09-29-2015 11:41 AM - last edited on 03-25-2019 08:36 PM by ciscomoderator
Hello,
Question 1) I have a SIP phone and am trying to pass my dialed digits to Unity Express Voicemail, it does not work. Is it the codec or dtmf-relay option?
Question 2) how come my Cisco 7960 sip phone takes 10 seconds to dial out?
Dial-peers are:
Voicemail number:
dial-peer voice 6000 voip
destination-pattern 69..
session protocol sipv2
session target ipv4:10.16.0.2
dtmf-relay sip-notify
codec g711ulaw
no vad
SIP Trunk, this DTMF works
dial-peer voice 2 voip
description **Outgoing calls to SIP.US SIP Trunk**
translation-profile outgoing SIP.US-Outgoing
destination-pattern .T
session protocol sipv2
session target dns:gw1.sip.us
voice-class sip profiles 1
dtmf-relay rtp-nte
codec g711ulaw
no vad
Solved! Go to Solution.
09-30-2015 03:13 AM
Hi,
Any reason why you are putting CUE and CME in different subnets?
You are missing bind all command. It can be applied globally or on dialpeers
Regarding DTMF problem, can you share the output of debug ccsip mess when you press DTMF digits. Also, can you share the output of show call active voice called XXXX. I need to see what is getting negotiated for this call (your config is showing SIP Notify). Is DTMF relay working for SCCP phones?
Regarding 10 secs delay on 7960 phones, this is because 7960 phones are TypeA phones and perform local digit analysis before sending the call to CME. You can overcome this problem by configuring SIP dial rules. For example:
voice register dialplan 1
type 7940-7960-others
pattern 1 #dialed-pattern# timeout 0
!
voice register pool 1
type 7960
dialplan 1
09-29-2015 01:10 PM
Hi Could you please elaborate a bit more about the topology?
Is it something like this:
SIP ITSP <====> CME / CUBE <=======> CUE?
Regards
09-29-2015 02:17 PM
Hi,
Thanks for the response. I am having trouble getting a voice mail setup with a SIP to CME to UNITY configuration. I've read that Unity Express doesn't work with SIP phones....I've heard conflicting things about this though.
Thus far to recap:
I've attached a Pdf of the configuration.
SIP#sh run
Building configuration...
Current configuration : 9642 bytes
!
! Last configuration change at 19:36:38 UTC Tue Sep 29 2015
! NVRAM config last updated at 19:36:40 UTC Tue Sep 29 2015
! NVRAM config last updated at 19:36:40 UTC Tue Sep 29 2015
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname SIP
!
boot-start-marker
boot-end-marker
!
!
! card type command needed for slot/vwic-slot 0/2
enable password XXXXXXXXXX
!
no aaa new-model
!
memory-size iomem 5
!
dot11 syslog
ip source-route
!
!
ip cef
!
ip dhcp excluded-address 10.15.0.0 10.15.0.20
ip dhcp excluded-address 10.25.0.0 10.25.0.20
!
ip dhcp pool SIP_VOICE
network 10.15.0.0 255.255.255.0
option 150 ip 10.15.0.1
default-router 10.15.0.1
option 66 ip 10.15.0.1
dns-server 8.8.8.8
!
ip dhcp pool 10_25
network 10.25.0.0 255.255.255.0
option 150 ip 10.25.0.1
default-router 10.25.0.1
option 66 ip 10.25.0.1
dns-server 8.8.8.8
!
!
ip name-server 8.8.8.8
ip multicast-routing
no ipv6 cef
!
multilink bundle-name authenticated
!
!
!
!
!
!
!
voice service voip
ip address trusted list
ipv4 65.254.44.194
ipv4 74.81.71.18
ipv4 192.168.XXXX
ipv4 67.176.XXXX
ipv4 67.176.XXXX
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 expires max 3600 min 120
!
voice class sip-profiles 1
request INVITE sip-header From modify "<sip:522795XXXX@67.176.XXXX>" "<sip:522795XXXX@gw1.sip.us>"
request INVITE sip-header From modify "<sip:522795XXXX@10.15.0.1>" "<sip:522795XXXX@gw1.sip.us>"
!
!
voice register global
mode cme
source-address 10.15.0.1 port 5060
timeouts interdigit 5
max-dn 40
max-pool 42
load 7960-7940 POS3-8-12-00.loads
timezone 8
time-format 24
tftp-path flash:
file text
create profile sync 00007930220XXXXXX
camera
video
!
voice register dn 1
number 5069
allow watch
name Joes SIP
!
voice register pool 1
id mac 001F.6C80.FDE9
type 7960
number 1 dn 1
dtmf-relay sip-notify
codec g711ulaw
no vad
camera
video
!
!
!
voice translation-rule 5
rule 1 /^.*/ /522795XXXXX/
!
voice translation-rule 99
!
!
voice translation-profile SIP.US-Outgoing
translate calling 5
!
!
voice-card 0
!
crypto pki token default removal timeout 0
!
!
!
!
license udi pid CISCO2811 sn FTX1151A3Q1
username XXXXX privilege 15 password 0 XXXXXXXXXXXXXX
!
redundancy
!
!
!
!
!
!
!
!
!
!
interface FastEthernet0/0
no ip address
duplex auto
speed auto
!
interface FastEthernet0/0.15
description SIP Phones
encapsulation dot1Q 15
ip address 10.15.0.1 255.255.255.0
ip nat inside
ip virtual-reassembly in
!
interface FastEthernet0/0.16
encapsulation dot1Q 16
ip address 10.16.0.1 255.255.255.0
!
interface FastEthernet0/0.25
description SCCP and PC
encapsulation dot1Q 25
ip address 10.25.0.1 255.255.255.0
ip nat inside
ip virtual-reassembly in
!
interface Service-Engine0/0
ip unnumbered FastEthernet0/0.16
service-module ip address 10.16.0.2 255.255.255.0
service-module ip default-gateway 10.16.0.1
!
interface FastEthernet0/1
description SIP Trunk Outside ip address 192.168.1.129
ip address dhcp
ip nat outside
ip virtual-reassembly in
duplex auto
speed auto
!
interface Serial0/0/0
no ip address
shutdown
no fair-queue
clock rate 2000000
!
interface Serial0/0/1
no ip address
shutdown
clock rate 2000000
!
ip forward-protocol nd
ip http server
ip http authentication local
no ip http secure-server
ip http path flash:CME_8.6.0_GUI
!
!
ip nat pool ovrld 67.176.XXXX 67.176.XXXX netmask 255.255.255.0
ip nat inside source list 7 pool ovrld overload
ip nat inside source static udp 10.15.0.1 5060 67.176.XXXX 5060 extendable
ip nat inside source static 10.15.0.1 67.176.XXXX route-map SIP_NAT
ip route 0.0.0.0 0.0.0.0 FastEthernet0/1 67.176.XXXX
ip route 10.16.0.2 255.255.255.255 Service-Engine0/0
!
ip access-list extended UDP_RTP
permit udp any any range 8000 20000
permit ip any any
!
access-list 7 permit 0.0.0.1 255.255.255.0
access-list 7 permit 10.15.0.0 0.0.0.255
access-list 7 permit 10.25.0.0 0.0.0.255
access-list 10 permit any
!
!
!
!
route-map SIP_NAT permit 10
match ip address UDP_RTP
!
!
tftp-server flash:/P0S3-8-12-00/OS79XX.TXT alias OS79XX.TXT
tftp-server flash:/P0S3-8-12-00/P003-8-12-00.bin alias P003-8-12-00.bin
tftp-server flash:/P0S3-8-12-00/P003-8-12-00.sbn alias P003-8-12-00.sbn
tftp-server flash:cmterm-7940-7960-8.12.00-sip.cop.sgn
tftp-server flash:/P0S3-8-12-00/P0S3-8-12-00.loads alias P0S3-8-12-00.loads
tftp-server flash:/cmterm-7940-7960-sccp.8-1-2/P00308010200.bin alias P00308010200.bin
tftp-server flash:/cmterm-7940-7960-sccp.8-1-2/P00308010200.loads alias P00308010200.loads
tftp-server flash:/cmterm-7940-7960-sccp.8-1-2/P00308010200.sb2 alias P00308010200.sb2
tftp-server flash:/cmterm-7940-7960-sccp.8-1-2/P00308010200.sbn alias P00308010200.sbn
tftp-server flash:/ringtones/RingList.xml alias RingList.xml
tftp-server flash:/ringtones/AreYouThere.raw alias AreYouThere.raw
tftp-server flash:/ringtones/Classic1.raw alias Classic1.raw
tftp-server flash:/ringtones/Drums1.raw alias Drums1.raw
tftp-server flash:/ringtones/KotoEffect.raw alias KotoEffect.raw
tftp-server flash:/ringtones/Analog1.raw alias Analog1.raw
tftp-server flash:/ringtones/Analog2.raw alias Analog2.raw
tftp-server flash:/ringtones/AreYouThereF.raw alias AreYouThereF.raw
tftp-server flash:/ringtones/Bass.raw alias Bass.raw
tftp-server flash:/ringtones/CallBack.raw alias CallBack.raw
tftp-server flash:/ringtones/Chime.raw alias Chime.raw
tftp-server flash:/ringtones/Classic2.raw alias Classic2.raw
tftp-server flash:/ringtones/ClockShop.raw alias ClockShop.raw
tftp-server flash:/ringtones/Drums2.raw alias Drums2.raw
tftp-server flash:/ringtones/FlimScore.raw alias FlimScore.raw
tftp-server flash:/ringtones/HarpSynth.raw alias HarpSynth.raw
tftp-server flash:/ringtones/Jamaica.raw alias Jamaica.raw
tftp-server flash:/ringtones/MusicBox.raw alias MusicBox.raw
tftp-server flash:/ringtones/Piano1.raw alias Piano1.raw
tftp-server flash:/ringtones/Piano2.raw alias Piano2.raw
tftp-server flash:/ringtones/Pop.raw alias Pop.raw
tftp-server flash:/CME_8.6.0_GUI/telephony_service.html alias telephony_service.html
tftp-server flash:ata_188_03_02_04_sccp_090202_a
tftp-server flash:cme-tsp-2.2.0.5/TSP-SingleLine-2.2.0.5.exe alias TSP-SingleLine-2.2.0.5.exedc
tftp-server flash:ata_03_02_04_sccp_090202_a/ATA030204SCCP090202A.zup alias ATA030204SCCP090202A.zup
tftp-server flash:cme-tsp-2.2.0.5/TSP-SingleLine-2.2.0.5.exe alias TSP-SingleLine-2.2.0.5.exe
tftp-server flash:/P0S3-8-12-00/P0S3-8-12-00.sb2 alias P0S3-8-12-00.sb2
!
control-plane
!
!
voice-port 0/3/0
!
voice-port 0/3/1
!
voice-port 0/3/2
!
voice-port 0/3/3
!
!
!
mgcp profile default
!
!
dial-peer voice 20 voip
destination-pattern 508.
session target ipv4:10.35.0.1
codec g711ulaw
!
dial-peer voice 21 voip
destination-pattern 506.
session target ipv4:10.15.0.1
!
dial-peer voice 2 voip
description **Outgoing calls to SIP.US SIP Trunk**
translation-profile outgoing SIP.US-Outgoing
destination-pattern .T
session protocol sipv2
session target dns:gw1.sip.us
voice-class sip profiles 1
dtmf-relay rtp-nte
codec g711ulaw
no vad
authentication username 522795XXXX password 7 13110510081A0A282D26XXXX realm gw1.sip.us
!
!
dial-peer voice 6000 voip
destination-pattern 69..
session protocol sipv2
session target ipv4:10.16.0.2
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 7245010 voip
destination-pattern 7245010
session protocol sipv2
session target ipv4:10.16.0.2
dtmf-relay sip-notify
codec g711ulaw
no vad
!
!
num-exp +131362XXXXXXXX 7245010
sip-ua
credentials username 522795XXXX password 7 13110510081A0A282D263XXXX realm gw1.sip.us
retry invite 2
retry register 10
timers connect 100
registrar 1 dns:gw1.sip.us expires 360 refresh-ratio 20 auth-realm gw1.sip.us
!
!
!
telephony-service
max-ephones 20
max-dn 20
ip source-address 10.25.0.1 port 2000
system message SCCP to SIP
load 7960-7940 P00308010200
time-zone 8
time-format 24
voicemail 6900
max-conferences 8 gain -6
web admin system name XXXXXXXXXXXXXX password XXXXXXXXXXXXXXXX
dn-webedit
time-webedit
transfer-system full-consult
create cnf-files version-stamp 7960 Sep 24 2015 17:58:27
!
!
ephone-dn 1
number 5069
label SIP5069
description SIP Phone
name SIP
call-forward busy 6900
call-forward noan 6900 timeout 10
hold-alert 30 originator
!
!
ephone-dn 10
number 5080
label SCCP_GUY
name SCCP GUY CIPC
call-forward busy 6900
call-forward noan 6900 timeout 10
!
!
ephone-dn 11
number 5081
label SCCP_GUY#2
name Michelle Teetly
!
!
ephone-dn 18
number 8000....
mwi on
!
!
ephone-dn 19
number 8001....
mwi off
!
!
ephone 1
device-security-mode none
mac-address 1111.1111.1111
!
!
!
ephone 10
device-security-mode none
mac-address 1060.4B47.1C77
username "SCCP_GUY" password 1234
type CIPC
button 1:10
!
!
!
ephone 11
device-security-mode none
mac-address 001F.6C7F.973B
username "cisco11" password XXXXXXXXXXXXXXX
type 7960
button 1:11
!
!
!
!
line con 0
password XXXXXXXXXXXXXXXXXXXX
logging synchronous
login
line aux 0
line 194
no activation-character
no exec
transport preferred none
transport input all
transport output lat pad telnet rlogin lapb-ta mop udptn v120 ssh
line vty 0 4
login
transport input all
!
scheduler allocate 20000 1000
ntp server ip time-c.nist.gov
end
09-30-2015 03:13 AM
Hi,
Any reason why you are putting CUE and CME in different subnets?
You are missing bind all command. It can be applied globally or on dialpeers
Regarding DTMF problem, can you share the output of debug ccsip mess when you press DTMF digits. Also, can you share the output of show call active voice called XXXX. I need to see what is getting negotiated for this call (your config is showing SIP Notify). Is DTMF relay working for SCCP phones?
Regarding 10 secs delay on 7960 phones, this is because 7960 phones are TypeA phones and perform local digit analysis before sending the call to CME. You can overcome this problem by configuring SIP dial rules. For example:
voice register dialplan 1
type 7940-7960-others
pattern 1 #dialed-pattern# timeout 0
!
voice register pool 1
type 7960
dialplan 1
10-10-2015 08:32 AM
Hi Mohammed,
I originally put both on separate subnets to help debug the devices, but it seems un-neccessary.
Thank you for the bind all on the dial-peer, that will be updated.
I did research the phones as you indicated and require at least a 7961 to try this. I now need to purchase one.
The timeouts is very helpful thank you!
Regards, Joe
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