01-21-2012 08:52 PM - edited 03-14-2019 09:13 AM
Hello I have a doubt how can I configure the vxml gateway for side remote o distribute vxml gateway to do not using wan bandwidth where I already have cvp in central site and I have another vxml gateway in central site but how can I do detect any parameters to send the audio a vxml GW in site remote for call do not across the wan when the call waiting in Queue.
Thanks.
01-22-2012 07:00 AM
Well, there is no easy answer here without explaining in entirely how CVP works and understanding different CVP deployment models. There are several ways you can accomplish this:
I suggest you take a look at latest CVP SRND and Admin Guides for more info, and if you feel this may be more than you can handle reach out to certified CVP ATP partner for assistance.
HTH,
Chris
01-22-2012 11:45 AM
Thanks for your time Chris.
I have a CVP compresive work SIP,
I think a new label in Network VRU within ICM, and this label in static route in CVP with same label to ip vxml gateway of side remote, also make sure my incoming GW's are VXML gatways and have the dial-peers for VXML to run my voice applications (.tcl scripts)and have all necessary files in flash.
But my doubt how my cvp return calls to correct vxml gateway in branch-office? how to directing the IVR leg back to the same originating gateway
Thanks.
01-22-2012 12:20 PM
No, ICM label is the same for all GWs, you should only have ICM VRU label for VRU originated calls, and separate one for CUCM calls. In order to control which VXML GW is selected for call treatment based on where the call originated/which devices you need to use one of the methods:
My preference is use of sigDigits but, as that provides full control over where call is routed for both IP and PSTN intiated calls. You essentially prefix digits to all calls to the VRU label which are then stripped by CVP before delivering the call to ICM (requires SigDigits configuration on call servers), and then on egress side from CVP the digits are prefixed again and Proxy server (if used) drives where the call is routed. There is a good writeup on it in SRND and CVP Admin Guide.
Chris
01-24-2012 02:36 PM
Hello Chris This is my configuration with central site this gateway is ingress gateway and vxml gateway coresident but is also vxml gateway of others sites, I would like to add new ingress gateway and vxml gateway coresident in site distributed, how can configure that?If call site arrive over site distribute the transfer to vxml local or return at same place, if call arrive another side transfer to vxml gateway central.
UCCE
CVP_VRUType 10 Label 81111111111
Call Server>SIP
DN on the Gateway to play the ringtone :91919191
DN on the Gateway to play the error tone: 92929292
Local Static Routes
92929292, vxmlgw
91919191, vxmlgw
81111>vxmlgw(10.6.20.211 and 10.6.20.212)
39> vxmlgw(To campaing)
30>CUCM (Agent Ext)
application
service new-call flash:bootstrap.vxml
paramspace english language en
paramspace english index 0
paramspace english location flash
paramspace english prefix en
!
!
service cvp-survivability flash:survivability.tcl
paramspace english language en
paramspace english index 0
paramspace english location flash
paramspace english prefix en
!
service ringtone flash:ringtone.tcl
paramspace english language en
paramspace english index 0
paramspace english location flash
paramspace english prefix en
!
service cvperror flash:cvperror.tcl
paramspace english language en
paramspace english index 0
paramspace english location flash
paramspace english prefix en
!
service load bootstrap
!
service handoff flash:handoff.tcl
paramspace english index 0
paramspace english language en
paramspace english location flash
paramspace english prefix enexit
!
service bootstrap flash:bootstrap.tcl
paramspace english index 0
paramspace english language en
paramspace english location flash
paramspace english prefix en
dial-peer voice 987654 voip
description Blocks ulaw
translation-profile incoming block
incoming called-number 987654
no vad
!
dial-peer voice 9191 voip
description Ringtone Dial-Peer for CVP
service ringtone
voice-class codec 10
incoming called-number 91919191
dtmf-relay rtp-nte h245-signal h245-alphanumeric
no vad
!
dial-peer voice 9292 voip
description Error Dial-peer for CVP
service cvperror
voice-class codec 10
voice-class sip rel1xx disable
incoming called-number 92929292
dtmf-relay rtp-nte h245-signal h245-alphanumeric
no vad
!
dial-peer voice 81 voip
description For Incoming Leg (Type 10 Label and Correlation ID)
service bootstrap
voice-class codec 10
incoming called-number 81T
dtmf-relay rtp-nte h245-signal h245-alphanumeric
no vad
Thanks
01-26-2012 01:55 PM
As Chris already pointed out a few times in his posts, there is no right answer here. There are several options and to make an informed decision and the right design choice for your network you need to better understand the CVP Callflow options. You should review the CVP SRND in greater detail to understand the call distribution methods.
Using Signaling Digits allows for very precise call routing decisions and is the method of choice, but you'll need to re-design / reconfigure your whole dialplan. Using Send To Originator is very easy to configure but is only relevant if you use combo gateways (ingress & vxml gateway functionality on the same box), you seem to have a separate VXML gateway so it does not apply to you.
If you have combo gateways, you could implenent Send To Originator by adding these patterns to the Send To Originator list on the Call Server SIP tab.
92929292
91919191
81111>
But again, I do not recommend you go and make changes to these settings if you do not fully understand what you are about to change.
Cheers,
Kris
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