Calls are disconnected after a while; after been transferred from operator in B-ACD

[jesusjimenez1]

Hi everyone!

I have  a problem  with Cisco CME 12.0 and B-ACD feature. The AA is working fine, but when a customer pass to talk with an operator and ask to be transferred with another extension the call is dropped after a silence. Do you have any advice?

My config to AA:

application

service aa flash:app-b-acd-aa-3.0.0.5.tcl

  paramspace english index 1

  param number-of-hunt-grps 1

  param handoff-string aa

  param dial-by-extension-option 1

  param operator 1000

  paramspace english language en

  param max-time-vm-retry 2

  param aa-pilot 6000

  paramspace english location flash:

  param second-greeting-time 30

  param welcome-prompt _bacd_welcome.au

  param queue-manager-debugs 1

  param call-retry-timer 20

  param max-time-call-retry 700

  param voice-mail 6050

  param service-name queue

!

service queue flash:app-b-acd-3.0.0.5.tcl

  param queue-len 5

  param aa-hunt10 1000

  param queue-manager-debugs 1

  param number-of-hunt-grps 1

[jesusjimenez1 · Accepted solution]

Well, if someone is getting this kind of problem, I solved recently.

Problem was an interoperability between H323 (B-ACD is based in H323) and SIP protocol (all of my phones are based in SIP).

Luckly, Release 11.6 onwards, BACD supports line side call from SIP, SCCP, FXS phones but you need to configure a loopback voip dialpeer (SIP) with codec g711ulaw, DTMF rtp-nte, and a proxy B-ACD pilot number as the destination pattern. Also you need to change the proxy destination pattern to actual B-ACD pilot number before sending out the call invite. You can achieve this using either voice class sip-profiles or translation rules.

Note: I used translation-rule to change the proxy, thats  'cause when the incoming call is showed in the display of end user it's show the 1234 number in the example, in the translation-rule you can transform this in the original incoming number.

Example using sip-profiles:

voice class sip-profiles 1

request INVITE sip-header SIP-Req-URI modify "6001@" "6000@"

request INVITE sip-header To modify "6001@" "6000@"

request INVITE sip-header From modify "<sip:(.*)@" "<sip:1234@"

request INVITE sip-header Remote-Party-ID remove

#dial-peer voice 1 voip

destination-pattern 6001

session protocol sipv2

session target ipv4:1.0.0.1

voice-class sip profiles 1

dtmf-relay rtp-nte

codec g711ulaw

dial-peer voice 2 voip

service aa-bcd

session protocol sipv2

incoming called-number 6000

dtmf-relay rtp-nte

codec g711ulaw

Example using Translation-rule:

translation-rule 1

Rule 1 6001 6000

Rule 2 7001 7000

translation-rule 2

Rule 1 * 6666

..

dial-peer voice 1 voip

destination-pattern 6001

translate-outgoing calling 2

translate-outgoing called 1

session protocol sipv2

session target ipv4:1.0.0.1

dtmf-relay rtp-nte

codec g711ulaw

dial-peer voice 2 voip

service aa-bcd1

session protocol sipv2

incoming called-number 6000

dtmf-relay rtp-nte

codec g711ulaw