cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
708
Views
0
Helpful
14
Replies

UCCX 9.0

jack samuel
Level 1
Level 1

Hi

I have configured and integrated UCCX 9.0 with CUCM 9.0, i have tested by dialling a trigger number and it gives me the Automated Attendent audio.

Now i shld ask ITSP to hit toll free number to my trigger number 1000,when the call will be thrown to my router, my router have a dial-peer for 1000.

I shld create a dial-peer for the trigger number to hit on CUCM and from there it will automatically route to UCCX

Is it work in same way what i am thinking or something else????

dial-peer voice 1 voip

description  to CUCM **

destination-pattern 1000----------------Trigger number

session-target ipv4:1.1.1.1------------------------- the CUCM IP

dial-peer voice 2 voip

description  to CUCM **

destination-pattern 2...----------------corporate DID

session-target ipv4:1.1.1.1------------------------- the CUCM IP

Thanks

2 Accepted Solutions

Accepted Solutions

OK,

so you should have an inbound dial-peer like this:

dial-peer voice 10 pots

description ** incoming POTS **

incoming called-number .

direct-inward-dial

And an outbound like this

dial-peer voice 20 voip

description ** outgoing H323 **

destination-pattern 1234...

session-target ipv4:10.20.30.40

dtmf-relay h245-signal

no vad

G.

View solution in original post

Check the CSS's on CTI Route points and CTI ports and also the partition of agent extensions. It looks like the call cannot be routed to the agent's phone.

G.

View solution in original post

14 Replies 14

barry
Level 7
Level 7

Hi Jack

Yes, that's basically it. Route the call into CUCM, and it will then pass the call to UCCX via the CTI Route Point / CTI Ports.

HTH.

Barry Hesk

Intrinsic Network Solutions

Not quite. You are missing the incoming dial-peer. Also, I don't see the reason why two separate dial-peers should be implemented for a single routing path.

Is it a VoIP (H.323 or SIP) connection you have towards the ITSP?

How many call processing CUCM's are there?

G.

Oh yeah,

Dears,

Yes correct no need of 2 dial-peer, on the existing it will work becz the trigger is from the valid DID.

But now i am facing 1 problem, whenever i dial a trigger from PSTN it routes to the UCCX it gives me the auto attendant script which i selected in application, but when the auto attendant ask me  to press 1  or press 2 or 0 for operator, when i press 1 or 2 or 0 there is no effect.she keeps on prompting are you still there ???

where things are missing???

Yes, DTMF relay.

Can you please answer the questions above first.

1. Type of connection to ITSP, so I can give you an example of an incoming dial-peer.

2. How many call processing CUCM's so I can give you an example of an outgoing dial-peer, or multiple redundant dial-peers, if necessary.

G.

Dear's

DTMF relay.  where to select the DTMP relay in which TAB.

E1 PRI,

at present 1 CUCM but in future redundant.

OK,

so you should have an inbound dial-peer like this:

dial-peer voice 10 pots

description ** incoming POTS **

incoming called-number .

direct-inward-dial

And an outbound like this

dial-peer voice 20 voip

description ** outgoing H323 **

destination-pattern 1234...

session-target ipv4:10.20.30.40

dtmf-relay h245-signal

no vad

G.

Dear's

yes it is working

1 advise i need from you.

Should i distribute the CTI ports on both the call manager for ex: if i have 30 CTI then 15 on CUCM-PUB and 15 CUCM-SUB

Should i need to create a  6 trigger for  the same application, i just dialled from PSTN by 2 mobile phone to the trigger number and the auto attendant is responding to both the phone so i think no need to create 6 trigger only 1 is enough

Gergely Szabo
VIP Alumni
VIP Alumni

Distributing CTI ports: I dont think it's necessary. Would make things more complicated.
One trigger is enough.
G.


Sent from Cisco Technical Support Android App

Dear,

I changed the sricpt to one of the existing script running in another UCCX, the same sript and audio file i have selected in the application management of the new UCCX, but the call are not routing to the agents it gives prompting all our staff our busy at the moment but actually the test agents are free.

I have selected 2 users in the resources in the CSQ.

Thanks

Gergely Szabo
VIP Alumni
VIP Alumni

Can you please tell me more. What did you change. Give me a screenshot.
G.


Sent from Cisco Technical Support Android App

i changed a script nothing else i am using the exsting  script of the running contaact center.

PSTN call hits the agent desktop when the agent is ready the agent status changes to  not ready but desk phone doesnt ring.

deskphone is already in rmcm controlled devices and the user profile is also controlled by rmcm. The user who is logged in is already in the resources group.

please help

Check the CSS's on CTI Route points and CTI ports and also the partition of agent extensions. It looks like the call cannot be routed to the agent's phone.

G.

Thanks Dear,

At last i am ending my stuff very nicely,

I have rated your must of the post which on reading will help others.

Help others GOD will help you.

Thanks

Thanks for the rating and good luck with your project.

G.