06-03-2013 12:17 AM - edited 03-16-2019 05:39 PM
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
Solved! Go to Solution.
06-03-2013 02:35 AM
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.
06-03-2013 06:35 AM
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.
06-03-2013 01:28 AM
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
06-03-2013 01:37 AM
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.
06-03-2013 01:55 AM
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???
06-03-2013 01:57 AM
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.
06-03-2013 02:23 AM
Dear's
DTMF relay. where to select the DTMP relay in which TAB.
E1 PRI,
at present 1 CUCM but in future redundant.
06-03-2013 02:35 AM
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.
06-03-2013 02:56 AM
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
06-03-2013 02:59 AM
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
06-03-2013 03:08 AM
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
06-03-2013 04:47 AM
Can you please tell me more. What did you change. Give me a screenshot.
G.
Sent from Cisco Technical Support Android App
06-03-2013 05:11 AM
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
06-03-2013 06:35 AM
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.
06-03-2013 08:14 AM
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
06-03-2013 08:21 AM
Thanks for the rating and good luck with your project.
G.
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