Hi MJ,
I did some research on this since I've been configuring MVA for a while but
have had some questions about underlying architecture. Here's some
responses to your info plus some of my findings.
1) If the MVA DID is in line with your standard DID range for the site,
why not just piggy back on the existing CUCM dial-peers instead of creating
a new one just for MVA. Say Site B for example with a 3XXX extension
range, you could use the CUCM dial-peer:
dial-peer voice 3000 voip
destination pattern 3...$
session target ipv4:10.10.210.11
no vad
voice-class codec 1
voice-class h323 1
dtmf-relay h245-alpha
incoming called-number .
2) Looks good. I change my service name to MVA since I think there's a
typo somewhere in the CUCM pages where I copy/paste from but as long as the
names match up between the service and dial-peer, no worries.
3) Right, I use the same to chop DID's to local extensions:
voice translation-rule 1
rule 1 /.+\(....\)/ /\1/
voice translation-profile PSTN
translate called 1
voice-port 0/0/0:23
translation-prof in PSTN
4) Here, I do not use partial match. I've heard from a truly reliable
source that there is some buggi-ness with this particular version of CUCM
and partial matches. In the end, I think it's less thinking and moving
parts if you just use a full match anyway. Just my POV on this one. Also,
the 'Mobile Voice Access Number' in the CCM service parameters isn't used
for VXML MVA. From what I understand, this parameter is for Mobile
Communicator. I've been through the SRND and several other pages and
cannot pin the exact meaning of the parameter, but in the SRND
configuration guide for VXML MVA, it cruises right over this parameter so I
believe it's safe to leave at default (blank).
5) I've never had a specific requirement for this. I'd say don't waste the
time setting it up if it's not required but if anyone has good reason to
think it should be configured, lemme know.
6) Agreed
7) Be sure to set the re-routing CSS on the RDP (if SNR is required).
CSS = MVA dialing
Rerouting CSS = SNR dialing
Also, just as a heads up you shouldn't use SLRL for SNR as it will use
the RG of the calling party (say HQ phone 2) so the call would try to go
out HQ GW. Make sure to create a route list for SB (if SNR is at site B)
and point the SNR pattern to it so it goes out the SB gateway as a local
call.
8) I use the full number here.
9) I never set this and have not had any issues with MVA/SNR. The CUCM
help file says its for CDR usage. Anyone know how/if this setting impacts
MVA/SNR?
10) Agreed
About your questions, I'm not clear on #1. Like I mentioned, I use full
match and don't do any manipulation of the calling number for SNR/MVA
questions. For #2, you haven't mentioned the Media Resources->Mobile Voice
Access->Mobile Voice Access Directory Number. Unlike the Service Parameter
Setting, this is the number that's used for calls from the H323 GW to CUCM.
Here are some debugs from a call into MVA from my lab. The process is
that the CUCM instructs the GW to play prompts and collect digits based on
the DTMF input from the caller. The call was placed from my configured
Remote Destination so I'm not prompted to enter my RD Number:
-------GET PIN------------
Here the gateway prompts to enter my pin to authenticate
<vxml version="2.0">
<form id="Pin">
<grammar type="application/grammar+regex">.</grammar>
<field name="pin" type="digits?minlength=1;maxlength=20">
<prompt>
<audio s
-------GET FUNCTION------------
Here the GW asks what I'd like to do ("Press 1 to place a call")
<vxml version="2.0">
<form id="GetFunctionSel">
<grammar type="application/grammar+regex">.</grammar>
<field name="funcsel" type="digits?length=1">
<pro
-------GET DIALED DIGITS------------
Here the GW plays the prompt to enter the digits followed by pound
<vxml version="2.0">
<form id="Getdialno">
<grammar type="application/grammar+regex">.</grammar>
<field name="dialno" type="digits?minlength=1;maxlength=50">
<prompt>
-------TRANSFER CALL------------
Here is where the gateway hands the call to CUCM.
<vxml version="2.0">
<form id="Transfer">
<transfer name="mycall"
dest="phone://3300" <<<<<<<<<<< This is the number that is
set within the CUCM Media Resource->MVA section
cisco-ani="phone://4088397263" <<<<<<<<<<<< ANI AKA Remote
Destination number
cisco-rdn="phone://2002" <<<<<<<<<< My dialed digits
cisco-rdntype="0"
cisco-rdnp
So the lesson for me was the the Media Resource->MVA Dir Num does not have
to match up with the actual MVA DID whatsoever. The requirement is that
the H323 GW has a dial-peer with a destination-pattern that matches the
number provided by CUCM to the GW in the dest="phone://3300" field and also
that the H323 GW CSS within CUCM has access to the PT that the MVA Dirn was
assigned to. In the case of using the existing CUCM outbound dial-peer
from above, it's a match. Here's an example of a failed transfer:
-------FAILED TRANSFER------------
<vxml version="2.0">
<form id="Transfer">
<transfer name="mycall"
dest="phone://33"
cisco-ani="phone://4088397263"
cisco-rdn="phone://2002"
cisco-rdntype="0"
cisco-rdnpla
In this case the 'dest' field was changed to just '33' within CUCM Media
Resource->MVA and the call failed. But, after I copied my CUCM dial-peer
on the GW and changed the destination-pattern to 33$, it worked.
I hope this helps, and thanks for prompting me to stop procrastinating on
researching MVA!
Marty
On Wed, Sep 18, 2013 at 12:25 AM, sanity insanity <
networksanitytoinsan...@gmail.com> wrote:
>
> Hi Guys,
>
>
>
> I have been trying to find the right way of configuring MVA. Below is my
> configuration....
>
>
> Details:
> =============
>
> My config is following....
>
> 1) The dial-peers are set in the following way
>
> dial-peer voice 102 voip
> preference 2
> destination-pattern 3300
> session target ipv4:<ip address of the CUCM Pub> dtmf-relay
> h245-alphanumeric
> codec g711ulaw no vad !
> dial-peer voice 3300 pots
> service cmm
> incoming called-number 3300
> no digit-strip
>
>
> 2) here is the MVA service url
> !
> application
> service cmm http://<ip address of the CUCM
> Pub>:8080/ccmivr/pages/IVRMainpage.vxml
> !
>
>
> 3) I am stripping 3033300 coming from pstn to last 4 digits using a
> translation-rule on the voice-port level . That is 3033300 becomes 3300
> when it
> reaches CUCM.
>
>
> 4) On CUCM in the service parameters...
>
> Enable Mobile Voice access is set to True Mobile voice access number is
> 3300
> Matching caller id with Remote Destination is Partial Match Number of
> digits of
> Caller ID Partial Match is 7
>
> 5) The Mobility softkey has been added for "on hold" and "connected" at
> the
> softkey template level and applied to the phone ( SB PH1)
>
>
> 6)At the User SB phone 1 I have enabled "Enable Mobility" and "Enable
> Mobile
> Voice Access"
> also selected the MAC address of the phone
>
>
> 7) Created a Remote Dest profile and selected user id of sb ph1 and the
> correct
> calling search space for the phone
>
>
> 8) Added a Remoted Destination number of 5252222
>
>
> 9) Also went to device > phone and selected the Owner User ID of SB Ph1
>
>
> 10) Cisco Unified Mobile Voice Access Service is running on both Sub and
> Pub on
> CUCM
>
>
>
> Questions :
> ====================
>
>
> 1) Do I need to change my incoming calling number (coming from pstn) from
> 5252222 to 95252222 because the busy trigger on 3001 (phone)
> is set to 1 and therefore any other calling coming to this number will
> head to Voicemail?
>
>
> 2) Anything else you find incorrect with my configuration?
>
>
> -MJ
>
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_______________________________________________
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