How do I extend incoming call duration before voicemail activates ?

How do I extend incoming call duration before voicemail activates ?

I had this problem and for 12 months no one could tell me how to fix it.
Finally a young kid told me.  Get onto your carrier web site, in this case Telstra in Australia,
and you will search for ringing or similar and there should be instructions how to do it. It worked!!!
good luck

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    Hi amit,
    Thanks for the  reply.
    But i already mentioned  that my users cann't search using tables.
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  • CME\7960 running SIP firmware - How do i setup incoming calls? - Can anyone help please?

    Hi Guys,
    I have a SIP trunk setup with a 2811 running CME version 7.  I can make outbound calls ok but having issues getting the incoming calls working, i have 1 number on my SIP trunk and that is 01133501788 and i want that to ring my Cisco 7960 which is running SIP firmware not SCCP.  I have included by config for anyone who can help me, i just want the incoming call to work. 
    Many Thanks.
    Matthew.
    version 12.4
    service timestamps debug datetime msec
    service timestamps log datetime msec
    no service password-encryption
    hostname Router
    boot-start-marker
    boot-end-marker
    logging message-counter syslog
    no aaa new-model
    clock timezone GMT 0
    dot11 syslog
    ip source-route
    ip cef
    no ip dhcp use vrf connected
    ip dhcp excluded-address 192.168.1.1
    ip dhcp excluded-address 10.10.10.1
    ip dhcp pool DATA_POOL
       network 10.10.10.0 255.255.255.0
       default-router 10.10.10.1
       dns-server 188.92.232.50 188.92.232.100
    ip dhcp pool VOICE_POOL
       network 192.168.1.0 255.255.255.0
       default-router 192.168.1.1
       dns-server 188.92.232.50 188.92.232.100
       option 150 ip 192.168.1.1
    ip name-server 188.92.232.50
    ip name-server 188.92.232.100
    no ipv6 cef
    multilink bundle-name authenticated
    voice service voip
    allow-connections h323 to h323
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    no supplementary-service sip moved-temporarily
    no supplementary-service sip refer
    sip
      bind control source-interface FastEthernet0/1.20
      bind media source-interface FastEthernet0/1.20
      registrar server
    voice class codec 1
    codec preference 2 g711ulaw
    codec preference 3 g711alaw
    voice register global
    mode cme
    source-address 192.168.1.1 port 5060
    max-dn 144
    max-pool 42
    load 7960-7940 P0S3-8-12-00
    authenticate register
    tftp-path flash:
    create profile sync 0008072514198272
    voice register dn  1
    number 6999
    allow watch
    name SIP
    label SIP
    voice register pool  1
    id mac 000F.902B.40E0
    type 7960
    number 1 dn 1
    dtmf-relay sip-notify
    username cisco password cisco
    codec g711ulaw
    voice translation-rule 1
    rule 1 /^9\(.*\)/ /\1/
    voice translation-rule 2
    rule 1 /^6...$/ /4143*002/
    voice translation-profile DiscardDigit9
    translate calling 2
    translate called 1
    voice translation-profile IncomingSIP
    translate calling 1133501788
    voice-card 0
    no dspfarm
    username matt privilege 15 secret 5 $1$DCD0$SjWqnKgDSGVzzIKRerXh11
    archive
    log config
      hidekeys
    interface FastEthernet0/0
    ip address 194.12.0.222 255.255.255.252
    ip nat outside
    ip virtual-reassembly
    duplex auto
    speed auto
    interface FastEthernet0/1
    no ip address
    ip nat inside
    ip virtual-reassembly
    duplex auto
    speed auto
    interface FastEthernet0/1.10
    description DATA
    encapsulation dot1Q 10
    ip address 10.10.10.1 255.255.255.0
    ip nat inside
    ip virtual-reassembly
    interface FastEthernet0/1.20
    description VOICE
    encapsulation dot1Q 20
    ip address 192.168.1.1 255.255.255.0
    ip nat inside
    ip virtual-reassembly
    ip forward-protocol nd
    ip route 0.0.0.0 0.0.0.0 194.12.0.221
    ip http server
    ip http authentication local
    no ip http secure-server
    ip nat inside source list 1 interface FastEthernet0/0 overload
    access-list 1 permit 192.168.1.0 0.0.0.255
    access-list 1 permit 10.10.10.0 0.0.0.255
    tftp-server flash:P003-8-12-00.bin
    tftp-server flash:P003-8-12-00.sbn
    tftp-server flash:P0S3-8-12-00.loads
    tftp-server flash:P0S3-8-12-00.sb2
    tftp-server flash:P003-8-12-00
    tftp-server flash:P003-8-12-00.loads
    tftp-server flash:P003-8-12-00.sb2
    tftp-server flash:SIP000F902B40E0.cnf.xml
    control-plane
    mgcp behavior g729-variants static-pt
    dial-peer cor custom
    dial-peer voice 2 voip
    description Outgoing Geographic
    translation-profile outgoing DiscardDigit9
    destination-pattern 0[7]........
    voice-class codec 1
    session protocol sipv2
    session target dns:sip.cloudcalling.co.uk
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 1 voip
    description IncomingSIP
    translation-profile incoming IncomingSIP
    voice-class codec 1
    session protocol sipv2
    session target dns:sip.cloudcalling.co.uk
    incoming called-number .T
    dtmf-relay sip-notify rtp-nte
    no vad
    sip-ua
    credentials username 4143*002 password 7 password realm sip.cloudcalling.co.uk
    authentication username 4143*002 password 7 password
    nat symmetric role passive
    nat symmetric check-media-src
    calling-info sip-to-pstn number set 4143*002
    no remote-party-id
    retry invite 3
    retry register 3
    timers connect 100
    registrar dns:sip.cloudcalling.co.uk expires 60
    sip-server dns:sip.cloudcalling.co.uk
      host-registrar
    gatekeeper
    shutdown
    telephony-service
    load 7960-7940 P0S3-8-12-00
    max-ephones 24
    max-dn 30
    ip source-address 192.168.1.1 port 2000
    max-conferences 8 gain -6
    web admin system name Admin secret 5 $1$Fktw$t9GQkdDdHmoYdwptO8.or.
    transfer-system full-consult
    create cnf-files version-stamp Jan 01 2002 00:00:00
    line con 0
    line aux 0
    line vty 0 4
    login
    scheduler allocate 20000 1000
    ntp server 85.119.80.232
    end
    Router#

    You my friend are a star! worked straight away, many thanks.  Just one more thing, when i make an outgoing call, it always appears as "blocked" on my phone, my sip trunk is set to allow CME to alter outgoing CLI's how would i program the outgoing CLI to 01133501788 also?
    The new working config is below with your suggestion, which works!
    version 12.4
    service timestamps debug datetime msec
    service timestamps log datetime msec
    no service password-encryption
    hostname Router
    boot-start-marker
    boot-end-marker
    logging message-counter syslog
    no aaa new-model
    clock timezone GMT 0
    dot11 syslog
    ip source-route
    ip cef
    no ip dhcp use vrf connected
    ip dhcp excluded-address 192.168.1.1
    ip dhcp excluded-address 10.10.10.1
    ip dhcp pool DATA_POOL
       network 10.10.10.0 255.255.255.0
       default-router 10.10.10.1
       dns-server 188.92.232.50 188.92.232.100
    ip dhcp pool VOICE_POOL
       network 192.168.1.0 255.255.255.0
       default-router 192.168.1.1
       dns-server 188.92.232.50 188.92.232.100
       option 150 ip 192.168.1.1
    ip name-server 188.92.232.50
    ip name-server 188.92.232.100
    no ipv6 cef
    multilink bundle-name authenticated
    voice service voip
    allow-connections h323 to h323
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    no supplementary-service sip moved-temporarily
    no supplementary-service sip refer
    sip
      registrar server
    voice class codec 1
    codec preference 2 g711ulaw
    codec preference 3 g711alaw
    voice register global
    mode cme
    source-address 192.168.1.1 port 5060
    max-dn 144
    max-pool 42
    load 7960-7940 P0S3-8-12-00
    authenticate register
    tftp-path flash:
    create profile sync 0015244443466064
    voice register dn  1
    number 6999
    allow watch
    name SIP
    label SIP
    voice register pool  1
    id mac 000F.902B.40E0
    type 7960
    number 1 dn 1
    dtmf-relay sip-notify
    username cisco password cisco
    codec g711ulaw
    voice translation-rule 1
    rule 1 /^6...$/ /4143*002/
    voice translation-rule 3
    rule 1 /^01133501788$/ /6999/
    rule 2 /^1133501788$/ /6999/
    voice translation-profile IncomingSIP
    translate called 3
    voice translation-profile Translatetrunk
    translate calling 1
    voice-card 0
    no dspfarm
    username matt privilege 15 secret 5 $1$DCD0$SjWqnKgDSGVzzIKRerXh11
    archive
    log config
      hidekeys
    interface FastEthernet0/0
    ip address 194.12.0.222 255.255.255.252
    ip nat outside
    ip virtual-reassembly
    duplex auto
    speed auto
    interface FastEthernet0/1
    no ip address
    ip nat inside
    ip virtual-reassembly
    duplex auto
    speed auto
    interface FastEthernet0/1.10
    description DATA
    encapsulation dot1Q 10
    ip address 10.10.10.1 255.255.255.0
    ip nat inside
    ip virtual-reassembly
    interface FastEthernet0/1.20
    description VOICE
    encapsulation dot1Q 20
    ip address 192.168.1.1 255.255.255.0
    ip nat inside
    ip virtual-reassembly
    ip forward-protocol nd
    ip route 0.0.0.0 0.0.0.0 194.12.0.221
    ip http server
    ip http authentication local
    no ip http secure-server
    ip nat inside source list 1 interface FastEthernet0/0 overload
    access-list 1 permit 192.168.1.0 0.0.0.255
    access-list 1 permit 10.10.10.0 0.0.0.255
    tftp-server flash:P003-8-12-00.bin
    tftp-server flash:P003-8-12-00.sbn
    tftp-server flash:P0S3-8-12-00.loads
    tftp-server flash:P0S3-8-12-00.sb2
    tftp-server flash:P003-8-12-00
    tftp-server flash:P003-8-12-00.loads
    tftp-server flash:P003-8-12-00.sb2
    tftp-server flash:SIP000F902B40E0.cnf.xml
    control-plane
    mgcp behavior g729-variants static-pt
    dial-peer cor custom
    dial-peer voice 1 voip
    description IncomingSIP
    translation-profile incoming IncomingSIP
    voice-class codec 1
    session protocol sipv2
    session target sip-server
    incoming called-number .T
    dtmf-relay sip-notify rtp-nte
    no vad
    dial-peer voice 2 voip
    description Outgoing Geographic
    translation-profile outgoing Translatetrunk
    destination-pattern 0[7]........
    voice-class codec 1
    session protocol sipv2
    session target dns:sip.cloudcalling.co.uk
    dtmf-relay rtp-nte
    no vad
    sip-ua
    credentials username 4143*002 password 7 password realm sip.cloudcalling.co.uk
    authentication username 4143*002 password 7 password
    nat symmetric role passive
    nat symmetric check-media-src
    calling-info sip-to-pstn number set 4143*002
    no remote-party-id
    retry invite 3
    retry register 3
    timers connect 100
    registrar dns:sip.cloudcalling.co.uk expires 60
    sip-server dns:sip.cloudcalling.co.uk
      host-registrar
    gatekeeper
    shutdown
    telephony-service
    load 7960-7940 P0S3-8-12-00
    max-ephones 24
    max-dn 30
    ip source-address 192.168.1.1 port 2000
    max-conferences 8 gain -6
    web admin system name Admin secret 5 $1$Fktw$t9GQkdDdHmoYdwptO8.or.
    transfer-system full-consult
    create cnf-files version-stamp 7960 Dec 17 2013 14:35:13
    line con 0
    line aux 0
    line vty 0 4
    login
    scheduler allocate 20000 1000
    ntp server 85.119.80.232
    end
    Router#

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