Transfer a call from Agent to particular IVR menu

Hi,
In IPCC Enterprise (ver 7.1.4) with CVP (3.1) solution, how I can configure that an Agent can transfer a call to a particular IVR menu.
Can anybody help me on this regards.
Muzammel Haque

Hello Abdul,
I am tryingto understand the user persepective on the agent what sort of number or perticular routing point should be able to transfer ,will there be set if listed option on the agent to transfer back to IVR where customer was last time ?
trying to understand interms of the what sort of list visibilty on the agnet will have to transfer back to same place where customer was there on the IVR ??
please CC reply email to [email protected]
Thanks

Similar Messages

How Can I make redirect caller from agent to any branch IVR?

I have IPCC Enterprise Edition (ICM 5.0, IVR 3.1)
How can I make redirect caller from agent to chosen branch IVR, and send with call any variable to IVR (e.g. account number).
Regards
Krzysztof

I would suggest that post-routing should not be used. The way to go is to use translation routing applications. Although post-routing is easy to set up, almost all deployments require translation routing. I can't think of anything good to say about post routing, other than it's trivial to configure.
You want the main route point to be on the CM_PIM.RC for a number of reasons. You can check to see which of your IVRs are on line, and/or to do load balancing. If calls don't need to go to the IVR if agents are ready, you can have an LAA Select node immediately, and then translation route. Even if you just have one IVR, translation routing will enable you to do RONA more easily.
You need to bite the bullet and learn translation routing. If you do it that way, peripheral variables attached to the call remain unaffected whether the call is under the control of the CM_PIM.RC or the IVR_PIM.RC. This is exactly what you want.
There is no need, in my opinion, to go to an external database. In your IVR, decisions made by the caller (e.g. which "skill" they want) set peripheral variables. When the call comes out of this script you check the peripheral variable, and depending on the value do a queue to the appropriate skill group and run external script (BasicQ.aef).
Separate the intelligence gathering IVR script from the Queuing script. Allow ICM to make the routing decisions.

Disable "Please wait While I Transfer your call" from cuc 9.0

Dear All,
I've installed the Cisco UC Solution with CUCM 9.0 and CUC9.0. I'm using CUC for auto attendant and i want someone who can help me how to remove the sound "Please wait While I Transfer your call" from CUC which comes at the time of call routing.
best regards
Kennedy

Hi Kennedy
Please go to system call handler - edit your Call handler which responsible for AA - transfer rules- On this page you will find transfer actions uncheck play "wait while i transfer your call prompt".
Thank you
Please rate all useful information

Transfer received calls by agents directly to another CSQ

Hi! I need advice. We have UCCX 7 Premium. I was asked to create a following script logic:
After an IVR greeting a received call comes in the queue "CSQ_Operators". An operator answers and depending on callers needs transfers the call to another specified queue (for instance to "CSQ_Support") if there is no answer from "CSQ_Support" the call must go back to the operator which transfered it. After the conversation a caller has to be offered to evaluate quality of service by dial from 1 to 5.
I have two questions.
1. Can the agents of "CSQ_Operators" transfer a call directly to "CSQ_Support" if both queues located withing the same application and the same script? Or I can only add the second application and script with the queue "CSQ_Support" and the operators will transfer calls to the route point number of second script?
2. Which step in the script would better use for saving and calculation the callers input (1 - 5) in the external file?
Thanks in advance!
Ruslan

You can write all that data out to text file yes. The problem is that writing out to files is done using the Doc Template step and the template has to be of a known configuration. So you could write one line of data or XML out to a file and then the next call would have to be a file with a different name. That is fine if you have a parser application somewhere picking up each file as it comes out.
If you are looking for a batch job over night thing, then I would suggest a custom historical report run on the Schedule that outputs to a data or csv type file.
Other than that, there is not much you can do without getting into crazy custom Java in the code.

How can i transfer a call from SIP 9971 to PBX system on CME router

hello everybody,
I have a critical problem about interaction of transfering feature between CME router and pbx panasonic system in some status. let me explain more detail about this issue..i have a SIP 9971(CP-9971) registered on CME at the one site and a voice gateway that is connect with PBX system through a E1 pri trunk connection at the other site. totally the integration between CME and PBX is ok and there is no problem in two direction, i mean i can call pbx system from cp-9971 and vise versa but when i call from a phone which is registered on PBX site to SIP 9971 which is registered on cisco CME call is connected,then when i try to transfer that call to another phone at PBX site, the session is open between two panasonic phones but no audio transmited in two direction. in addition every thing works fine about SCCP phones(transfer feature works fine). here is my configuration file. i hope someone could help me because i've searched a lot but no result help help help plz....
cme router 3845 configuration
VOIP-3845#show running-config
Building configuration...
Current configuration : 12657 bytes
! Last configuration change at 11:44:01 UTC Mon Oct 31 2011 by admin
! NVRAM config last updated at 11:44:02 UTC Mon Oct 31 2011 by admin
! NVRAM config last updated at 11:44:02 UTC Mon Oct 31 2011 by admin
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname VOIP-3845
boot-start-marker
boot-end-marker
no aaa new-model
clock calendar-valid
dot11 syslog
ip source-route
ip cef
no ipv6 cef
multilink bundle-name authenticated
voice-card 0
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
sip
bind control source-interface Loopback10
bind media source-interface Loopback10
registrar server
voice register global
mode cme
source-address 192.168.2.1 port 5060
max-dn 720
max-pool 262
load 9971 sip9971.9-1-1SR1.loads
authenticate register
authenticate realm cisco.com
tftp-path flash:
file text
create profile sync 0063544528862458
camera
video
voice register dn 1
number 500
voice register dn 2
number 600
voice register dn 3
number 700
name test
voice register template 1
softkeys idle Newcall Redial Cfwdall
softkeys connected Confrn Endcall Hold Trnsfer
voice register pool 1
id mac B8BE.BF23.5242
type 9971
number 1 dn 1
template 1
username test password test
camera
video
blf-speed-dial 4 600 label "test"
voice register pool 2
id mac B8BE.BF9C.5476
type 9971
number 1 dn 2
template 1
username bank password bank
camera
video
voice register pool 3
id mac B8BE.BF9C.51D4
type 9971
number 1 dn 3
template 1
username test1 password test1
camera
video
voice register pool 4
id mac B8BE.BF9C.4FA2
number 1 dn 1
camera
video
crypto pki token default removal timeout 0
crypto pki trustpoint TP-self-signed-1576175886
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-1576175886
revocation-check none
rsakeypair TP-self-signed-1576175886
crypto pki certificate chain TP-self-signed-1576175886
certificate self-signed 01
30820241 308201AA A0030201 02020101 300D0609 2A864886 F70D0101 04050030
31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
69666963 6174652D 31353736 31373538 3836301E 170D3131 31303038 30393034
34365A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D31 35373631
37353838 3630819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
8100D6EC 47BCDC3C 82F43FF3 23522678 2616868D 9910DCD2 E36016B3 D7B40DA7
53A6E339 4978D451 21F051BE B21F8AD5 86B952DC 1ECCE371 3E094B54 26A41E14
A3055C06 AE860756 425E5C50 E62B3287 631B1E87 9BAC2E39 2810E120 DA3BF823
947EA591 81CA5489 1B868239 E835EC7C 0AA7651A 22D6E47F 545EBEF3 A172C9A3
5A0D0203 010001A3 69306730 0F060355 1D130101 FF040530 030101FF 30140603
551D1104 0D300B82 09564F49 502D3338 3435301F 0603551D 23041830 1680146C
934AD072 99DDC600 ECD6F389 8F71E0C2 18EC2E30 1D060355 1D0E0416 04146C93
4AD07299 DDC600EC D6F3898F 71E0C218 EC2E300D 06092A86 4886F70D 01010405
00038181 000E82F6 5FBB847C 49226955 6F7DECE7 0B093513 D57C35D5 4CD22FA7
8144A080 B0D56C8D 86AF8156 0152443A A3FBE59F B1AEFFBC BEB43E09 35757BAD
4C06FC4A 0F3695E0 B00FBD30 4E8F36CE 7748F39C F9602650 7A1D2D48 DBC31237
AE3D63CE 593D31F5 62E4916F D20E30E8 30DC55C0 120FBD26 D2768DBC A67DDC34
5BDB66B1 E3
quit
license udi pid CISCO3845-MB sn FOC14421Q1Y
archive
log config
hidekeys
username admin privilege 15 secret 5 $1$Zf7j$P93opukmmEBIioVpjmHB3.
redundancy
interface Loopback10
ip address 192.168.2.1 255.255.255.0
interface Tunnel1
ip address 172.25.10.1 255.255.255.0
no ip redirects
ip nhrp map multicast dynamic
ip nhrp network-id 10
tunnel source GigabitEthernet0/1.1
tunnel mode gre multipoint
tunnel key 100
interface Tunnel2
ip address 172.25.11.1 255.255.255.0
no ip redirects
ip nhrp map multicast dynamic
ip nhrp network-id 20
tunnel source GigabitEthernet0/1.2
tunnel mode gre multipoint
interface Tunnel14
ip address 192.168.13.129 255.255.255.252
tunnel source GigabitEthernet0/1.1
tunnel destination 10.2.68.25
interface Tunnel18
ip address 192.168.13.137 255.255.255.252
tunnel source GigabitEthernet0/1.1
tunnel destination 10.9.160.236
interface GigabitEthernet0/0
no ip address
shutdown
duplex auto
speed auto
media-type rj45
interface GigabitEthernet0/1
no ip address
duplex auto
speed auto
media-type rj45
interface GigabitEthernet0/1.1
encapsulation dot1Q 10
ip address 10.9.160.25 255.255.255.0
interface GigabitEthernet0/1.2
encapsulation dot1Q 50
ip address 10.10.9.25 255.255.255.0
router eigrp 202
network 172.25.11.0 0.0.0.255
network 192.168.2.0 0.0.0.15
redistribute static route-map MYMAP1
router eigrp 201
network 172.25.10.0 0.0.0.255
network 192.168.2.0 0.0.0.15
redistribute static route-map MYMAP1
ip forward-protocol nd
ip http server
ip http secure-server
ip http path flash:/gui
ip route 10.2.68.0 255.255.255.0 10.9.160.1
ip route 10.10.0.0 255.255.0.0 10.10.9.1
ip route 10.64.164.30 255.255.255.255 10.9.160.1
ip route 192.168.14.0 255.255.255.0 192.168.13.130
ip route 192.168.17.0 255.255.255.0 Tunnel18
ip access-list standard REDIS1
permit 192.168.14.0
permit 192.168.17.0
route-map MYMAP1 permit 10
match ip address REDIS1
snmp-server community test RO
tftp-server flash:term11.default.loads
tftp-server flash:dkern9971.100609R2-9-0-3.sebn
tftp-server flash:kern9971.9-0-3.sebn
tftp-server flash:rootfs9971.9-0-3.sebn
tftp-server flash:sboot9971.111909R1-9-0-3.sebn
tftp-server flash:sip9971.9-0-3.loads
tftp-server flash:skern9971.022809R2-9-0-3.sebn
tftp-server flash:sccp11.9-0-2sr1s
tftp-server flash:SCCP11.9-1-1SR1S.loads
tftp-server flash:apps11.9-1-1TH1-16.sbn
tftp-server flash:cnu11.9-1-1TH1-16.sbn
tftp-server flash:cvm11sccp.9-1-1TH1-16.sbn
tftp-server flash:dsp11.9-1-1TH1-16.sbn
tftp-server flash:jar11sccp.9-1-1TH1-16.sbn
tftp-server flash:term06.default.loads
tftp-server flash:sip9971.9-1-1SR1.loads
tftp-server system:cme/sipphone
tftp-server flash:Desktops/320x212x12/NantucketFlowers.png
tftp-server flash:Desktops/320x212x12/TN-CampusNight.png
tftp-server flash:Desktops/320x212x12/TN-CiscoFountain.png
tftp-server flash:Desktops/320x212x12/TN-Fountain.png
tftp-server flash:Desktops/320x212x12/TN-MorroRock.png
tftp-server flash:Desktops/320x212x12/TN-NantucketFlowers.png
tftp-server flash:Desktops/320x212x12/Fountain.png
tftp-server flash:Desktops/320x212x12/CiscoLogo.png
tftp-server flash:Desktops/320x212x12/TN-CiscoLogo.png
tftp-server flash:Desktops/320x212x12/List.xml
tftp-server flash:Desktops/320x216x16/List.xml
tftp-server flash:Desktops/320x212x16/List.xml
tftp-server flash:gui/admin_user.html
tftp-server flash:gui/admin_user.js
tftp-server flash:gui/CiscoLogo.gif
tftp-server flash:gui/Delete.gif
tftp-server flash:gui/dom.js
tftp-server flash:gui/downarrow.gif
tftp-server flash:gui/ephone_admin.html
tftp-server flash:gui/logohome.gif
tftp-server flash:gui/normal_user.html
tftp-server flash:gui/normal_user.js
tftp-server flash:gui/Plus.gif
tftp-server flash:gui/sxiconad.gif
tftp-server flash:gui/Tab.gif
tftp-server flash:gui/telephony_service.html
tftp-server flash:gui/uparrow.gif
tftp-server flash:gui/xml-test.html
tftp-server flash:gui/xml.template
tftp-server flash:ringtones/Analog1.raw
tftp-server flash:ringtones/Analog2.raw
tftp-server flash:ringtones/AreYouThere.raw
tftp-server flash:ringtones/AreYouThereF.raw
tftp-server flash:ringtones/Bass.raw
tftp-server flash:ringtones/CallBack.raw
tftp-server flash:ringtones/Chime.raw
tftp-server flash:ringtones/Classic1.raw
tftp-server flash:ringtones/Classic2.raw
tftp-server flash:ringtones/ClockShop.raw
tftp-server flash:ringtones/DistinctiveRingList.xml
tftp-server flash:ringtones/Drums1.raw
tftp-server flash:ringtones/Drums2.raw
tftp-server flash:ringtones/FilmScore.raw
tftp-server flash:ringtones/HarpSynth.raw
tftp-server flash:ringtones/Jamaica.raw
tftp-server flash:ringtones/KotoEffect.raw
tftp-server flash:ringtones/MusicBox.raw
tftp-server flash:ringtones/Piano1.raw
tftp-server flash:ringtones/Piano2.raw
tftp-server flash:ringtones/Pop.raw
tftp-server flash:ringtones/Pulse1.raw
tftp-server flash:ringtones/Ring1.raw
tftp-server flash:ringtones/Ring2.raw
tftp-server flash:ringtones/Ring3.raw
tftp-server flash:ringtones/Ring4.raw
tftp-server flash:ringtones/Ring5.raw
tftp-server flash:ringtones/Ring6.raw
tftp-server flash:ringtones/Ring7.raw
tftp-server flash:ringtones/RingList.xml
tftp-server flash:ringtones/Sax1.raw
tftp-server flash:ringtones/Sax2.raw
tftp-server flash:ringtones/Vibe.raw
tftp-server flash:APPS-1.2.1.SBN
tftp-server flash:SYS-1.2.1.SBN
tftp-server flash:GUI-1.2.1.SBN
tftp-server flash:CP7921G-1.2.1.LOADS
tftp-server flash:TNUX-1.2.1.SBN
tftp-server flash:TNUXR-1.2.1.SBN
tftp-server flash:WLAN-1.2.1.SBN
tftp-server flash:apps37sccp.1-2-1-0.bin
tftp-server flash:APPSH-1.3.1.SBN
tftp-server flash:GUIH-1.3.1.SBN
tftp-server flash:CP7925G-1.3.1.LOADS
tftp-server flash:SYSH-1.3.1.SBN
tftp-server flash:TNUXH-1.3.1.SBN
tftp-server flash:WLANH-1.3.1.SBN
tftp-server flash:SCCP11.9-2-1S.loads
tftp-server flash:Desktops/320x212x12/CampusNight.png
tftp-server flash:Desktops/320x212x12/CiscoFountain.png
tftp-server flash:Desktops/320x212x12/MorroRock.png
tftp-server flash:skern9971.022809R2-9-2-1.sebn
tftp-server flash:sip9971.9-2-1.loads
tftp-server flash:sboot9971.031610R1-9-2-1.sebn
tftp-server flash:rootfs9971.9-2-1.sebn
tftp-server flash:dkern9971.100609R2-9-2-1.sebn
tftp-server flash:kern9971.9-2-1.sebn
tftp-server flash:United_States/g4-tones.xml
tftp-server flash:English_United_States/gd-sip.jar
tftp-server flash:sboot9971.031610R1-9-1-1SR1.sebn alias sboot9971.031610R1-9-1-1SR1.sebn
tftp-server flash:rootfs9971.9-1-1SR1.sebn alias rootfs9971.9-1-1SR1.sebn
tftp-server flash:kern9971.9-1-1SR1.sebn alias kern9971.9-1-1SR1.sebn
tftp-server flash:dkern9971.100609R2-9-1-1SR1.sebn alias dkern9971.100609R2-9-1-1SR1.sebn
tftp-server flash:skern9971.022809R2-9-1-1SR1.sebn alias skern9971.022809R2-9-1-1SR1.sebn
control-plane
mgcp profile default
dial-peer voice 1 voip
description connection-trough-PBX
destination-pattern 0....
session target ipv4:192.168.13.130
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 100 voip
description K
destination-pattern 9T
session target ipv4:192.168.13.130
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 5 voip
shutdown
destination-pattern *3709
session protocol sipv2
session target ipv4:192.168.13.130
session transport tcp
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
dial-peer voice 2 pots
incoming called-number .
dial-peer voice 10 voip
gatekeeper
shutdown
telephony-service
em logout 0:0 0:0 0:0
max-ephones 262
max-dn 400
ip source-address 192.168.2.1 port 2000
load 7911 SCCP11.9-2-1S
max-conferences 12 gain -6
web admin system name admin secret 5 $1$IKnn$tyKyuBcGqXFl6nhxCSu.z0
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp 7960 Oct 29 2011 12:39:25
ephone-template 1
softkeys connected Confrn Endcall Trnsfer Hold
keep-conference endcall
ephone-dn 1 dual-line
number 200
label test
name test
ephone-dn 2 dual-line
number 300
label Sepahbod
name Sepahbod
ephone-dn 4 dual-line
number 666
ephone-dn 5 dual-line
number 660
ephone-dn 6 dual-line
number 670
ephone-dn 7 dual-line
number 770
ephone-dn 8 dual-line
number 770
ephone-dn 9 dual-line
number 999
ephone 1
device-security-mode none
mac-address 18EF.639F.BCB0
keep-conference endcall
button 1:1
ephone 2
device-security-mode none
mac-address 0025.8418.B017
ephone-template 1
keep-conference endcall
button 1:2
ephone 3
device-security-mode none
mac-address F04D.A243.3154
keep-conference endcall
button 1:4
ephone 4
device-security-mode none
mac-address 6CF0.496A.69E9
button 1:4
ephone 5
device-security-mode none
mac-address 0015.E987.345F
keep-conference endcall
button 1:5
ephone 6
device-security-mode none
mac-address 0024.1DEA.614A
keep-conference endcall
button 1:6
ephone 9
device-security-mode none
mac-address 001D.7D4D.4DCB
button 1:9
line con 0
line aux 0
line vty 0 4
login local
transport input telnet
scheduler allocate 20000 1000
end
and Voice Gateway connected two PBX system configuration
Current configuration : 3486 bytes
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Voice-GW
boot-start-marker
boot-end-marker
card type e1 0 2
no aaa new-model
network-clock-participate wic 2
dot11 syslog
ip source-route
ip cef
no ipv6 cef
multilink bundle-name authenticated
isdn switch-type primary-net5
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
h323
voice-card 0
crypto pki token default removal timeout 0
license udi pid CISCO2811 sn FHK1352F0E9
username admin privilege 15 secret 5 $1$O6AN$1kvvqiLdIl3/ZTHoyYRy0/
redundancy
controller E1 0/2/0
framing NO-CRC4
pri-group timeslots 1-31
controller E1 0/2/1
interface Tunnel14
ip address 192.168.13.130 255.255.255.252
tunnel source FastEthernet0/1
tunnel destination 10.9.160.25
interface Tunnel17
ip address 192.168.13.134 255.255.255.252
tunnel source FastEthernet0/1
tunnel destination 10.9.160.25
interface FastEthernet0/0
ip address 192.168.14.252 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
ip address 10.2.68.25 255.255.255.0
duplex auto
speed auto
interface Serial0/2/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn overlap-receiving
isdn incoming-voice voice
no cdp enable
router eigrp 201
network 172.25.10.0 0.0.0.255
network 192.168.14.0
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 10.9.160.0 255.255.255.0 10.2.68.1
ip route 10.128.0.69 255.255.255.255 Tunnel14
ip route 192.168.2.1 255.255.255.255 192.168.13.129
ip route 192.168.17.0 255.255.255.0 Tunnel14
tftp-server flash:SCCP11.9-2-1S.loads
tftp-server flash:jar11sccp.9-2-1TH1-13.sbn
tftp-server flash:dsp11.9-2-1TH1-13.sbn
tftp-server flash:cvm11sccp.9-2-1TH1-13.sbn
tftp-server flash:cnu11.9-2-1TH1-13.sbn
tftp-server flash:apps11.9-2-1TH1-13.sbn
control-plane
voice-port 0/0/0
caller-id enable
voice-port 0/0/1
voice-port 0/0/2
supervisory disconnect dualtone mid-call
dial-type pulse
disc_pi_off
output attenuation 1
echo-cancel coverage 32
timeouts call-disconnect 5
timeouts wait-release 1
timing hookflash-out 50
timing sup-disconnect 50
connection plar 600
caller-id enable
voice-port 0/0/3
caller-id enable
voice-port 0/2/0:15
mgcp profile default
dial-peer voice 1 pots
description connection-to-PBX
destination-pattern 0....
direct-inward-dial
port 0/2/0:15
forward-digits 4
dial-peer voice 10 voip
destination-pattern ...
session target ipv4:192.168.13.129
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 20 pots
description FXO-K
destination-pattern 9T
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
direct-inward-dial
port 0/0/2
prefix 9
dial-peer voice 30 pots
description FXO-K2
destination-pattern 9T
direct-inward-dial
port 0/0/1
prefix 9
telephony-service
max-ephones 20
max-dn 100
ip source-address 192.168.14.252 port 2000
cnf-file location flash:
load 7911 term11.default.loads
max-conferences 8 gain -6
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 1
number 770
line con 0
line aux 0
line 1/0 1/15
line vty 0 4
login local
transport input telnet
scheduler allocate 20000 1000
end

Having looked at your spreadsheet I see you're failing H323 transfers back to your ISDN system, but only under certain circumstances. Quite why, I'm not sure, possibly because you haven't codec defined on your H323 dial peers. or it could be something else
I think you may be able to work around the problem by adding
" supplementary-service h450.12 " under voice service voip on your CME router as a quick fix.
reference
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmetrans.html#wpxref44614
worth a try
Adam

UCS320, is it possible to transfer a caller from VM indication to receitionist?

Hi, my customer asked if the following function can be realized.
was looking for a way to see if they can do a “You’ve reached Flower's VM, if this matter is urgent, please press 0 to speak to the receptionist”. Right now her message is “You’ve reached Flower’s VM, if this matter is urgent, please call back and speak to the receptionist”.
When she told about this, I tried # * and 0 during the greeting and it does nothing. I told her we would look into this, as I didn’t see anything in the documentation.
Best Regard

Hi Rex,
The UC320W doesn't support the dial 0 to reach attendant feature while connected to voicemail. One option they could do though is to change the Call Forward No Answer target for the Flower's user to be the receptionist, instead of their voicemail box. The receptionist could either a) reroute the call to another user extension or b) transfer the call to Flower's VM to leave a message (default is 7 + extension).
Hope this helps.
Chris

TS5458 Continuity: Transfer a call from Mac back to iPhone?

Hardware/Software:
iPhone 6 running iOS8.1.2.
2013 MBP Retina running OS X 10.10.1
Scenario:
Continuity works and correctly forwards an incoming call on the iPhone to the Mac.
Call is answered on the Mac and conversation starts.
Discussion requires me to withdraw with the iPhone to somewhere private.
How do I transfer the call back to the iPhone?
I've tried taking my headphones out of the Mac and plugging into the iPhone and pressing the green bar at the top of the iPhone screen that says "Touch to return to call". This hasn't worked. Nothing else comes to mind.
Any ideas?
Thanks

Having looked at your spreadsheet I see you're failing H323 transfers back to your ISDN system, but only under certain circumstances. Quite why, I'm not sure, possibly because you haven't codec defined on your H323 dial peers. or it could be something else
I think you may be able to work around the problem by adding
" supplementary-service h450.12 " under voice service voip on your CME router as a quick fix.
reference
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmetrans.html#wpxref44614
worth a try
Adam

UCCE send a caller from agent to IVR and back to the same agent

Hello,
I am trying to come up with a way to implement the following logic in UCCE 7.5 with IP IVR 7.0
A caller gets to an agent and requests a service that requires identification
The agent sends the caller back to IVR for identification
UCCE reserves the agent. If caller hangs up the agent is chaged back to ready state.
After authentication the caller is taken back to the same agent.
I am not sure how to do steps 3 and 4. I need to be sure that the agent will be available to take the call and not make the customer wait again in queue. I guess I just need a variable to store the agent ID and pass it to the other script so it can make a queue to agent step, but then how do I make the agent reserved or not ready (so he's not taken by another call) and then present the same call back to him? Then I need a way to monitor the customer's activity in IVR and notify the agent if the customer dropped the call.
Tough one...
Stoyan

Hi Stoyan
I've been looking into the same thing, did you find a solution yet? I had considered letting the Agent do a transfer to the IVR, collect the Agent ID (or if possible the instrument number they were currently logged in with) for routing back to the Agent. Set the Agent to permanent Not Ready so they don't receive any other Contact Centre calls. Let the IVR play out as long as it takes without having to calculate for retries etc. or leaving long delays before the agent can be reached again. If I could only set the AgentID in the transfer script I would perform a database lookup on the ICM real time tables to find out what extension they were currently logged in with and route direct to that (reachable even if in the Not Ready state). The main implication of this would be that I would lose visibility of the remainder of this call with regards the skill group statistics. However, the default skill group could be set for each agent to account for this....not ideal as reporting would become more manual but still recorded at some level.
Regards
Brenda

Transfer & Conference calls from Intercom

How I configure the intercom stations to do transfer or to a conferece calls
We have the normal stations under route patern and CSS diferent to Intercom stations .... Is It My inconvenient?
thank
Juan

You might be running into a bug here. The bug-id is: CSCee00446

Transfer VOIP Calls Between Cisco Desk Phone and Cisco Jabber For IPhone 9.5

Does anyone know how to transfer an active voip call from a Cisco IP Desk Phone to Cisco Jabber for IPhone? I can transfer a call from Cisco Jabber for IPhone to my Cisco IP Desk Phone no problem. I put the call on hold and then click "Resume" on my Cisco IP Desk Phone. However I cannot do the same but the other way around. If I put the call on hold on my Cisco IP Desk Phone, I see "no active call" on my Jabber client. The only information I could find slighlty relevant was using the Mobility Key/Remote Destination Profile feature however this defeats the object as this will forward to an external number, e.g. mobile and I just want to transfer the call within the VOIP environment between the two devices that are using the same directory number.
I am using Cisco Call Manager 9.1(2), Cisco Presence 9.1 and Cisco Jabber for IPhone 9.5.
Any help would be greatly appreciated.
Kind Regards,
Paul Parker.

Did you ever find an answer to this ?
I am seeing the same behavior and trying so see if I can put calls on hold and pick them up both ways also.
The only answer I seem to have found is to use park instead
That would/should work but I would just prefer to hold/unhold
Just not sure why we would not be able to hold/unhold on what is essentially a "shared" line
Does anyone have this working for them ?

JTAPI Get Agent Extension from Agent ID

Dear all,
Please pardon me if have asked this question in the worng place. I'm new to this forums and not familer with the way around.
I wish to build an application using JTAPI connected to UCCX which will provide an CRM integration which will facilitate making calls from agent phones to customer numbers.
I have working code {Attached bellow, which has been refernced} with Agent Extensions but I have faced an issue while integrating with the CRM.
The CRM can only supply the Agent ID not the Agent Extension. My code works using the agent Extension.
I tried but I was not able to find comeup with a soulution which will present the agent extension when the Agent ID is passed.
Please help me to find a way to get the Agent extension using the Agent ID.
Thank you in advance.
--Current code which uses agent extension--
import javax.telephony.*;
import javax.telephony.events.*;
import com.cisco.cti.util.Condition;
public class MakeCall {
public MakeCall(String[] args) throws Exception {
String hostname = args[0];
String login = args[1];
String passwd = args[2];
String src = args[3];
String dst = args[4];
/* start up JTAPI */
JtapiPeer peer = JtapiPeerFactory.getJtapiPeer(null);
/* connect to the provider */
String providerString = hostname;
providerString += ";login=" + login;
providerString += ";passwd=" + passwd;
Provider provider = peer.getProvider(providerString);
/* wait for it to come into service */
final Condition inService = new Condition();
provider.addObserver(new ProviderObserver() {
public void providerChangedEvent(ProvEv[] eventList) {
if (eventList == null)
return;
for (int i = 0; i < eventList.length; ++i) {
if (eventList[i] instanceof ProvInServiceEv) {
inService.set();
inService.waitTrue();
/* get an object for the calling terminal */
Address srcAddr = provider.getAddress(src);
srcAddr.addCallObserver(new CallObserver() {
public void callChangedEvent(CallEv[] eventList) {
/* and make the call */
Call call = provider.createCall();
call.connect(srcAddr.getTerminals()[0], srcAddr, dst);
public static void main(String[] args) {
try {
new MakeCall(args);
} catch (Exception e) {
e.printStackTrace();
} finally {
System.exit(0);

Hi Sudeera,
I am not good at coding :-), but looking into your requirement, you want to know "Agent extension using the Agent ID".
If you access the UCCX Admin->Subsystems->RmCm->Resoures page, click on any resource you will see the Resource ID and the IPCC Extension.
Hope it helps.
Anand
Please rate helpful posts by clicking on the stars below the right answer !!

Transferring call from phone to bluetooth device

My car has built in blue tooth. When I make a call from the phone I have 3 options on the screen:
1. handsfree link (bluetooth)
2. iphone
3. speaker
If I press "handsfree link" it does NOT transfer the call from the phone to the bluetooth until the other party answers. This is really strange, and causes a delay when the other person answers.
Now I know I can use the car's bluetooth to place the call but there are times when I dial from the phone.
Does anyone else have this issue? Is there a work-around for it?
Thanks!

I would contact your car manufacturer as they have all the information of teh bluetooth setup thats in your car. They will have a better idea whats going on as its there system.
Have you tried it with any other phone or car bluetooth if its possible?
Message was edited by: ataylor87

UCCE: Forceful Release an Agent Call from ICM Script, Can I?

Hi, let me explain the requirement first. Customer wants to make their IVR free of cost but they want to start billing only when the call is landed to skill group/agent. So far I can think to make it possible by triggering their billing server by ODBC gateway through Application Gateway process. But also the customer wants to release that particular call when that pre-paid caller is out of charge. They might trigger one of my application or can modify any particular database field and put the calling# into there and my task would be release that call.
I have thought an idea to develop a TCL script run into the voice gateways and release the call from there by searching the particular call with calling#, but I do not know TCL scripting or any idea how to develope TCL , can't I release that call from ICM script? Do I have any control on calls from ICM when the call is landed and connected to agent?
Any help would be hightly appreciated.

That's a nasty piece of work. Just imagine how jacked off you would be if you are the customer, you have enough in your bank account to get to an agent who is then starting to help you, and in the middle of your conversation you are simply cut off!
I don't think it's possible - although CVP would be your best shot because of the switch leg.
But not only that, I don't think it is desirable. If you check the customer's balance before going to an agent, that should be sufficient. Anything else is just terrible customer service.
Regards,
Geoff

Passing ICM peripheral variables from Agent to IVR

Hello,
I need to know if this is possible since i can't manage to make it work.
The call-flow is :
VG->CVP->ICM->Agent
While the call is in the ICM script , i saved the ICM peripheral variable 4 as the language the customer chooses at the beginning of the call.
I pass this variable to the Agent (we are using CAD 8.5) and i can see it in the Agent Layout.
What i need now is to give the Agent the ability to send the call back to any IVR menu (already managed to make that works using single step transfer)
with the ability to pass the peripheral variable 4 to script so that i can use it as the locale instaed of letting the customer chooses the language again.
Is this possible ?
Amer

David ,
Here is the assigment from the script for the local to ICM peripheral variable 4
Then when the call arrived to the agent , i can see the ICM peripheral variable 4 as the value of the language (correctly)
when i use a task (single step transfer) to another script , at the begining i assign the call.user.microapp.locale to PV4 and then i try to play wav file (failure)
see below

Transfer call from 78xx phone series

we are experiencing a problem on phone 78xx series when transfer a call.
When the 7861 telephone transfers a call and the called party does not answer, can not resume the call on hold.
I configure my sip profile in any particular way?

We recently bought some 78xx IP phone and we found the same problem: when I try to transfer a call to another phone (either an internal or external phone) I cannot cancel the transfer, neither by pressing the line button nor the cancel softkey.
Once the cancel button is pressed the call is automatically diverted from the originating caller to the third phone, and is terminated on 78xx phone.
We already installed the latest CUCM version (9.1.2.11900) and phone firmware (sip78xx.10-1-1-9) but the issue is still present.
Hope it will be solved in next firmware update.

Transfer a call from Agent to particular IVR menu

Similar Messages

Maybe you are looking for