[cisco-voip] CUE 8.0 with CUCM Issue

[Jorge L. Rodriguez Aguila]

voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 supplementary-service h450.12
 no supplementary-service h450.2
 no supplementary-service h450.3
 fax protocol pass-through g711ulaw
 h323
  call start interwork
 modem passthrough nse codec g711ulaw
 sip
  bind control source-interface Vlan389
  bind media source-interface Vlan389
  midcall-signaling passthru
!
!
!
voice class codec 100
 codec preference 1 g711ulaw
 codec preference 2 g729r8
 codec preference 3 g729br8
!

sccp ip precedence 3
sccp
!
sccp ccm group 10
 bind interface Loopback0
 associate ccm 1 priority 1
 associate ccm 2 priority 2
 associate profile 1 register TAJM-HW-XCODE
!
dspfarm profile 1 transcode
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 maximum sessions 24
 associate application SCCP
!
!
dial-peer voice 200 voip
 description To_CM
 preference 2
 destination-pattern 2...
 voice-class codec 100
 voice-class h323 500
 session target ipv4:192.168.200.2
 dtmf-relay h245-alphanumeric
 ip qos dscp cs3 signaling
 no vad
!
dial-peer voice 210 voip
 description To_CM
 destination-pattern 2...
 voice-class codec 100
 voice-class h323 500
 session target ipv4:192.168.200.3
 dtmf-relay h245-alphanumeric
 ip qos dscp cs3 signaling
 no vad
!
dial-peer voice 100 voip
 translation-profile incoming pstn-in
 session protocol sipv2
 session target sip-server
 incoming called-number 0901876618109591AD
 dtmf-relay rtp-nte
!
dial-peer voice 110 pots
port 0/3/0:0
incoming called-number .

Other Dial-peers omitted, as they are outbound

From: Buchanan, James [mailto:jbuchanan at presidio.com]
Sent: Sunday, April 17, 2011 9:21 PM
To: Jorge L. Rodriguez Aguila; 'cisco-voip at puck.nether.net'
Subject: RE: CUE 8.0 with CUCM Issue

Can you send the relevant parts of your gateway config?

James Buchanan | Technology Manager, UC | South Region | Presidio Networked Solutions
12 Cadillac Dr, Suite 130, Brentwood, TN 37027 | jbuchanan at presidio.com<mailto:jbuchanan at ctiusa.com>
D: 615-866-5729 | F: 615-866-5781 | www.presidio.com<http://www.presidio.com/>
CCIE #25863, Voice

From: Jorge L. Rodriguez Aguila [mailto:jorge.rodriguez at netxar.com]
Sent: Sunday, April 17, 2011 8:11 PM
To: Buchanan, James; 'cisco-voip at puck.nether.net'
Subject: RE: CUE 8.0 with CUCM Issue

GW has a CSS that points to a DID PT, inbound calls hit a translation that has a CSS that has the Phones and the VM ports as well.

Jorge

From: Buchanan, James [mailto:jbuchanan at presidio.com]
Sent: Sunday, April 17, 2011 9:05 PM
To: Jorge L. Rodriguez Aguila; 'cisco-voip at puck.nether.net'
Subject: Re: CUE 8.0 with CUCM Issue

Does your inbound CSS have acces to the partition the CTI ports are in?

From: Jorge L. Rodriguez Aguila [mailto:jorge.rodriguez at netxar.com]
Sent: Sunday, April 17, 2011 08:58 PM
To: Buchanan, James; Cisco-Voip (cisco-voip at puck.nether.net) <cisco-voip at puck.nether.net>
Subject: RE: CUE 8.0 with CUCM Issue

CTI Ports with a JTAPI user

From: Buchanan, James [mailto:jbuchanan at presidio.com]
Sent: Sunday, April 17, 2011 8:42 PM
To: Jorge L. Rodriguez Aguila; Cisco-Voip (cisco-voip at puck.nether.net)
Subject: RE: CUE 8.0 with CUCM Issue

How is your CUE integrated to CUCM?

James Buchanan | Technology Manager, UC | South Region | Presidio Networked Solutions
12 Cadillac Dr, Suite 130, Brentwood, TN 37027 | jbuchanan at presidio.com<mailto:jbuchanan at ctiusa.com>
D: 615-866-5729 | F: 615-866-5781 | www.presidio.com<http://www.presidio.com/>
CCIE #25863, Voice

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jorge L. Rodriguez Aguila
Sent: Sunday, April 17, 2011 7:39 PM
To: Cisco-Voip (cisco-voip at puck.nether.net)
Subject: [cisco-voip] CUE 8.0 with CUCM Issue

I have integrated a CUE 8.0 with CUCM 8.0.3 and internally it works perfectly but when calls coming in from a T1 or a SIP trunk are transferred to CUE I get Busy tone on T1 and dropped call on the SIP trunk(IP2IP GW running SIP towards Provider and H323 towards Call Manager). I have verified that even though I am running G711u I have HW transcoding resources available. Any ideas on what Might be? This is my first CUE to Call Manager (non express) I don’t seem to find what is wrong and some pointers into where to look are greatly appreciated.


Jorge Rodriguez
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20110417/bb72a55c/attachment.html>