cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
1792
Views
6
Helpful
4
Replies

SIP DTMF to Unity Express Not working

jsulliva
Level 1
Level 1

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

 

 

1 Accepted Solution

Accepted Solutions

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
 

View solution in original post

4 Replies 4

Wilson Samuel
Level 7
Level 7

Hi Could you please elaborate a bit more about the topology?

Is it something like this:

 

SIP ITSP <====>  CME / CUBE <=======>  CUE?

 

Regards

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:

  • DTMF on SIP trunk works
  • DTMF to Unity Voicemail does not work :(
  • There is a 10 second dialing delay on the SIP phone for call.

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

 

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
 

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