Transfer a call

Hi - is it still not possible to transfer a call on the iPhone? We cannot use it for business ? iPhone 4) thank you

That would be a carrier feature. Ask your carrier.

Similar Messages

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 ?

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

How can I transfer a call to someone else?

Hi all,
how can I transfer a call (that I've answered and noticed that it's someome elses business) to someone else with Lumia 920 / 820 or 610? Is this basic function even possible?
(It was with all the older phones).
Please help

All the older Nokia phones you answer a call – you can put it on hold – call someone else, tell him/her that you are transferring a call – press 4 and hang up. Those two can continue the conversation together.
That’s what I’m looking for.
With Lumia I don’t want to be the 3rd wheel in the conversation...
I haven't found the way to end the call so that the others could continue (not even with conference call)

The Response Group application was unable to transfer the call to the configured destination and no fallback exists

Hey,
I get this error message when calling into an unassigned number which redirects to a response group:
From user URI:
sip:[email protected];gruu;opaque=srvr:Microsoft.Rtc.Applications.Acd:RS6nRGV9DlmpNsLtmz5qeQAA
To user URI:
0220198611;phone-context=DefaultProfile
From user agent:
RTCC/4.0.0.0 Response_Group_Service Announcement_Service
Diagnostic header:
26005; reason="The Response Group application was unable to transfer the call to the configured destination and no fallback exists."
Interestingly "To user URI: 0220198611;phone-context=DefaultProfile" is the number off the caller not the destination. I wonder is this a bug? So is the response group trying to transfer to this number and failing because of course it doesnt exist?
As you can see the below the number I am calling is not 0220198611:
From phone number: 0220198611;phone-context=DefaultProfile
To phone number: +6493760053 From mediation server: onzlyncfe1.domain.co.nz To mediation server: From gateway: 192.168.100.70
To gateway:
Disconnected by: +6493760053
Does the calling party's number have to be normalised? If so how can I do this because the global normailisation rules dont seem to apply
in this situation. These rules do work when when calling into a users DDI.
Also to be clear....
+6493760053 is an unassigned number which is setup to redirect to a response group.
If I assign +6493760053 to a user then it works.
Additionally this works perfectly when the gateway sends the call to our legacy 2007r2 mediation server then on to Lync. If the gateway sends the call directly to the co-located Lync mediation server I get the error described.
I hope I make sense. If you are confused let me know :)
Help is appreciated.
Thanks,
Andrew

Hi ANdrew
Kindly advise how you transfered the unassigned numbers to a specific user, i used the below command but it failled, the message displayed but the call never routed:
New-CsAnnouncement -Parent service:ApplicationServer:LyncFE.squareone.local -Name "SQ unassigned number announcement" -TextToSpeechPrompt "You entered an invalid extinsion you will be forwarded to the operator" -Language "en-US" -TargetUri "sip:[email protected];user=phone"
While [email protected] is the sip uri in my lync for the operator
could you advise what is my issue?

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.

CTI OS Toolkit - Buttons are disable when transfer the call

Hello guys, i've been in a bad situation and i don´t know how to resolve the problem.
Here where I work, I installed the CTI Toolkit, but when a agent try to transfer the call to another agent, the toolkit disable all the functions, but it still on, the agent still logged,receive others call, but he can't do anything, because all the buttons are disabled.
I found a workaround, when occour this, i reset the IP Cisco 7960 Hardphone, so I logged out the agent and logged in again.
I collect the cti server log, cti os log and the pg log, and I found something, but i don't know how to resolve this problem. Here is a little part of the logs, but I have all the logs completely here, if you want.
Here is the problem: "The agent Simone Minelli (Id: 5012, ext: 2215) reported that days 17.09.2010 at 10:25 until 10:26, the CTI OS toolkit lost
their task and got the buttons disabled, thereafter transfer the call to the agent Laryssa Oliveira (ID: 5096, ext: 2202)".
In appended, i fixed the cti os log, cti server log and the pg log. Thanks everybody!!
Here is only the cti os log, but fixed there are others:
10:25:32 CTIOS1-ctios Trace: [agent.5000.5012] AGENT_STATE_EVENT ( eTalking ) [CtiosBaseAgentObject,
     T_BASE_AGENT_AGENT_STATE_EVENT, 89501]
10:25:32 CTIOS1-ctios Trace: [call.5000.17056904] RTP_STARTED_EVENT [CtiosBaseCallObject,
     T_BASE_CALL_RTP_STARTED_EVENT, 40001]
10:25:44 CTIOS1-ctios Trace: ClientMgr[1]::OnConnectionClosed, Client[0x40754a0][01211]
10:26:29 CTIOS1-ctios Trace: [call.5000.17056970] CALL_FAILED_EVENT [CtiosBaseCallObject,
     T_BASE_CALL_CALL_FAILED_EVENT, 2751]
10:26:52 CTIOS1-ctios Trace: [call.5000.17056904] CALL_CONNECTION_CLEARED_EVENT [CtiosBaseCallObject,
     T_BASE_CALL_CALL_CONNECTION_CLEARED_EVENT, 39001]
10:27:24 CTIOS1-ctios Trace: [skillgroup.5000.40] QUERY_SKILL_GROUP_STATISTICS_REQ
     [CtiosBaseSkillGroupObject, T_BASE_SKILL_QUERY_SKILL_GROUP_STATISTICS_REQ,
     788751]
10:27:24 CTIOS1-ctios Trace: [skillgroup.5000.40] QUERY_SKILL_GROUP_STATISTICS_CONF
     [CtiosBaseSkillGroupObject, T_BASE_SKILL_QUERY_SKILL_GROUP_STATISTICS_CONF,
     788751]
10:27:51 CTIOS1-ctios Trace: [agent.5000.5012]::DisableSkillGroupStatisticsReq()- Empty skill group number
     list provided.[CtiosBaseAgentObject,
     T_BASE_AGENT_DISABLE_EMPTY_SKILLGROUP_LIST, 985]
10:27:51 CTIOS1-ctios Trace: [agent.5000.5012]::DisableSkillGroupStatisticsReq()- Empty skill group number
     list provided.[CtiosBaseAgentObject,
     T_BASE_AGENT_DISABLE_EMPTY_SKILLGROUP_LIST, 986]
10:27:51 CTIOS1-ctios Trace: ClientMgr[1]::OnConnectionClosed, Client[0xf841a8][SILVMIN1-VM-2872-2796]
10:29:41 CTIOS1-ctios Trace: [skillgroup.5000.35] QUERY_SKILL_GROUP_STATISTICS_REQ
     [CtiosBaseSkillGroupObject, T_BASE_SKILL_QUERY_SKILL_GROUP_STATISTICS_REQ,
     789001]
10:29:41 CTIOS1-ctios Trace: [skillgroup.5000.38] QUERY_SKILL_GROUP_STATISTICS_CONF
     [CtiosBaseSkillGroupObject, T_BASE_SKILL_QUERY_SKILL_GROUP_STATISTICS_CONF,
     789001]
10:30:03 CTIOS1-ctios Trace: [call.5000.17057099] CALL_DATA_UPDATE_EVENT [CtiosBaseCallObject,
     T_BASE_CALL_CALL_DATA_UPDATE_EVENT, 37001]
10:30:39 CTIOS1-ctios Trace: [agent.5000.5056] AGENT_STATE_EVENT ( eAvailable ) [CtiosBaseAgentObject,
     T_BASE_AGENT_AGENT_STATE_EVENT, 89751]
10:30:49 CTIOS1-ctios Trace: ClientMgr[1]::OnConnectionClosed, Client[0x40754a0][01213]
10:30:56 CTIOS1-ctios Trace: [agent.5000.5012] QUERY_AGENT_STATE_REQ [CtiosBaseAgentObject,
     T_BASE_AGENT_QUERY_AGENT_STATE_REQ, 1501]
10:30:56 CTIOS1-ctios Trace: [agent.5000.5012] QUERY_AGENT_STATE_CONF ( eLogout ) [CtiosBaseAgentObject,
     T_BASE_AGENT_QUERY_AGENT_STATE_CONF, 1501]
Thanks!

Is this with the out of the box CTIOS desktop? Any customization?
david

Unable to transfer outside calls twice with CCME 4.0

Hello,
I have a 3825 with a CCME 4.0 and one E1. There is an IVR (from stonevoice) implemented.
When the operator receives the call, she is able to transfer the call to a internal extension (for example 230), but when the user (ext. 230) tries to transfer the call again to another internal extension (ext. 255), the outside caller is still listening the moh and the second destination extension (ext. 255) don't receives the call.
On the phone screen of the ext.230, remains the 2 calls on hold with a message like "transfer not valid or unable to transfer".
I have enabled the h450.2 and h450.3 services:
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
service script1 http://192.168.10.114/fw/Apps/IOSIVR/Server/TargetScript/script1.tcl
paramspace english index 1
param operator 99910
param msg_attesa_selection NONE
param alert_time 15
paramspace english language en
param msg_attesa_operator NONE
paramspace english location http://192.168.10.114/fw/Apps/IOSIVR/Server/AudioFiles/
param num_selection1 99906
param num_selection2 99907
param num_selection3 99908
param num_selection4 99909
param welcome_msg _menu_GC.au
param num_selection5 99910
paramspace english prefix en
service cua flash:app-b-acd-2.1.0.0.tcl
param queue-len 20
param aa-hunt1 99900
param queue-manager-debugs 1
param number-of-hunt-grps 2
param aa-hunt2 99901
service grupocastilla flash:app-b-acd-aa-2.1.0.0.tcl
paramspace english index 0
param drop-through-option 1
param second-greeting-time 30
param drop-through-prompt _dt_prompt.au
paramspace english language en
param max-time-vm-retry 2
param voice-mail 1991
param max-time-call-retry 700
param aa-pilot 99910
param number-of-hunt-grps 1
paramspace english location flash:
param handoff-string grupocastilla
param call-retry-timer 15
param service-name cua
dial-peer voice 104 voip
service grupocastilla
destination-pattern 99910
description queue loopback
session target ipv4:192.168.10.50
incoming called-number 99910
codec g711ulaw
telephony-service
load 7960-7940 P0030702T023
load 7914 S00104000100
load 7905 CP7905080001SCCP051117A
load 7920 cmterm_7920.4.0-02-00
load 7912 CP7912080001SCCP051117A
max-ephones 168
max-dn 500
ip source-address 192.168.100.1 port 2000
timeouts interdigit 5
system message CCME
cnf-file perphone
user-locale ES
network-locale ES
time-zone 26
time-format 24
date-format dd-mm-yy
voicemail 1999
max-conferences 24 gain -6
call-forward pattern .T
moh music-on-hold.au
web admin system name ird secret 5 $1$9raP$zImXvNHnQIUFoM.EfkeP31
time-webedit
transfer-system full-consult
transfer-pattern ...
transfer-pattern ....
transfer-pattern 1...
transfer-pattern 9...
transfer-pattern .T
after-hours block pattern 1 0906 7-24
after-hours block pattern 2 0905 7-24
after-hours block pattern 3 080 7-24
after-hours block pattern 4 000 7-24
create cnf-files version-stamp 7960 Sep 21 2006 18:07:37
ephone-hunt 1 longest-idle
pilot 99900
list 99980, 99981
timeout 14, 14
statistics collect
description Operator

Can you test the following and tell us what happens ?
a. Call 230 from an internal extension, transfer to internal extension B, and then do a second transfer from B to C.
b. Call 230 from outside, transfer to internal extension B , and then do a second transfer from B to C.
c. Call the operator from inside, transfer to 230, and then do a second transfer from 230 to 255.
d. Call the operator from outside, transfer to 230, then transfer to 255. (We know this fails, so this is just FYI, no need to test this).
HTH
Sankar.

Unity 7.0(2) cannot transfer from Call Handler to Non Sub Extn

Hi,
Got Unity 7 with Exchange onbox. Need to setup a Call Handler that will forward calls to another extensionon the CUCM that has no VM. I created 2 call handlers:
CH1 DN:1234 (same as CUCM DN)
Transfer incoming calls to a phone=No,send directly to Greeting
Greeting=Send caller to Call Handler CH2
CH2
Transfer incoming call to a phone=Yes. Extn 5678.
Problem is I keep getting "Sorry Example Administrator is not available, please leave a message".
Any ideas?
Rich

The call flow needs to be:
CH1 either via a caller input or an after greeting action
Attempt transfer to: CH2
In CH2 under call transfer settings, yes, ring subscriber at this DN
Your restriction tables need to allow what you're trying to dial.
HTH
java
If this helps, please rate
www.cisco.com/go/pdihelpdesk

Unable to perform call transfer or call park for an outbound call via SIP Trunk (SKYPE)

We have configured the SIP Trunk & SIP profile and successfull make outbound call through SIP Trunk (SKYPE). However, we are not able to perform call transfer or call park when the call is connected.
The scenario is:
A call to an phone number via SIP trunk, when call established, A perform call-transfer to B. After the call-transfer, the call Drop and Phone B show error code "Temp Fail"
When i select "enable MTP" in SIP trunk, we are able to call transfer and call park. But it limit the number of call session to 1.

You are probably running into some sort of Codec issue. IE, your phone is G.711 and the trunk is G.729. You will need to transcode the call at somepoint.

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

7940 phone resets when try to transfer a call

When I am engaged in a call on my 7940 and wish to transfer a call. I press the transfer softkey and the phone resets. If anyone can give me some ideas and or solutions it would be greatly appreciated.

Following guide line help you for transfer the call.
if user B has an active call on a particular line (from user A) and user B has not reached the maximum number of calls on this line, the Cisco Unified IP Phone provides a Transfer softkey to user B. If user B presses the Transfer softkey (or Transfer button, if available) once, user B receives dial tone and can make a secondary call: user B dials the number of a third-party (user C). Cisco Unified CallManager provides a Transfer softkey to user B again. If user B presses the Transfer softkey again (or Transfer button, if available), the transfer operation completes.
With the onhook call transfer implementation, user B can hang up after dialing the number of user C, and the transfer completes. Both the existing and new implementations work in the case of a blind transfer (user B disconnects before user C answers) and also in the case of a consult transfer (user B waits for user C to answer and announces the call from user A).
The previous implementation remains unchanged: user B can press the Transfer softkey twice to complete the transfer.

Can I transfer a call to another Verizon cell in the middle of a call?

Does anyone know if it is possible to transfer a call to another Verizon cell in the middle of a call? Not call forwarding, but transfer during the call?
So, I answer the phone and talk to the caller. Then, I want to transfer the caller to Joe on another cell phone. Is there any way to put the caller on hold and transfer the call to Joe's Verizon cell?
My understanding is that call forwarding only works for the initial phone call - the call forwards to another number, but it won't work in the middle of a call.
Any help is appreciated!
Thanks

All the older Nokia phones you answer a call – you can put it on hold – call someone else, tell him/her that you are transferring a call – press 4 and hang up. Those two can continue the conversation together.
That’s what I’m looking for.
With Lumia I don’t want to be the 3rd wheel in the conversation...
I haven't found the way to end the call so that the others could continue (not even with conference call)

Nokia E61 Transfer VoIP Call Bug

Hi! I have an Asterisk 1.4 VoIP Server and found two ways to transfer a call ... Unfortunately both ways are not working properly ... 1. Automatic transfer When you have an active call and choose Options -> Automatic transfer and then enter a number you get the "Waiting for acceptance to transfer call" message. The call is not transfered until you hang-up. Question: Is this normal or should the phone hang-up automatically? 2. New call and then transfer When you have an active call you can choose Options -> New call -> Internet call. After entering the number ... a. the Standby application crashes b. the call gets transfered and the Standby application crashes and displays a light bubble with the text "call1". In both cases I can only restart the phone by pressing the power on button. Does anyone know the solution to my problems? TIA, Mike

Hi! I have an Asterisk 1.4 VoIP Server and found two ways to transfer a call ... Unfortunately both ways are not working properly ... 1. Automatic transfer When you have an active call and choose Options -> Automatic transfer and then enter a number you get the "Waiting for acceptance to transfer call" message. The call is not transfered until you hang-up. Question: Is this normal or should the phone hang-up automatically? 2. New call and then transfer When you have an active call you can choose Options -> New call -> Internet call. After entering the number ... a. the Standby application crashes b. the call gets transfered and the Standby application crashes and displays a light bubble with the text "call1". In both cases I can only restart the phone by pressing the power on button. Does anyone know the solution to my problems? TIA, Mike

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

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 a call

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