Prefixing a 9 and 91 to incoming calls from SIP provider for callback

I am wondering what would be the best options for prefixing a 9 or 91 to incoming calls over a sip connection to allow callback from missed calls and recieved calls. The setup is
callmanager 7.1.5 >>>>sip trunk>>>>>>>>>>>>CUBE>>>>>>>>>>>sip to ITSP
I am thinking voice translation rules is the only option for this? any configuration examples for this would be greatly appreciated.
would this work?
voice-translation rule 1
rule 1 // /9/
voice-translation profile prefix_9
translate calling 1
dial-peer voice 101 voip
destination-pattern ???????...$
voice-class codec 1
session protocol sipv2
session target ipv4: to callmanager
incoming called-number .
dtmf-relay rtp-nte
dial-peer voice 1001 voip
translation profile incoming prefix_9
destination-pattern T
session protocol sipv2
session target ipv4: to sip provider
incoming called-number ???????...$
dtmf-relay rtp-nte

Your config should work fine, except your profile is only applied to one dial-peer, make sure you apply it to the one that is used to redirect the call to CUCM.
Also, you did not mention what country you are in, but if this is US you may want to prefix 91 to national calls as carriers don't provide 9 as part of the CLID delivery, also what about your international calls, you may would be more explicit in your first rule to match for national digit string and then have another rule for international.
HTH,
Chris

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Dear friends,
We are planning to setup an out of band management connectivity, in our test setup we have a Cisco 2811 ISR (with IOS 12.4(9)T5 ) with NM-8AM (Analog modem module) installed.
Test Setup:
1. One Analog line (PSTN) connected to PC via built-in modem
2. Second Analog line (PSTN) connected to Cisco 2811 (NM-8AM) terminated on the analog modem module
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version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Dial-Up-CPE
boot-start-marker
boot-end-marker
logging buffered 16384 debugging
enable password cisco
ip cef
voice-card 0
no dspfarm
username cisco password 0 cisco
interface Loopback0
ip address 10.10.10.1 255.255.255.255
interface FastEthernet0/0
ip address 20.20.20.1 255.255.255.0
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speed auto
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shutdown
duplex auto
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dialer in-band
dialer idle-timeout 600
dialer-group 1
async mode interactive
peer default ip address pool DIALIN
ppp authentication pap chap
group-range 1/0 1/7
ip local pool DIALIN 172.16.16.1 172.16.16.5
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control-plane
line con 0
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modem InOut
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flowcontrol hardware
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stopbits 1
flowcontrol hardware
line vty 0 4
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Any links/references would be of great help.
Thank you
Pradeep

Hi,
I guess I'm having exactly the same setup. The integrated modem on the other end does pick up the call but soon disconnect after displaying "Waiting for carrier".
Any clue?
Thanks,
Eyad

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Config is attached.
Model: SPA122, LAN, 2 FXS
Hardware Version: 1.0.0 Boot Version: 1.0.1 (Oct 6 2011 - 20:04:00)
Firmware Version: 1.3.2-XU (014) Jul 2 2013
Recovery Firmware: 1.0.2 (001)
WAN MAC Address: 6C:20:56:55:3A:B6
Host Name: SPA122
Domain Name: (none)
Serial Number: CCQ16450LG3
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[2] is misconfiguration on your's side. You have CPC turned on, but no CPC capable device. Set CPC Duration to zero to turn off CPC.
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Hi,
we have a 2610XM router with CCME 3.2 and 4 bri
and we don't redirect incomming call from pstn on a internal IP Phone.
which command for redirect call ?
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debug isdn q931:
Mar 22 12:53:46.350: ISDN BR1/0 Q931: Ux_DLRelInd: DL_REL_IND received from L2
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Bearer Capability i = 0x8090A3
Standard = CCITT
Transer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
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Calling Party Number i = 0x2083, '685284832'
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Plan:ISDN, Type:Unknown
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Router#
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http://www.cisco.com/en/US/products/sw/iosswrel/ps5012/products_feature_guide_chapter09186a00801812db.html

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Please rate helpful posts.
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Here is the inof for the Call Manager Express.
Enabling Call Pickup
To enable Call Pickup features on SCCP or SIP phones, perform the following steps.
Prerequisites
•SIP phones require Cisco Unified CME 7.1 or a later version.
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Restrictions
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•A directory number can be assigned to only one pickup group.
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SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. service directed-pickup [gpickup]
5. fac {standard | custom pickup {direct | group | local} custom-fac}
6. exit
7. ephone-dn dn-tag [dual-line | octo-line]
or
voice register dn dn-tag
8. pickup-group group-number
9. pickup-call any-group
10. end
DETAILED STEPS
Command or Action
Purpose
Step 1
enable
Example:
Router> enable
Enables privileged EXEC mode.
•Enter your password if prompted.
Step 2
configure terminal
Example:
Router# configure terminal
Enters global configuration mode.
Step 3
telephony-service
Example:
Router(config)# telephony-service
Enters telephony-service configuration mode.
Step 4
service directed-pickup [gpickup]
Example:
Router(config-telephony)# service directed-pickup gpickup
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•gpickup—(Optional) Enables using the GPickUp soft key to perform Directed Call Pickup on SCCP phones. This keyword is supported in Cisco Unified CME 7.1 and later versions.
•This command determines the specific soft keys used to access different Call Pickup features on SCCP and SIP phones. For a description, see the service directed-pickup command in the Cisco Unified CME Command Reference.
Step 5
fac {standard | custom pickup {direct | group | local} custom-fac}
Example:
Router(config-telephony)# fac custom pickup group #35
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•custom—Creates a custom FAC for a feature.
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Step 6
exit
Example:
Router(config-telephony)# exit
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Step 7
ephone-dn dn-tag [dual-line | octo-line]
or
voice register dn dn-tag
Example:
Router(config)# ephone-dn 20 dual-line
or
Router(config)# voice register dn 20
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Step 8
pickup-group group-number
Example:
Router(config-ephone-dn)# pickup-group 30
or
Router(config-register-dn)# pickup-group 30
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Step 9
pickup-call any-group
Example:
Router(config-ephone-dn)# pickup-call any-group
or
Router(config-register-dn)# pickup-call any-group
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Step 10
end
Example:
Router(config-ephone-dn)# end
or
Router(config-register-dn)# end
Exits configuration mode.
Examples
The following example shows the Group Pickup and Local Group Pickup features enabled with the service directed-pickup gpickup command. Extension 1005 on phone 5 and extension 1006 on phone 6 are assigned to pickup group 1.
telephony-service
load 7960-7940 P00308000500
load E61 SCCP61.8-2-2SR2S
max-ephones 100
max-dn 240
ip source-address 15.7.0.1 port 2000
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voicemail 8900
max-conferences 8 gain -6
call-park system application
transfer-system full-consult
fac standard
create cnf-files version-stamp 7960 Sep 25 2007 21:25:47
ephone-dn 5
number 1005
pickup-group 1
ephone-dn 6
number 1006
pickup-group 1
ephone 5
mac-address 0001.2345.6789
type 7962
button 1:5
ephone 6
mac-address 000F.F758.E70E
type 7962
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HT201229 How can I access a list of incoming calls from a blocked number

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Hello,friends!
I have already published a discussion here https://supportforums.cisco.com/discussion/12089656/cisco-cme-and-calls-through-sip-provider and you helped me, everything works well for Russian numbers.
When I tried to add the configuration for calls to Belarus, again, there was a problem. I do not understand why, although the configuration ideintichnaya.
My config:
voice service voip
ip address trusted list
ipv4 178.16.26.122 255.255.255.255
ipv4 144.76.42.108 255.255.255.255
ipv4 176.9.145.115 255.255.255.255
ipv4 5.9.108.25 255.255.255.255
ipv4 78.46.95.118 255.255.255.255
ipv4 89.249.23.194 255.255.255.255
ipv4 178.16.26.124 255.255.255.255
ipv4 176.9.85.133 255.255.255.255
ipv4 46.4.53.86 255.255.255.255
ipv4 5.9.84.165 255.255.255.255
ipv4 78.16.26.122 255.255.255.255
ipv4 77.235.62.222 255.255.255.255
ipv4 81.88.86.11 255.255.255.255
ipv4 192.168.1.50 255.255.255.255
ipv4 217.150.198.44 255.255.255.255
ipv4 178.63.96.3 255.255.255.255
ipv4 178.63.96.28 255.255.255.255
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
sip
registrar server
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
codec preference 3 g711alaw
voice class sip-profiles 20
request INVITE sip-header From modify "\"(.*)\" <sip:(.*)@(.*)>" "\"\" <sip:[email protected]>"
voice translation-rule 9
rule 1 /^98/ /7/
voice translation-rule 10
rule 1 /^9/ //
voice translation-rule 1020
rule 1 /^.*$/ /141756/
voice translation-rule 1030
rule 1 /^.*/ /141756/
voice translation-rule 1040
rule 1 /^.*$/ /21/
voice translation-profile incoming
translate called 1040
voice translation-profile outgoing
translate calling 1030
translate called 9
voice translation-profile outgoing-mezhdunarod
translate calling 1030
translate called 10
voice-card 0
dial-peer voice 2 voip
description TO-RUSSIA
translation-profile outgoing outgoing
preference 1
destination-pattern 98..........
session protocol sipv2
session target sip-server
voice-class codec 1
no voice-class sip outbound-proxy
voice-class sip profiles 20
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay rtp-nte sip-notify
no vad
dial-peer voice 3 voip
translation-profile incoming incoming
incoming called-number 141756
voice-class codec 1
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay rtp-nte
no vad
dial-peer voice 4 voip
description To-Belarus
translation-profile outgoing outgoing-mezhdunarod
destination-pattern 9375.........
session protocol sipv2
session target sip-server
voice-class codec 1
no voice-class sip outbound-proxy
voice-class sip profiles 20
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay rtp-nte sip-notify
no vad
sip-ua
credentials username 141756 password 7<pass> realm sip.zadarma.com
authentication username 141756 password 7 <pass>
no remote-party-id
registrar 1 dns:sip.zadarma.com expires 3600
sip-server dns:sip.zadarma.com
connection-reuse
host-registrar
DEBUG ccsip message:
Jun 17 14:23:09.033: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK36571F65
From: "" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>
Date: Tue, 17 Jun 2014 09:23:09 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3536040247-4114026979-2924673736-0741251102
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1402996989
Contact: <sip:[email protected]:5060>
Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 309
v=0
o=CiscoSystemsSIP-GW-UserAgent 6656 8059 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18252 RTP/AVP 0 18 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Jun 17 14:23:09.089: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK36571F65;rport=5060
From: "" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.6d40
Call-ID: [email protected]
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="U6AYAFOgFtT86kmu2Fr5tYxLYGEexIl1", qop="auth"
Server: kamailio (4.1.2 (x86_64/linux))
Content-Length: 0
Jun 17 14:23:09.169: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK36571F65
From: "Vankuver" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.6d40
Date: Tue, 17 Jun 2014 09:23:09 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Jun 17 14:23:09.169: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK365820F2
From: "" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>
Date: Tue, 17 Jun 2014 09:23:09 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3536040247-4114026979-2924673736-0741251102
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1402996989
Contact: <sip:[email protected]:5060>
Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="141756",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="9534322838cbf2e265b2004bc0aa240e",nonce="U6AYAFOgFtT86kmu2Fr5tYxLYGEexIl1",cnonce="FFF9A231",qop=auth,algorithm=md5,nc=00000001
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 309
v=0
o=CiscoSystemsSIP-GW-UserAgent 6656 8059 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18252 RTP/AVP 0 18 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Jun 17 14:23:09.637: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK365820F2
From: "" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>
Date: Tue, 17 Jun 2014 09:23:09 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3536040247-4114026979-2924673736-0741251102
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1402996989
Contact: <sip:[email protected]:5060>
Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="141756",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="9534322838cbf2e265b2004bc0aa240e",nonce="U6AYAFOgFtT86kmu2Fr5tYxLYGEexIl1",cnonce="FFF9A231",qop=auth,algorithm=md5,nc=00000001
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 309
v=0
o=CiscoSystemsSIP-GW-UserAgent 6656 8059 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18252 RTP/AVP 0 18 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Jun 17 14:23:10.621: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK365820F2
From: "" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>
Date: Tue, 17 Jun 2014 09:23:10 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3536040247-4114026979-2924673736-0741251102
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1402996990
Contact: <sip:[email protected]:5060>
Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="141756",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="9534322838cbf2e265b2004bc0aa240e",nonce="U6AYAFOgFtT86kmu2Fr5tYxLYGEexIl1",cnonce="FFF9A
All possible debugging has been turned off
DC#231",qop=auth,algorithm=md5,nc=00000001
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 309
v=0
o=CiscoSystemsSIP-GW-UserAgent 6656 8059 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18252 RTP/AVP 0 18 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Debug voice ccapi inout:
Destination Pattern=9375........., Called Number=375298911396, Digit Strip=FALSE
Jun 17 15:22:13.073: //14425/13366763AF35/CCAPI/ccCallSetupRequest:
Calling Number=141756(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=375298911396(TON=Unknown, NPI=Unknown),
Redirect Number=, Display Info=Vankuver
Account Number=, Final Destination Flag=FALSE,
Guid=13366763-F540-11E3-AF35-FAC82C2E981E, Outgoing Dial-peer=4
Jun 17 15:22:13.073: //14425/13366763AF35/CCAPI/cc_api_display_ie_subfields:
ccCallSetupRequest:
cisco-username=
----- ccCallInfo IE subfields -----
cisco-ani=141756
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=0
dest=375298911396
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=0
cisco-rdnsi=0
cisco-redirectreason=0 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
Jun 17 15:22:13.073: //14425/13366763AF35/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x6968AA04, Interface Type=3, Destination=, Mode=0x0,
Call Params(Calling Number=141756,(Calling Name=Vankuver)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=375298911396(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=RegularLine, FinalDestinationFlag=FALSE, Outgoing Dial-peer=4, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
Jun 17 15:22:13.073: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jun 17 15:22:13.073: :cc_get_feature_vsa malloc success
Jun 17 15:22:13.073: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jun 17 15:22:13.077: cc_get_feature_vsa count is 2
Jun 17 15:22:13.077: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jun 17 15:22:13.077: :FEATURE_VSA attributes are: feature_name:0,feature_time:1819298856,feature_id:3371
Jun 17 15:22:13.077: //14427/13366763AF35/CCAPI/ccIFCallSetupRequestPrivate:
SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
Jun 17 15:22:13.077: //14427/13366763AF35/CCAPI/ccCallSetContext:
Context=0x6C726BF4
Jun 17 15:22:13.077: //14425/13366763AF35/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=4
Jun 17 15:22:13.085: //14427/13366763AF35/CCAPI/cc_api_call_proceeding:
Please help me... I don't know what to do!

You need to contact service provider for this , after authentication challenge your sip provider is not sending any response.
Contact them and ask whether they had received INVITE with proxy authentication details or not.

Help please...iphone5 when trying to make a call, after hitting green button to make call, screen goes straight back to keypad and all incoming calls go straight to voicemail, any ideas please? Everything else seems ok. No problem with sim or carrier

Hi...I have just obtained a secondhand iphone 5. I'm not technically minded and I don't understand 'geek speak' so please be gentle. I have done all I can think of to bottom this problem. I have switched on and off. Tried different o2 sims (including 2 brand new) that I know are fine, restored to factory settings, changed carrier from 'Auto' to O2 and spent ages on line with o2 chat but everything at their end is absolutely fine. Everything on the phone seems to be working apart from the phone aspect of things. All incoming calls go straight to voicemail and show on the phone as 'Cancelled calls' and when dialling out, I can key in the numbers and get the keypad tone (not dialling tone) but as soon as I press the green key to make the call, all that happens is the keypad pops back up onto the screen. The phone was bought on ebay as an o2 phone (the photo showed Giffgaff on the phone's screen but I understand this to be one and the same?) and as being 6 months old but I have put the serial number into the support website and...surprise surprise..,it must be at least twice that age as the warranty has expired. I have made 2 attempts to have a web chat with Apple Support, have reached the 'You're all set, we'll be with you in 2 mins' and have been left hanging there for over an hour on each occasion, so I've given that up as a bad job!! Any help will be much appreciated...and also, will Apple charge me £25 for each of the aborted chat attempts, please? Over to you...hopefully and many thanks in anticipation.

My phone works all fine when am in mumbai but as soon s i leave mumbai I am not able to make an otgoing call
This is entirely a carrier issue. If your carrier (airtel) doesn't provide service outside of Mumbia, this has nothing to do with your iPhone.
i feel i need to change my brand
This is a user-to-user tech support forum, not Apple. No one here cares at all about your threats.
NEVER post personal info in this public forum

CUPC Incoming Call from Hunt Group

Hi all!
Is is possible that I can see whenever someone calls me on my extension, or whenever a call comes in from a hunt group? Can I differ the incoming calls?
Regards
Rene

Are you talking about the attendant console hunt group? If yes, then you can do this by using the broadcast hunting feature. What it does is, all calls to the hunt pilot will be placed in queue and will be displayed in the Broadcast calls window within the Attendant Console. All uses can see the calling number of calls in queue and pick and chose the calls they want to answer.
Check this link for more details
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/4_1/sys_ad/4_1_3/ccmfeat/fsccmac.htm#wp1144539
Regards,
Anup

Prefixing a 9 and 91 to incoming calls from SIP provider for callback

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