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Adding New Gateway existing Call center with Customer voice portal 7.0

harhasnapm
Level 1
Level 1

The redundant setup of CUCM 7.0,  CVP 7.0  and CUP with 2 X PSTN cisco voice gateway – 800 Toll Free service is running right now. 

We need to add one more cisco gateway with SIP trunk   . Telco will shift the 800 service to this new SIP telephone ranges .  Call center service must be available  trough new SIP gateway.  Lines are configured and call is working fine with Normal Extensions. But it is not working with call center agents

18 Replies 18

Chris Deren
Hall of Fame
Hall of Fame

Are you adding or replacing a Gateway?

Is this ingress GW only or VXML GW as well?

Chris

HI Chris,

The existing PSTN gateway are VXML gateway and it is working fine. We need to add the new gateway with SIP trunk. What to do in CVP Side or CUP side to work with new gateway

Current CALL FLOW ---- 800 toll free service---->PSTN--->VG---->CUP---->CVP

Do you want the new CUBE gateway to also be VXML GW? If not where would the vxml call need to be send to?

How is your current CVP routing setup, are you using sigDigits, send to originator, etc? You need to understand your existing environment to this level in order to configure it accordingly.

HTH,

Chris

New Cube gateway to be in vxml. Sending pilot number 5050 to cups via sip Udp with Rtp-nte

VGA------->CUPS------->CVp


Sent from Cisco Technical Support iPad App

OK, did you configure the GW with VXML applicaitons, appropriate dial-peers, etc?

Are you using sigDigits for routing or send to originator?

Is 5050 defined in CUPS, if so how?

Chris

Sounds to me like they are just adding a new ingress gateway... if he could keep his old VXML the same as the old this would probably be alot easier for him to take this in chunks.  Then he would only need to worry about getting the call from the ingress gateway into cups with the same digits as before

Could you post you existing gateway configuration so we can be sure of what your doing?  Also maybe a screenshots of some of the relevent CUPS routes and the CVP OAMP SIP Call Server ICM Tab.

Cheers,

Chad

Now the issue is Agent phone disconnect after one ring trough sip trunk call . note that cvp to cups sip routes are pointed by srv lookup. Need to add any sip commands in gateway which is connected to sip trunk

Now the caller is able to speak with agent  via SIP trunk . but the conversation is disconnected after 18 minutes at all time . However the call is working fine trough Analog trunk . So hope some addtional sip commnads to be done in VG

XML application can be copied from pstn VG to new cube gateway.
5050 is defined in cups as route point to CVP
I can send you existing VG configuration to sigDigits or send to originator


Building configuration...


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!
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supervisory custom-cptone DISCONNECT_SIGNAL
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timing guard-out 300
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connection plar 5050
caller-id alerting dsp-pre-allocate
!
voice-port 0/3/2
supervisory disconnect dualtone mid-call
supervisory custom-cptone DISCONNECT_SIGNAL
input gain 5
output attenuation -3
cptone BR
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 1
timeouts wait-release 1
timing guard-out 300
timing sup-disconnect 50
connection plar 5050
caller-id alerting dsp-pre-allocate
!
voice-port 0/3/3
supervisory disconnect dualtone mid-call
supervisory custom-cptone DISCONNECT_SIGNAL
input gain 5
output attenuation -3
cptone BR
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 1
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timing guard-out 300
timing sup-disconnect 50
connection plar 5050
caller-id alerting dsp-pre-allocate
!
!
!
!
!
dial-peer voice 100 pots
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!
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!
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!
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!
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!
dial-peer voice 8111 voip
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service bootstrap
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codec g711ulaw
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!
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progress_ind connect enable 8
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dial-peer voice 5050 voip
description Call Server Dialpeer with CUPS test
destination-pattern 5050
voice-class codec 1
session protocol sipv2
session target sip-server
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no vad
!
!
num-exp 7900 7202
gateway
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!
sip-ua
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!
banner login ^CCC

|==========================================================================|
| WARNING |
| ======= |
| THE PROGRAMS AND DATA HELD ON THIS SYSTEM ARE THE PROPERTY OF, OR |
| LICENCED BY, THE SHL GROUP. |
| |
| IF THE COMPANY HAS NOT AUTHORISED YOUR ACCESS TO THIS SYSTEM YOU |
| WILL COMMIT A CRIMINAL OFFENCE IF YOU DO NOT IMMEDIATELY DISCONNECT. |
| |
| UNAUTHORISED ACCESS IS STRICTLY FORBIDDEN AND IS A DISCIPLINARY OFFENCE. |
|==========================================================================|

^C
!
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end


Sent from Cisco Technical Support iPad App

OK, does not look like you are using the sigDigits.

Do you see the call making it to the ICM script?  Can you do a monitor to see if it fails at the sendToVRU step?

Did you load all the vxml and tcl scripts onto the flash of the router?

Can you show screen shots of your CVP call server configuration (from OAMP)?

Chris

OK cool, well assuming that the CVP OAMP just sends all calls back into CUPS, you should really only need to adjust you current route in CUPS to the 8111T label to the new PSTN SIP gateway.  This could also be done with a static route in the OAMP on 8111>.  Beyond that in CUPS you need to make sure the new gateway IP is in the incoming and outgoing ACL....  Looks to me like you pretty close.


I agree no sig digits here... how many sites do you have?  (Maybe they only have one site Chris?)  It could also be controlled on the UCCE ICM with different customers and diff labels to each site.

Chad

Only One site is having

Please find the CUPS static routes screenshot , Unable to find CVP server - see the screenshot

CVP.png           

t

CUPS.png

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