Internal call transfer
Hi there,
If a customer gets in the wrong queue the agent transfers the customer to the correct queue by transferring the call to a script that just puts the call into the desire queue.
At the moment this means that every time an agent transfers a call into another queue it shows up in the reports as an new call.
Is there anyway way in the reports internal transfers can be distinguished from an external call?
Thanks
Alexis
The call will be always showed up as new call. The only thing you can try is creating a custom report to know if the call is an internal transfer or an external call.
If you check the 'ContactCallDetail' table, the sessionID will be the same for every leg of the call. When you transfer the call to another script, the sessionID will not change.
Please, check the database schema for more information:
http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_7_0/user/guide/uccx70dbschema.pdf
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022247: *Nov 25 02:08:27.598: %SYS-5-CONFIG_I: Configured from console by admin on vty0 (10.53.5.38)sh run
Building configuration...
Current configuration : 22590 bytes
! Last configuration change at 10:08:27 GMT Tue Nov 25 2014 by admin
version 15.2
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
service sequence-numbers
hostname Cisco_CME
boot-start-marker
boot-end-marker
no logging queue-limit
logging buffered 5000000
no logging rate-limit
no logging console
enable secret 4 tnhtc92DXBhelxjYk8LWJrPV36S2i4ntXrpb4RFmfqY
no aaa new-model
clock timezone GMT 8 0
ip dhcp excluded-address 10.53.4.1 10.53.4.10
ip dhcp pool IPPHONE_POOL
network 10.53.4.0 255.255.255.0
dns-server 10.53.4.2
default-router 10.53.4.2
option 150 ip 10.53.4.2
ip cef
no ipv6 cef
multilink bundle-name authenticated
trunk group POTSTRUNK
crypto pki trustpoint TP-self-signed-2913731135
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-2913731135
revocation-check none
rsakeypair TP-self-signed-2913731135
crypto pki certificate chain TP-self-signed-2913731135
certificate self-signed 01
3082022B 30820194 A0030201 02020101 300D0609 2A864886 F70D0101 05050030
31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
69666963 6174652D 32393133 37333131 3335301E 170D3133 30323034 30343232
32315A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D32 39313337
33313133 3530819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
8100C8B8 A89478A0 8CE7456A 8AAA6EB7 16096839 49A19DF1 D6ABCB99 0C6E9134
951497B5 94538B46 E1193EF4 B6A3B9E7 F3F48299 22B256DD 6D43E5F0 79E2C466
AE742B15 E2F928DE 2166E885 17EBFB45 F36F628C F579BA86 09DBC177 18B50E28
BF278246 722BB661 D9383AD8 49BE2CF2 AEA9FCD6 E1372878 EBEDFDE5 469E3E6B
A1C50203 010001A3 53305130 0F060355 1D130101 FF040530 030101FF 301F0603
551D2304 18301680 141E3A7C 08DDD58E 33243B1D 3CDC300A 959677E7 DD301D06
03551D0E 04160414 1E3A7C08 DDD58E33 243B1D3C DC300A95 9677E7DD 300D0609
2A864886 F70D0101 05050003 81810045 31667A3E 9ADCE589 EC6A9756 EC63BA20
2A5F6B01 71E14526 7A634355 D1B5C7C0 8F2D7602 45B54483 F4FBFBE3 A03F407B
05264954 C18837E9 90072CB6 9AA40BAC 4A0FECB0 DCD17EEC 15BC2CBE 4C27372A
E9048E75 E09AE0EA ED9B871C 487D8163 D7EA0F9D A87D5EAB B750AEF7 FAD8653A
5993BF22 2638198C 6ACBE8B6 EF4365
quit
voice-card 0
dspfarm
dsp services dspfarm
voice call send-alert
voice rtp send-recv
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server expires max 1200 min 300
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
voice register global
mode cme
source-address 10.53.4.2 port 5060
max-dn 200
max-pool 42
load 7942 SCCP42.9-2-1S
load 6921 SCCP69xx.9-2-1-0
authenticate register
authenticate realm archerwell.local
timezone 41
time-format 24
date-format D/M/Y
tftp-path flash:
create profile sync 0324344033123267
voice register dn 1
number 1111
call-forward b2bua unregistered 11111111
allow watch
name 1111
label PolycomCon-110
voice register pool 1
id mac 0004.F2EA.1461
number 1 dn 1
presence call-list
dtmf-relay sip-notify
username 1111 password 1111
codec g711ulaw
voice translation-rule 1111
voice translation-rule 1112
rule 1 /^9/ //
voice translation-rule 2001
rule 5 /^/ /9/
voice translation-rule 2111
voice translation-rule 2112
rule 5 /^6/ //
voice translation-profile CALLER_ID_TRANSLATION_PROFILE
translate calling 1111
voice translation-profile INBOUND_TRANSLATION_PROFILE
translate calling 2001
translate called 2000
voice translation-profile OUTBOUND_NORWAY_TP
translate calling 1111
translate called 2112
voice translation-profile OUTGOING_TRANSLATION_PROFILE
translate calling 1111
translate called 1112
voice translation-profile OUTGOING_TRANSLATION_PROFILE_D9
translate calling 2111
translate called 2112
license udi pid CISCO2901/K9 sn FGL1646215J
hw-module pvdm 0/0
hw-module pvdm 0/1
file privilege 0
username admin privilege 15 secret 4 g1rTD89b38NIXbGJse.zLc7Cega1TBTlKQNvYDh9Qo6
username josadmin password 0 P@ssw0rd
redundancy
gw-accounting aaa
gw-accounting file
primary ifs flash:
acct-template callhistory-detail
interface Loopback0
ip address 10.1.10.2 255.255.255.252
ip virtual-reassembly in
interface Embedded-Service-Engine0/0
ip unnumbered Loopback0
ip virtual-reassembly in
service-module ip address 10.1.10.1 255.255.255.252
service-module ip default-gateway 10.1.10.2
interface GigabitEthernet0/0
no ip address
duplex auto
speed auto
interface GigabitEthernet0/0.1
encapsulation dot1Q 1 native
ip address 10.53.5.2 255.255.255.0
interface GigabitEthernet0/0.10
encapsulation dot1Q 3
ip address 10.53.4.2 255.255.255.0
interface GigabitEthernet0/1
ip address 210.186.148.150 255.255.255.252
shutdown
duplex auto
speed auto
ip forward-protocol nd
ip http server
ip http authentication local
ip http secure-server
ip http path flash:/GUI
ip route 0.0.0.0 0.0.0.0 10.53.5.1
ip route 10.52.100.0 255.255.255.0 10.53.4.1
ip route 10.52.138.0 255.255.255.0 10.53.4.1
ip route 10.54.2.0 255.255.255.0 10.53.4.1
tftp-server flash0:apps42.9-2-1TH1-13.sbn alias apps42.9-2-1TH1-13.sdn
tftp-server flash0:cnu42.9-2-1TH1-13.sbn alias cnu42.9-2-1TH1-13.sdn
tftp-server flash0:cvm42sccp.9-2-1TH1-13.sbn alias cvm42sccp.9-2-1TH1-13.sdn
tftp-server flash0:dsp42.9-2-1TH1-13.sbn alias dsp42.9-2-1TH1-13.sdn
tftp-server flash0:jar42sccp.9-2-1TH1-13.sbn alias jar42sccp.9-2-1TH1-13.sdn
tftp-server flash0:term42.default.loads alias term42.default.loads
tftp-server flash0:term62.default.loads alias term62.default.loads
control-plane
voice-port 0/0/0
timeouts ringing infinity
description FAX
station-id number 21668880
caller-id enable
voice-port 0/0/1
trunk-group POTSTRUNK
supervisory disconnect dualtone mid-call
no battery-reversal
cable-detect
echo-cancel erl worst-case 0
echo-cancel mode 2
cptone MY
timeouts call-disconnect 3
timeouts ringing infinity
timeouts wait-release 3
timing digit 200
timing inter-digit 200
connection plar 200
description FXS
voice-port 0/0/2
trunk-group POTSTRUNK
supervisory disconnect dualtone mid-call
no battery-reversal
cable-detect
echo-cancel erl worst-case 0
echo-cancel mode 2
cptone MY
timeouts call-disconnect 3
timeouts ringing infinity
timeouts wait-release 3
timing digit 200
timing inter-digit 200
connection plar 200
description FXS
voice-port 0/0/3
trunk-group POTSTRUNK
supervisory disconnect dualtone mid-call
no battery-reversal
cable-detect
echo-cancel erl worst-case 0
echo-cancel mode 2
cptone MY
timeouts call-disconnect 3
timeouts ringing infinity
timeouts wait-release 3
timing digit 200
timing inter-digit 200
connection plar 200
description FXS
voice-port 0/1/0
trunk-group POTSTRUNK
supervisory disconnect dualtone mid-call
no battery-reversal
cable-detect
echo-cancel erl worst-case 0
echo-cancel mode 2
cptone MY
timeouts call-disconnect 3
timeouts ringing infinity
timeouts wait-release 3
timing digit 200
timing inter-digit 200
connection plar 200
description FXS
voice-port 0/1/1
trunk-group POTSTRUNK
supervisory disconnect dualtone mid-call
no battery-reversal
cable-detect
echo-cancel erl worst-case 0
echo-cancel mode 2
cptone MY
timeouts call-disconnect 3
timeouts ringing infinity
timeouts wait-release 3
timing digit 200
timing inter-digit 200
connection plar 200
description FXS
voice-port 0/1/2
trunk-group POTSTRUNK
supervisory disconnect dualtone mid-call
no battery-reversal
cable-detect
echo-cancel erl worst-case 0
echo-cancel mode 2
cptone MY
timeouts call-disconnect 3
timeouts ringing infinity
timeouts wait-release 3
timing digit 200
timing inter-digit 200
connection plar 200
description FXS
voice-port 0/1/3
trunk-group POTSTRUNK
supervisory disconnect dualtone mid-call
no battery-reversal
cable-detect
echo-cancel erl worst-case 0
echo-cancel mode 2
cptone MY
timeouts call-disconnect 3
timeouts ringing infinity
timeouts wait-release 3
timing digit 200
timing inter-digit 200
connection plar 200
description FXS
mgcp fax t38 ecm
mgcp profile default
dial-peer voice 1 pots
destination-pattern 21668880
port 0/0/0
dial-peer voice 2 pots
service stcapp
port 0/0/1
dial-peer voice 3 pots
service stcapp
port 0/0/2
dial-peer voice 4 pots
service stcapp
port 0/0/3
dial-peer voice 54 pots
trunkgroup POTSTRUNK
description ** FXO pots dial-peer **
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 5
shutdown
destination-pattern 9.T
direct-inward-dial
no sip-register
dial-peer voice 104 pots
description "Incoming Call to DN"
translation-profile incoming INBOUND_TRANSLATION_PROFILE
destination-pattern 1...
incoming called-number .%
direct-inward-dial
port 0/0/0
dial-peer voice 2000 voip
destination-pattern 8..
session protocol sipv2
session target ipv4:10.52.138.41
dtmf-relay sip-notify
codec g711ulaw
dial-peer voice 105 pots
description "Incoming Call to DN"
translation-profile incoming INBOUND_TRANSLATION_PROFILE
destination-pattern 1...
incoming called-number .%
direct-inward-dial
port 0/0/0
dial-peer voice 106 pots
description "Incoming Call to DN"
translation-profile incoming INBOUND_TRANSLATION_PROFILE
destination-pattern 1...
incoming called-number .%
direct-inward-dial
port 0/0/2
dial-peer voice 107 pots
description "Incoming Call to DN"
translation-profile incoming INBOUND_TRANSLATION_PROFILE
destination-pattern 1...
incoming called-number .%
direct-inward-dial
port 0/0/3
dial-peer voice 313 pots
trunkgroup POTSTRUNK
description ** Mobile No. (8-digits) **
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE_D9
destination-pattern 69011........
direct-inward-dial
dial-peer voice 306 pots
trunkgroup POTSTRUNK
description ** Mobile No. (7-digits) **
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE_D9
destination-pattern 6901[0,2-9].......
direct-inward-dial
dial-peer voice 307 pots
trunkgroup POTSTRUNK
description ** Outstation No. **
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE_D9
destination-pattern 690[2,4-7,9]T
direct-inward-dial
dial-peer voice 308 pots
trunkgroup POTSTRUNK
description ** Sabah/Sarawak **
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE_D9
destination-pattern 6908[2-9]T
direct-inward-dial
dial-peer voice 310 pots
trunkgroup POTSTRUNK
description ** Toll-Free **
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE_D9
destination-pattern 691[3,6,8]00......
direct-inward-dial
dial-peer voice 311 pots
trunkgroup POTSTRUNK
description ** IDD to Singapore **
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE_D9
destination-pattern 690065[0,1-9].......
direct-inward-dial
dial-peer voice 312 pots
trunkgroup POTSTRUNK
description ** IDD to the rests **
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE_D9
destination-pattern 6900T
direct-inward-dial
dial-peer voice 2001 voip
destination-pattern 7..
session protocol sipv2
session target ipv4:10.54.2.2
dtmf-relay sip-notify
codec g711ulaw
dial-peer voice 5 pots
service stcapp
port 0/1/0
dial-peer voice 6 pots
service stcapp
port 0/1/1
dial-peer voice 7 pots
service stcapp
port 0/1/2
dial-peer voice 8 pots
service stcapp
port 0/1/3
dial-peer voice 108 pots
description "Incoming Call to DN"
translation-profile incoming INBOUND_TRANSLATION_PROFILE
destination-pattern 1...
incoming called-number .%
direct-inward-dial
port 0/1/0
dial-peer voice 109 pots
description "Incoming Call to DN"
translation-profile incoming INBOUND_TRANSLATION_PROFILE
destination-pattern 1...
incoming called-number .%
direct-inward-dial
port 0/1/1
dial-peer voice 110 pots
description "Incoming Call to DN"
translation-profile incoming INBOUND_TRANSLATION_PROFILE
destination-pattern 1...
incoming called-number .%
direct-inward-dial
port 0/1/2
dial-peer voice 111 pots
description "Incoming Call to DN"
translation-profile incoming INBOUND_TRANSLATION_PROFILE
destination-pattern 1...
incoming called-number .%
direct-inward-dial
port 0/1/3
dial-peer voice 305 pots
trunkgroup POTSTRUNK
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE_D9
destination-pattern 69[2-9].......
direct-inward-dial
gatekeeper
shutdown
telephony-service
max-ephones 40
max-dn 100
ip source-address 10.53.4.2 port 2000
auto assign 1 to 40
service phone ehookEnable 1
service directed-pickup gpickup
service dnis dir-lookup
timeouts interdigit 4
cnf-file location flash:
load 7916-12 B016-1-0-4.SBN
load 7942 SCCP42.9-2-1S
load 7962 SCCP42.9-2-1S
load 6921 SCCP69xx.9-2-1-0
time-zone 43
date-format dd-mm-yy
max-conferences 8 gain -6
moh "flash:/music-on-hold.au"
web admin system name admin password P@ssw0rd
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn-template 1
ephone-template 1
softkeys idle Redial Newcall Cfwdall Gpickup Dnd
softkeys connected Hold Endcall Trnsfer Park Confrn ConfList
ephone-dn 1
number 100
pickup-call any-group
pickup-group 1
ephone-dn 2
number 101
pickup-call any-group
pickup-group 1
ephone-dn 3
number 102
pickup-call any-group
pickup-group 1
ephone-dn 4
number 103
pickup-call any-group
pickup-group 1
ephone-dn 5
number 105
pickup-call any-group
pickup-group 1
ephone-dn 6
number 104
pickup-call any-group
pickup-group 1
ephone-dn 7
number 106
pickup-call any-group
pickup-group 1
ephone-dn 8
number 107
pickup-call any-group
pickup-group 1
ephone-dn 9
number 116
pickup-call any-group
pickup-group 1
ephone-dn 10
number 109
pickup-call any-group
pickup-group 1
ephone-dn 11
number 111
pickup-call any-group
pickup-group 1
ephone-dn 12
number 112
pickup-call any-group
pickup-group 1
ephone-dn 13
number 113
pickup-call any-group
pickup-group 1
ephone-dn 14
number 114
pickup-call any-group
pickup-group 1
ephone-dn 15
number 115
pickup-call any-group
pickup-group 1
ephone-dn 16
number 108
pickup-call any-group
pickup-group 1
ephone-dn 17
number 117
pickup-call any-group
pickup-group 1
ephone-dn 18
number 118
pickup-call any-group
pickup-group 1
ephone-dn 19
number 119
pickup-call any-group
pickup-group 1
label Alex
name Alex
ephone-dn 20
number 120
pickup-call any-group
pickup-group 1
ephone-dn 21
number 121
pickup-call any-group
pickup-group 1
ephone-dn 22
number 122
pickup-call any-group
pickup-group 1
label Acher Well MY 8
name Acher Well MY 8
ephone-dn 23
number 123
pickup-call any-group
pickup-group 1
ephone-dn 24
number 124
pickup-call any-group
pickup-group 1
label OilTools
name OilTools
ephone-dn 25
number 125
pickup-call any-group
pickup-group 1
label Guest
name Guest
ephone-dn 26
number 321
description IT ServiceDesk
call-forward all 690048717608503
ephone-dn 27
number 127
pickup-call any-group
pickup-group 1
ephone-dn 28
number 128
pickup-call any-group
pickup-group 1
ephone-dn 29
number 129
pickup-call any-group
pickup-group 1
ephone-dn 30
number 130
pickup-call any-group
pickup-group 1
label Reception
name Reception
ephone-dn 39
number 188
park-slot directed timeout 60 limit 2 transfer 100
ephone 1
mac-address 1833.9D15.43C2
ephone-template 1
type 7942
button 1:8
ephone 2
mac-address 1CE6.C799.7E7E
ephone-template 1
type 6921
button 1:3
ephone 3
mac-address 1CE6.C799.7EF9
ephone-template 1
type 6921
button 1:4
ephone 4
mac-address 1CE6.C799.3F86
ephone-template 1
type 6921
button 1:2
ephone 5
mac-address 1CE6.C799.7E43
ephone-template 1
type 6921
ephone 6
mac-address 10BD.1800.7793
ephone-template 1
type 7942
button 1:7
ephone 7
mac-address 1CE6.C799.83C2
ephone-template 1
type 6921
ephone 8
mac-address 10BD.1800.7807
ephone-template 1
type 7942
button 1:19
ephone 9
mac-address 1833.9D15.C947
ephone-template 1
type 7962
button 1:6
ephone 10
mac-address A418.758A.F2FF
ephone-template 1
type 7962
button 1:10
ephone 11
mac-address 1833.9D15.E9ED
ephone-template 1
type 7962
button 1:11
ephone 12
mac-address 1833.9D14.4058
ephone-template 1
type 7962
button 1:12
ephone 13
mac-address 1833.9D15.EA99
ephone-template 1
type 7962
button 1:13
ephone 14
mac-address A418.7528.2A3C
ephone-template 1
type 7962
button 1:14
ephone 15
mac-address 44AD.D9D4.0A96
ephone-template 1
type 7965
button 1:15
ephone 16
mac-address 44AD.D9D4.1717
ephone-template 1
type 7965
button 1:16
ephone 17
mac-address 44AD.D9D4.1746
ephone-template 1
type 7965
button 1:17
ephone 18
mac-address E8ED.F3AB.0CEC
ephone-template 1
type 7965
button 1:18
ephone 19
mac-address E8ED.F3AB.0BD6
ephone-template 1
presence call-list
type 7965 addon 1 7916-12 2 7916-12
button 1:30
ephone 20
mac-address E8ED.F3AB.0EBC
ephone-template 1
speed-dial 1 69001800815412 label "Operation Meeting"
type 7965
button 1:20
ephone 21
mac-address 5C50.15A8.0D96
ephone-template 1
type 7962
button 1:21
ephone 22
mac-address 1833.9D15.EA88
ephone-template 1
type 7962
button 1:22
ephone 23
mac-address 1833.9D15.E9D8
ephone-template 1
type 7962
button 1:23
ephone 24
mac-address 1833.9D15.EA02
ephone-template 1
type 7962
button 1:24
ephone 25
mac-address 1833.9D15.C558
ephone-template 1
type 7962
button 1:25
ephone 26
mac-address 1833.9D15.C5D0
ephone-template 1
type 7962
button 1:4
ephone 27
mac-address A418.7529.95FA
ephone-template 1
type 7962
button 1:5
ephone 28
mac-address 1833.9D15.8D05
ephone-template 1
type 7962
button 1:1
ephone 29
mac-address 1833.9D15.EAA9
ephone-template 1
type 7962
button 1:2
ephone 30
mac-address 1833.9D15.EA0F
ephone-template 1
type 7962
button 1:9
ephone-hunt 1 sequential
pilot 200
list 130, 100, 103
final 103
timeout 20, 20, 10
statistics collect
present-call onhook-phone
ephone-hunt 2 sequential
line con 0
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
password cisco
login local
transport input all
line vty 5 15
login
transport input all
scheduler allocate 20000 1000
ntp server 119.110.97.148 prefer
end
Thank youHi,
Can you post the output from "debug voice ccapi inout"? This will tell us which Dial-Peer it's matching.
Thanks
Rob -
Be careful about using your phone for data transfer in Europe! I Just got back from a 2-week trip to Europe with my family of four- we all brought our iPhones. I had international roaming- World Traveller- turned on before we left. I was told that phone calls would be $1.99 per minute and data transfer (Internet, email, etc.) would be unlimited because we already pay the $20 for that. It turns out it is NOT unlimited. We just got a bill for $3700. AT&T has agreed they told us wrong and they will credit our data usage charges but I'm still not sure it is really going to happen. They have since instituted a new plan something like this: International Data Transfer is about $20 per month for about 15mb of data. But there is a 1 year commitment for this plan. Hopefully, they will revise this plan to something more useful.
Yesterday afternoon the advice I received from AT&T was to call and have the data part of the phone turned off while traveling. That would, I was told, keep me from getting a big bill. This morning I telephoned again to ask about wi fi, and was told, essentially, that if wi fi is free where I am, there would be no charge for web browsing or reading and sending email. Also, I was told that the advice I received yesterday was incorrect. The way to keep from getting the data charges that are very large is to go to mail settings and for each account, to turn it "off." That way the phone will not look for email. Presumably not clicking the Safari icon is a part of this plan. So the work around today sounds like it is to use free wi fi if you can find it.
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Call transfer through Hunt group
Hi,
we have a call manager version 8.6 where we have mutiple huntgroups configured. Following is the call flow
PSTN callers call a DID number from outside which after the translation on the gateway, hits a hunt pilot number on call manager and eventually the call rings on all the line members of the line group. One any of the line group member picks up the call and tries to transfer the call to any internal or external number, we get the following error message on phone screen and transfer does not work
" External transfer Restricted"
Now this happens even if we dial the hunt pilot number internally from an iphone and then if one of the members tries to transfer to an internal extension or external number, we get the error message
More Information:
There are many other hunt groups on the same call manager server and tries to transfer the call same way as the non working one and it works for them
So i checked the difference. All the members of all the hunt groups are device profile logged into physical phone. The only difference was that working hunt group members are logged into 6921 phone where as non working users are logged into 7942 phones (sip firmare) and the profile are created for 9971 although we dont have any physical 9971 phone
So it seems that call transfer through hunt group does not work only if the members are logged into 7942 (sip) phones and works when they are logged into 6921 with 9971 Device profile
I hope its clear. Please let me know if its a known issue or limitation with 7942 phone. Any help will be appreciatedTry changing the Block Offnet to Offnet Transfer CallManager Service Parameter to False and test again. If it then works, we'll probably need CallManager traces to see why both numbers are getting marked as offnet.
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Passing CLID and AgentID for CVP Call Transfer
Hi, all.
Currently we are using CVP 7.0(2) integrated with ICM 7.5(9), Just want to check if the following is feasible:
1. Caller calls the hotline being routed to a queue.
2. An agent picks up the call.
3. Agent does a consult transfer to another ICM routing script configured with another Dialed Number. It is a post call survey.
4. The post call survey takes over the call when the agent complete the call transfer
5. The caller's no (CLID) and AgentID are passed over to the ICM Routing script of this post call survey.
Can we achieve the above mentioned scenario particularly on item 5? Would appreciate any help and advice.
Thanks & Regards,
EricHi, Geoff.
Thanks for your reply. I have copied and pasted here a list of steps which you have posted previously about the CVP Warm Transfer.
1. PG Explorer - CUCM PIM Routing Client - Network Transfer Preferred not checked
2. PG Explorer - CVP PIM Routing Client - Network Transfer Preferred not checked
3. PG Explorer - CUCM PIM Advanced - Network VRU - NONE
4. PG Explorer - CVP PIM Advanced - Network VRU - Type 10
5. NVRU Explorer - Type 10 Network VRU, label for the CUCM routing client associated with the customer instance. Let's say this is 8222222222.
6. NVRU Explorer - Type 10 Network VRU, label for the CVP routing client associated with the customer instance. Let's say this is 8111111111.
7. Dialed Number List - dialed number for the incoming call associated with the customer instance. This dialed number is on the CVP PIM Routing Client. This DN is associated with a call type which is then mapped to the initial script.
8. Dialed Number List - transfer dialed number associated with the customer instance. This dialed number is on the CUCM PIM Routing Client. The transfer dialed number 3151 is associated with a call type which is mapped to the transfer script.
9. DNP. The number transferred to from CAD is 3141 which is a pattern in the DNP that maps to the Dialed Number 3151 with a post route to CUCM PIM Routing Client. The DNP Type is "PBX" - and PBX is set up in the Agent Desk Settings
10. Agent Desk Settings - All but "Operator" are checked
11. When the second call is placed for the warm transfer, the label defined on the CUCM RC plus the correlation ID will be sent back via EAPIM/JGW to CUCM (for example, if the label is 8222222222, with a correlation ID it could be 822222222212345) since the call originated from the CUCM RC and since the NetworkTransferPreferred check box is not checked.
12. A route pattern 8222222222! in CUCM sends the call down a SIP trunk to CUPS.
13. CUPS has a static route on 8222222222* to send the call to the CVP Call Server.
14. CUPS has a static route on 8111111111* to get the IP call to the gateway. Note that in a branch office deployment, TDM calls into the gateway use "Send to Originator" pattern in the Call Server to force the transfer label back to the voice gateway; so this pattern in CUPS is ONLY used by VoIP calls.
15. In all preliminary scripts that get the customer to the agent, set the variable Call.NetworkTransferEnabled to the value 1. This is set before the transfer is called.
18. For the device targets, you need a label on the CVP RC, but you do not need one on the CUCM RC, so do not add one.
I did try the above mentioned steps in my system, the transfer works alright. The only difference is that I did not set the variable Call.NetworkTransferEnabled to the value 1. To the ICM, is is considered as Internal Out Transfer or External Out? Another thing I have noticed is that when the call transfer takes place, the IP phone would display the Route Pattern of the CM Label + correlation ID configured, e.g. 822222222212345. this is rather confusing for the agents when they see this long string of number as it does not make sense to them. Is there any way to display some other meaningful line display or number (E.g. Transfer to mainline or 2900) while maintaining the CM label pattern 8222222222!?
Thanks in advanced.
Regards,
Eric -
LINKSYS spa9000 call transfer problem
Hi, I have been reading through the forums and changing settings without success. If I set the handset to callfwd then it rings and then goes silent when an outside caller is connected. Also the same occurs when I try to transfer a call. However if I try to do the same with a call which has been initiated between internal handsets then it works fine - any ideas? I have already changed the settings on the bridge mode and enabled force proxy. Hope this makes sense Phones are spa941/942 Cheers
Can I clarify at least the call transfer part of your posting? Is it the case that: - an external caller calls in, and is answered by one of your SPA9xx phones - you then do an 'attended call transfer' to transfer the call to another internal extension - the other extension answers, but on completion of the transfer does not have two-way audio with the external caller Is that the scenario you have?
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Music on Hold and Call Transfer
Got a weird issue that popped up recently. When a user places a call on hold and then transfers the call to another Lync user, the caller will hear the ring back transfer tone but also hear the music on hold playing in the background. This just started recently
after a reboot of the front end. Anyone have any ideas on what can be causing this. Below is some information that can be helpful. Thanks.
-Trunk is configured with no refer support since our SIP provider's gateway does not support it.
-Music on hold is set globally for all users with a user policy. The music on hold file is the default Lync .wav file.
-This happens with calls coming in from the PSTN, internal Lync calls that are transferred to other Lync users, and calls that come into an RGS and are transferred.Hi,
Please update to the latest version for Lync server 2013 and Lync client 2013 on Microsoft Website and then try again:
http://technet.microsoft.com/en-us/lync/dn146015
Please also try to disable Music on hold, then able Music on hold again and test again.
If the issue persists, you can use the Lync server logging tool to test the process of call transfer.
If Lync server 2013 not have Lync server logging tool, you can download Logging debugging tool in link below for troubleshooting.
http://www.microsoft.com/en-in/download/details.aspx?id=35453
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support -
H323 Gateway International calls fails
Hello,
International call fails from one of the PRI port 0/2/0, but national and GSM works fine, cause code (31) need advice from experts on CSC.
Attached are the debug output from the working PRI and non working PRI & configuration.
thanksFor the successful call, once you send the call to the telco, it is being routed by the telco.
Aug 5 08:45:33.771: ISDN Se0/1/0:15 Q931: TX -> SETUP pd = 8 callref = 0x4930
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98387
Exclusive, Channel 7
Calling Party Number i = 0x0081, '877'
Plan:Unknown, Type:Unknown
Called Party Number i = 0x80, '24773376'
Plan:Unknown, Type:Unknown
Aug 5 08:45:33.807: ISDN Se0/1/0:15 Q931: RX <- SETUP_ACK pd = 8 callref = 0xC930
Channel ID i = 0xA98387
Exclusive, Channel 7
Aug 5 08:45:34.859: ISDN Se0/1/0:15 Q931: RX <- CALL_PROC pd = 8 callref = 0xC930
Progress Ind i = 0x8488 - In-band info or appropriate now available
The Telco is responding with Call proceeding which means the call is being routed, then an alerting. So they are reaching the number called.
However, in the non-working case, the telco is sending a disconnect. You will need to ask the telco why we are receiving a disconnect from them. Is there some information that we are not sending them?
Do we need to send a different format of digits? Or do we need to change the type of number?
Once they answer those questions or check their side, the problem should get solved.
Aug 5 08:40:43.551: ISDN Se0/2/0:15 Q931: TX -> SETUP pd = 8 callref = 0x491F
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98382
Exclusive, Channel 2
Calling Party Number i = 0x0081, '080'
Plan:Unknown, Type:Unknown
Called Party Number i = 0x80, '00912223710485'
Plan:Unknown, Type:Unknown
Aug 5 08:40:43.583: ISDN Se0/2/0:15 Q931: RX <- SETUP_ACK pd = 8 callref = 0xC91F
Channel ID i = 0xA98382
Exclusive, Channel 2
Aug 5 08:40:43.639: ISDN Se0/2/0:15 Q931: RX <- DISCONNECT pd = 8 callref = 0xC91F
Cause i = 0x829F - Normal, unspecified
Facility i = 0x91A10B02017F0606040087690107
Protocol Profile = Remote Operations Protocol
0xA10B02017F0606040087690107
Component = Invoke component
3PTY Request
Progress Ind i = 0x8288 - In-band info or appropriate now available -
I am not able to make International calls from iPhone 5s
I got this new iphone 5s as a gift a week days ago, and It was working all great for 2-3 days. But from last 4days i AM not able to make international calls
Whenever I dial, I can't hear any ring going on and all I can hear is my own voice reverting back to me. Moreoever I am being charged for every call I make. I tried restarting my device, Also by connecting to itunes to find if any Update is needed but it says my iOS7 is recent no updates needed.
Urgent help needed
Thanks in advance.A call is a call so far as the phone is concerned. Internation or across the street, it doesn't diferentiate between them.
Contact your carrier. -
I changed my data plan from 6g to 8g while my daughter who attends college outside of the US at Toronto Canada (and we have on a international calling and international data plan) was on spring break at her grandparents house here in the US. I made the change online since I had been waiting on the phone for over 10 minutes for a customer service rep to come available. Well when I made the change online since that seems to be the thing that Verizon wants it's customers to do and I didn't see all the different plans available and just did the upgrade to 8g. Next bill had over $900 in roaming charges on her phone line. I called the 1-800 number and waiting for a service rep and after 20 minutes of waiting and being put on hold was told it was the customers mistake and there was nothing they could do.Thanks for nothing. I called back after thinking about it and wondered why changing a data plan for the phones in the US would change a international call plan. Waiting over 10 minutes again between waiting for a service rep and hold for one to answer the call. Gave her all the information about it and she said she would call back. Well, 4 days later over the weekend she had nevered called back. So on the phone again for the third time and after 20 plus minutes again was told that when I did it online I click the plan that didn't include international call only the data plan. Explained that I never saw the difference in the plan packages so put on hold again and was told that they could credit $100 to my bill. Wow, thanks alot !!! We have been Verizon customers for probably atleast 12 years and this is how you treat your long term customers?
Verizon Wireless Customer Support wrote:
AHARDY454,
We definitely want to review options on what has happened. We are now connection, so you can hover over my username and send me a Direct Meesage so we can review your account information. We look forward to reviewing.
Thank you,
TonyG_VZW
Follow us on Twitter @VZWSupport
TonyG_VZW they can't exactly hover over your username unless you actually link it in your post. The generic username for all the reps just doesn't fly. -
Local call transfer not working on CME10 for 7821
My setup is made of 4 cisco 7821 phones ( phone A, Phone B ,Phone C and Phone D)
when phone B get a call from phone A and transfer call to phone D or C
phone B is able to complete the transfer but phone A get a busy tone.
Problem is only with call transfer conferencing is working
My CME router is running ios image c2900-universalk9-mz.SPA.153-3.M3.bin.
find attached configuration
What i am doing wrong ?
Can someone please help?Hello,
You are missing your call-transfer config, i guess:
telephony-service
transfer-system {blind | full-blind | full-consult | local-consult}
transfer-pattern transfer-pattern [blind] !!!... In case consultive transfer is enabled globally, blind transfer can be applied to specific patterns using blind keyword
call-forward pattern pattern
timeouts transfer-recall seconds
calling-number local !!!... This command will replace the calling-party name/number with the forwarding-party name/number for hairpin forwarded calls.
transfer-digit-collect {new-call | orig-call}
Please, let me know the results
Best Regards -
Set up prepaid calling card for international calls?
I have a prepaid calling card that I use for calling international. The way it works is that I have to call a 866 number, followed by entering a 10 digit code, and then proceed to calling the actual number (with international prefix codes). Is there a way to set this up as international call default so that I do not have to modify all my international contact's phone numbers to include this procedure?
Thanks.
Kspeed dial: Your smartphone can store nine
phone numbers that you can call
with a single keypress. To store a
speed dial number, touch Phone
> Menu > Speed dial setup, and
insert a number in an empty speed
dial slot.
To call a speed dial number, touch
and hold the single-digit speed dial
number in the dialer screen. -
International calls not going through
Just I have paid for 800 min India subscribtion and calls are not going through
Regards
SubhashHello,
We don't have any reported issues of call malfunction. (There aren't any "international" calls with Skype). All Skype calls are "local" worldwide including calls from one country to another.
Please contact customer service
TIME ZONE - US EASTERN. LOCATION - PHILADELPHIA, PA, USA.
I recommend that you always run the latest Skype version: Windows & Mac
If my advice helped to fix your issue please mark it as a solution to help others.
Please note that I generally don't respond to unsolicited Private Messages. Thank you. -
International Calls - Pakistan to UK / USA and Eur...
As with other international call rates seen on Skype, is it possible to have a flat rate per month with Skype if you want to make unlimited internation calls from Pakistan to the UK / Europe and possibly the USA?
Rauch_MSH wrote:
As with other international call rates seen on Skype, is it possible to have a flat rate per month with Skype if you want to make unlimited internation calls from Pakistan to the UK / Europe and possibly the USA?
Hello
ALWAYS start your own thread please.
With Skype your location is irrelevant. There are no international calls. All calls are local worldwide. You can purchase Subscriptions for calls to/within almost any country in the world.
http://www.skype.com/en/rates/#FR
TIME ZONE - US EASTERN. LOCATION - PHILADELPHIA, PA, USA.
I recommend that you always run the latest Skype version: Windows & Mac
If my advice helped to fix your issue please mark it as a solution to help others.
Please note that I generally don't respond to unsolicited Private Messages. Thank you. -
I am on holiday and have had the Samsung S5 for a few weeks without any problems. I upgarded to the new EE tarrif with roaming minutes and texts as I travel a lot with work. Having arrived on holiday I find I cannot make calls to uk numbers nor to 150 and am redirected to 1407 which is in Spanish only - wonderful idea EE. After dumping 15euros into a public phone I am told to check the setting for Call Barring - Voice Call - International calls to find it is enabled and EE cannot tell me th password but assured me we haved moved forward on the problem (without solving it) - andother great service EE. Can anyone who has experinved this help with the code to disable it please? It is the same on my wife´s handeset where it is enabled too. Thanks in Advance
Sorted! Called EE again and spoke to Eileen who was very cheerful and determined to help me. After she had a couple of conversations with level 2 all she had to do was reset the sim. She called me back after this had been done, about 30 mins, I powered off then on and the tick simply disappeared. So persistence is required, actually seems to be a simple fix if you talk to the right person. Seems to me that some training, or at least a memo, is required for the whole customer service team. Of course, this odd anamoly should not even be there, but you don't know about it until it is too late, but all EE customer services team should be aware and know what to do. Good luck!
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