p>
<div style="text-align: center;" align="center">
<hr align="center" size="2" width="100%">
</div>
<p><b>From</b>: Moataz Mamdouh <br>
<b>Date</b>: Thu, 9 Jul 2009 03:23:14 -0400<br>
<b>To</b>: Bob Fronk<<a href="mailto:bob@btrfronk.com" target="_blank">bob@btrfronk.com</a>><br>
<b>Subject</b>: Re: [cisco-voip] Mobile Voice Access</p>
<table border="0" cellpadding="0" cellspacing="0">
<tbody><tr>
<td style="padding: 0in; font-style: inherit; font-variant: inherit; font-weight: inherit; font-size: inherit; line-height: inherit; font-size-adjust: inherit; font-stretch: inherit;" valign="top">
<p><span>Did u
download the Voice XML application from CCM?<br>
are you able to call the application and hear the script?<br>
<br>
--- On <b>Wed, 7/8/09, Bob Fronk <i><<a href="mailto:bob@btrfronk.com" target="_blank">bob@btrfronk.com</a>></i></b> wrote:</span></p>
<p style="margin-bottom: 12pt;"><span><br>
From: Bob Fronk <<a href="mailto:bob@btrfronk.com" target="_blank">bob@btrfronk.com</a>><br>
Subject: [cisco-voip] Mobile Voice Access<br>
To: "<a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a>"
<<a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a>><br>
Date: Wednesday, July 8, 2009, 3:59 PM</span></p>
<div>
<div>
<p>I have been struggling with setting up Mobile Voice Access. If
someone has this feature setup and working, I would appreciate a config
sample.</p>
<p> </p>
<p>Using UCM 6.1.2.1000-13</p>
<p> </p>
<p>Incoming is PRI to VWIC2-1MFT-T1 in a 3825 router running 12.4(20)T1.</p>
<p> </p>
<p>The setup is currently working for incoming / outgoing calls and we
would like to add the Mobile Voice Access. Mobile Connect is
already configured and working.</p>
<p> </p>
<p>Essentially, I have followed this article, but have had no luck
getting this to work. It is as if the number is not assigned.
We get “we’re sorry, the number you have called is not in service”.</p>
<p> </p>
<p><a rel="nofollow" href="http://edge.networkworld.com/community/comment/reply/32666/189184" target="_blank"><u><span style="color: blue;">http://edge.networkworld.com/community/comment/reply/32666/189184</span></u></a></p>
<p> </p>
<p>I would appreciate any dial-peer configs that include a 10 digit DID
number and any CUM config assistance.</p>
<p> </p>
<p>Thanks for your help.</p>
<p> </p>
<p><b>Bob</b></p>
<p> </p>
<p> </p>
<p> </p>
<p> </p>
</div>
</div>
<p style="margin-bottom: 12pt;"><span><br>
-----Inline Attachment Follows-----</span></p>
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<p><span style="font-size: 10pt;"> </span></p>
</div>
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