Dtmf-relay sip-notify

Evening Cisco techs,
Having trouble enabling sip-notify
Using CME 3.3 via (C2691-IPVOICEK9-M), Version 12.4(25d), RELEASE SOFTWARE (fc1)
Have registered cisco phones set up in CME
SIP is enabled on router. So is H323
In conf t mode via voice service voip
Upon using dtmf-relay sip-notify I get the error of unrecognised command
If other information is required. Please let me know

Just go to your SIP dial-peers:
R01#conf t
R01#dial-peer voice XX voip    << XX is you SIP dial-peer
R01(config-dial-peer)#dtmf-relay ?
cisco-rtp          Cisco Proprietary RTP
h245-alphanumeric DTMF Relay via H245 Alphanumeric IE
h245-signal        DTMF Relay via H245 Signal IE
rtp-nte            RTP Named Telephone Event RFC 2833
sip-kpml           DTMF Relay via KPML over SIP SUBCRIBE/NOTIFY
sip-notify         DTMF Relay via SIP NOTIFY messages
dial-peer voice 555 voip
destination-pattern 55555
session protocol sipv2
session target ipv4:1.1.1.1
dtmf-relay sip-notify
-Terry

Similar Messages

CME/CUE SIP Phones DTMF-Relay

Hi all,
Just looking for some clarification on this one. I'm seeing some conflicting advice about setting the DTMF-Relay on SIP Phones registered to CME with a CUE Module. I've read some documentation indicating that rtp-nte RFC2833 is the only dtmf-relay supported for SIP Phones registered to CME, however I've also read some documents indicating that sip-notify must be configured as the dtmf-relay on SIP phones when they are communicating to a CUE module. I'm assuming I'm going to need to configure an MTP on the CME, but just wondering what the official DTMF config should be under the voice register pool for SIP phones.
Thanks!

Hi logan
When doing lab with cme 7.0 and sip phones .sip phones are not recognizing the "sip-notify" dtmf-relay method .It can only recognize "rtp-nte" method and it does not matter weather you are using sip-notify or rtp-nte for a dial-peer pointing to cme .
i configured on cue
ccn subsystem sip
dtmf-relay sip-notify
end
on cme i configured a dial-peer pointing to cue
dial-peer v 3888 voip
destination-pattern 3888
session target ipv4:177.3.11.10
codec g711ulaw
no vad
session protocol sipv2
dtmf-relay sip-notiy
on my sip phones
voice register pool 1
dtmf-relay sip-notify ------> now in this case cue wont recognize dtmf tones
when i change this dtmf-relay method to rtp-nte it recognizes dtmf tones to when recording a message

DTMF relay

Hello,
I have problem relaying DTMF signals from MCU to H.320 GW.
I was trying to invite an IVR number via H.320 GW. I was able to call the IVR number. Then the IVR asks for extension number. I provided the number. However, this information was not relayed by the MCU.
Thanks

Just go to your SIP dial-peers:
R01#conf t
R01#dial-peer voice XX voip    << XX is you SIP dial-peer
R01(config-dial-peer)#dtmf-relay ?
cisco-rtp          Cisco Proprietary RTP
h245-alphanumeric DTMF Relay via H245 Alphanumeric IE
h245-signal        DTMF Relay via H245 Signal IE
rtp-nte            RTP Named Telephone Event RFC 2833
sip-kpml           DTMF Relay via KPML over SIP SUBCRIBE/NOTIFY
sip-notify         DTMF Relay via SIP NOTIFY messages
dial-peer voice 555 voip
destination-pattern 55555
session protocol sipv2
session target ipv4:1.1.1.1
dtmf-relay sip-notify
-Terry

3241 ISDN Gateway--DTMF Relay Type & local vs. national signaling

Question 1:
Is there a way to configure the 3241 so that it will flag local vs. national call types on the d-channel appropriately? The local TSP sees all calls flagged national.. They do accept both 7 and 10 digit formats for local calls so we set dialing rules to add the local area code (without a +1) to 7-digit numbers and that seems to work OK.
Querstion 2:
Does anyone know what type of DTMF relay is being used by the Cisco-acquired Tandberg/Codian 3241 ISDN Gateway? I'm pretty sure the DTMF relay type is not configurable in that box. Does this fall under the alphanumeric H.245 relay type?
I have a customer that, for outbound long distance calls (Primarily ISDN voice, but some ISDN video), requires their ISP to provide a DTMF-based billing code service. We've worked with the local TSP to verifiy what they are seeing on the switch and it looks like all digits are passing appropriately (although I also don't see any way to tell the Cisco 3241 GW to flag the call as Local vs. National
The PRI itself is provided through one local LEC and handed to a secondary TSP to provide the DTMF/billing solution.
The PRI has been works fine for local calls, but long distance (national) calls route throuth the primary provider and then pass to the secondary provider. We've worked with the local TSP to verifiy what they are seeing on the switch and it looks like all digits are passing appropriately. Logs from the ISDN gateway do not show opening any audio or h.245 channel, no 'biiling tone-prompt' is heard, and we cannot access/enable the touch tone menu on the endpoint during the time the ISDN GW is attempting to set up the call. I suspect that either the billing code service is not turned up yet, but I'm hoping its is not a H323/H.245 and/or endpoint cpability that is preventing this from working.
In order to troubleshoot with the service provider I assume I'll need to know what DTMF type we are sending, I am hoping they could also tell me by debuggin the switch for any signaling they are receiving from us.
If anyone has any experience setting up similar service on CUCM i'd love to hear what was required on that system for DTMF relay, and if there is anything else I may need to ask the TSP
Thanks!

DTMF relay for H323 is handled with 3 commands.
cisco-rtp Cisco Proprietary RTP
h245-alphanumeric DTMF Relay via H245 Alphanumeric IE
h245-signal DTMF Relay via H245 Signal IE
Your best bet is h245-alphanumeric. The dtmf-relay rtp-nte command is used for SIP dial-peers.
The DTMF relay feature transports DTMF tones generated after call establishment out of band using either a standard H.323 out-of-band method and a proprietary RTP-based mechanism, or for SIP calls, an NTE RTP packet.
http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a00800d62d2.shtml#Solution4
Please rate any helpful posts
Thanks
Fred

Send DTMF with SIP INFO (c2600) configuration question

I have a cisco 2600 with VIC-2FXS port as VOIP Gateway, connecting to SIP Server to receive SIP Incoming calls. I am able to receive call and the vocie has been pass through both way; and I would like the 2600 send DTMF as SIP info but was not able to do so. I have ios 12.3, and from this configuration guide http://www.cisco.com/en/US/docs/ios/12_3/sip/configuration/guide/chapter8.html#wp1048824 , it does not require any config for SIP info. I must missing something here, please advice. Thanks.
The config is following -
version 12.3
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
hostname abc
boot-start-marker
boot-end-marker
enable secret 5 xx
enable password xx
memory-size iomem 10
no aaa new-model
ip subnet-zero
no ip routing
no ip cef
interface Ethernet0/0
ip address 192.168.1.15 255.255.255.0
no ip route-cache
full-duplex
no ip http server
ip classless
voice-port 1/0/0
voice-port 1/0/1
voice-port 1/1/0
voice-port 1/1/1
dial-peer voice 1 pots
destination-pattern 1000
port 1/1/0
dial-peer voice 2 pots
destination-pattern 1001
port 1/1/1
dial-peer voice 10 voip
destination-pattern 1.T
session protocol sipv2
session target ipv4:192.168.1.224:5061
session transport udp
codec g711ulaw
dial-peer voice 3 pots
destination-pattern 1100
port 1/0/0
dial-peer voice 4 pots
destination-pattern 1101
port 1/0/1
line con 0
line aux 0
line vty 0 4
login
end

I tried to set it, and for IOS 12.3(26) - the latest for 2610 - which dose not have that option. I use dtmf-relay rtp-nte instead; but it did not send RFC2833 event. From ethereal, no OOB events. It seems that the config I have does not have OOB DTMF enable; I compare the config I have with other examples but can not found anything wrong. Any suggestion, and what debug message I should enable, that may help to identify the issue.
Thanks.
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
interface Ethernet0/0
ip address 192.168.1.15 255.255.255.0
full-duplex
voice-port 1/0/0
voice-port 1/0/1
voice-port 1/1/0
voice-port 1/1/1
dial-peer voice 1 pots
destination-pattern 1000
port 1/1/0
dial-peer voice 2 pots
destination-pattern 1001
port 1/1/1
dial-peer voice 10 voip
description Outbound Calls
destination-pattern 1.T
voice-class codec 1
session protocol sipv2
session target ipv4:192.168.1.250
session transport udp
dtmf-relay rtp-nte
no vad
dial-peer voice 3 pots
destination-pattern 1100
port 1/0/0
dial-peer voice 4 pots
destination-pattern 1101
port 1/0/1
dial-peer voice 100 pots
destination-pattern 8...
port 1/1/0
forward-digits 3
dial-peer voice 20 voip
description Incoming calls from PBX
incoming called-number .T
voice-class codec 1
session protocol sipv2
session target ipv4:192.168.1.250
dtmf-relay rtp-nte
no vad

CUBE - New Deployment Issue - Not working DTMF Relay

Hello,
Scheme:
Cisco SCCP-based IP Phone > CUCM 9.1 w/ SIP Trunk > CUBE (28XX, 151-4.M7) > SIP ITSP
CUCM Active Call Proc. Node IP: 10.10.10.9
CUBE Inside Interface IP: 10.10.10.10
CUBE Outside Interface IP: 20.20.20.20
Cisco IP Phone: 10.10.10.8
ITSP SBC IP: 30.30.30.30
ITSP SIP domain: itsp.domain
Calling Pty: 9017654321 (translated in CUCM's route pattern which addresses CUBE)
Called Pty: 9011234567
While call was connected calling party dialed consequently 0,1,2,3,4 but far-end IVR does not react :(
Symptom:
While outbound call is connected calling party (IP Phone) dials digits which are not detected by any far-end PSTN (non-corporate) IVR at all.
Thoughts:
ITSP support only inband relay (RFC2833, Named Telephone Events or NTEs).
Using NTE provides a standard way to transport DTMF tones in RTP packets.
Thus rtp-nte is configured for both CUCM and ITSP dial-peers on CUBE.
While initial troubleshooting found that for the active call inbound CUBE's leg shows rtp-nte, but outbound inband-voice.
A have an assumption that ITSP doesn't give us 101=rtp-nte payload in 183 Response but I'm not sure.
m=audio 10318 RTP/AVP 8
b=AS:64
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:20
Questions:
1.How to make CUBE to successfully relay DTMF in according to ITSP requirement?
2. Why 'show call act/hist voice brief' doesn't show call id? All my attempts are identified as 2... )
It is hard to differentiate b/w call active/history records..

Ayodeji,
Thanks for your feedback.
If you look through the already attached output 'show call act voice' of the file 'case-no-dtmf_-_cube-show-20140210-1.txt' you will find the following:
PeerId=101
CallOrigin=2
tx_DtmfRelay=rtp-nte
CallDuration=00:00:05 sec
PeerId=201
CallOrigin=1
tx_DtmfRelay=inband-voice
CallDuration=00:00:05 sec
2 : 230 18:40:38.048 MSK Tue Feb 10 2015.1 +1890 pid:101 Answer 79017654321 active
dur 00:00:04 tx:234/37440 rx:233/37280
IP 10.10.10.8:22688 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711alaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
2 : 231 18:40:38.068 MSK Tue Feb 10 2015.1 +1860 pid:201 Originate 79011234567 active
dur 00:00:04 tx:233/37280 rx:307/49120
IP 30.30.30.30:10318 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711alaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
This means that the target dial-peers (inbound and outbound are matched as designed).
dial-peer voice 101 voip
description -= inbound leg from CUCM to CUBE =-
session protocol sipv2
incoming called-number x
voice-class codec 1
voice-class sip bind control source-interface Loopback0
voice-class sip bind media source-interface Loopback0
dtmf-relay rtp-nte
no vad
dial-peer voice 201 voip
description -= outbound leg from CUBE to ITSP =-
translation-profile outgoing cdpn-delete-prefix-00XX7
max-conn 40
destination-pattern x
session protocol sipv2
session target dns:sbc.itsp.domain
voice-class codec 1
voice-class sip profiles 1
voice-class sip bind control source-interface Vlan100
voice-class sip bind media source-interface Vlan100
dtmf-relay rtp-nte
no vad
Now I have an argue with ITSP to make them send me NTE in their 183/200 response..
I've also:
1. Tried to disable dtmf-relay at all on dial-peers (trying inband-voice) but this doesn't work and also not recommended AFAIK.
2. Changed the value for SIP Trunk DTMF Signaling Method from 'No preference' to 'RFC 2833' w/ reset applied recommended by Suresh. No luck.

DTMF Issue in SIP

Hi All,
I have an issue here. The DTMF is not recognized by the Unity when user wants to do remote login to voicemail box by pressing *
Call Flow : T1 --> AS5400 --> SIP Trunk --> CUCM 9.1.2 --> SCCP --> CUC 9.1.2
Time : Nov 12 20:06:56.417 UTC
Calling Party Number i = 0x1183, '914466553077'
Called Party Number i = 0xA1, '2067677' - 99992067677
I can see in CCAPI, * being pressed and NOTIFY message is sent to CUCM, and I get 403 Forbidden as response.
The dial-peer configuration point to CUCM is below
dial-peer voice 4320 voip
tone ringback alert-no-PI
description --- PSTN to XXX 9999.XXXXXXX ---
preference 1
destination-pattern 9999.......$
no modem passthrough
session protocol sipv2
session target ipv4:XXXXX
voice-class codec 1
voice-class sip early-offer forced
voice-class sip options-keepalive
dtmf-relay sip-notify rtp-nte
fax rate 7200
ip qos dscp cs3 signaling
no vad
Logs are attached. Please help me to find out the issue.

ok..We need to use a different approach to resolve this..We need to prefix calls coming from cucm so as to break up the overlapping issue..
do this..
go to cucm, search for the Route list you use for outbound calls, click on the route group associated with it.
Under called party xformation
under discard digits: use to none
prefix digit outgoing calls: add 141 as shown below

Cisco SIP Phone 9971 won't register on CME 8.6 or 8.5 Please HELP

Please help me , I have problem with registering Cisco SIP phone 9971 with CME 8.6 on ISR 2901.
I configured CME for SIP clients, then I add configuration for 9971 phone and create profiles. Phone downloaded SEP...xml file from CME,after that phone look for g4-tones.xml and gd-sip.jar files, I added them to CME after that phone downloaded them and reboot. Now phone is stuck in some kind of loop and does not register on CME.
On phone log I can see repeting next few messeges.
12:01:58a No DNS Server IP
12:01:59a Updating Trust list
12:01:59a No Trust List instaled
12:01:59a SEP04C5AB03B0D.cnf.xml (TFTP) // at this time phone download SEP...xml file from CME
12:02:00a VPN Error: VPN is not Configured
on CME if issue DEBUG TFTP EVENTS i receive next few lines
*Aug 18 18:20:19.891: TFTP: Looking for CTLSEP04C5A4B03B0D.tlv
*Aug 18 18:20:19.987: TFTP: Looking for ITLSEP04C5A4B03B0D.tlv
*Aug 18 18:20:20.083: TFTP: Looking for ITLFile.tlv
*Aug 18 18:20:20.347: TFTP: Looking for SEP04C5A4B03B0D.cnf.xml
*Aug 18 18:20:20.351: TFTP: Opened flash:/SEP04C5A4B03B0D.cnf.xml, fd 14, size 4585 for process 141
*Aug 18 18:20:20.363: TFTP: Finished flash:/SEP04C5A4B03B0D.cnf.xml, time 00:00:00 for process 141
here you can see verison info of CME
Cisco IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.1(4)M, RELEASE SOFTWARE (fc1)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2011 by Cisco Systems, Inc.
Compiled Thu 24-Mar-11 15:31 by prod_rel_team
ROM: System Bootstrap, Version 15.0(1r)M9, RELEASE SOFTWARE (fc1)
ELTOSAN_ROUTER uptime is 1 hour, 50 minutes
System returned to ROM by reload at 16:29:20 UTC Thu Aug 18 2011
System image file is "flash:/c2900-universalk9-mz.SPA.151-4.M.bin"
Last reload type: Normal Reload
Last reload reason: Reload Command
Cisco CISCO2901/K9 (revision 1.0) with 471040K/53248K bytes of memory.
Processor board ID FGL1508252Y
3 Gigabit Ethernet interfaces
2 terminal lines
1 Virtual Private Network (VPN) Module
4 Voice FXO interfaces
4 Voice FXS interfaces
1 Internal Services Module (ISM) with Services Ready Engine (SRE)
   Survivable Remote Site Voicemail (SRSV) on Cisco Unity Express (CUE) 8.5.1 in slot/sub-slot 0/0
DRAM configuration is 64 bits wide with parity enabled.
255K bytes of non-volatile configuration memory.
254464K bytes of ATA System CompactFlash 0 (Read/Write)
License Info:
License UDI:
Device#   PID                   SN
*0        CISCO2901/K9          xxxxxxxxxxxxx
Technology Package License Information for Module:'c2900'
Technology    Technology-package          Technology-package
              Current       Type          Next reboot
ipbase        ipbasek9      Permanent     ipbasek9
security      securityk9    Permanent     securityk9
uc            uck9          Permanent     uck9
data          None          None          None
Configuration register is 0x2102
this is RUNNING CONFIGURATION
! Last configuration change at 16:10:12 UTC Thu Aug 18 2011
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname ELTOSAN_ROUTER
boot-start-marker
boot system flash:/c2900-universalk9-mz.SPA.151-4.M.bin
boot-end-marker
no aaa new-model
no ipv6 cef
ip source-route
no ip routing
no ip cef
no ip dhcp use vrf connected
ip dhcp excluded-address 192.168.5.1 192.168.5.10
ip dhcp excluded-address 192.168.5.200 192.168.5.255
ip dhcp pool phone
   network 192.168.5.0 255.255.255.0
   default-router 192.168.5.251
   option 150 ip 192.168.5.251
ip dhcp pool data
   relay source 192.168.2.0 255.255.255.0
   relay destination 192.168.2.201
multilink bundle-name authenticated
crypto pki token default removal timeout 0
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
fax protocol pass-through g711alaw
sip
registrar server expires max 3600 min 120
voice register global
mode cme
source-address 192.168.5.251 port 5060
max-dn 6
max-pool 6
load 9971 sip9971.9-1-1SR1.loads
authenticate register
tftp-path flash:
create profile sync 0005135312289902
voice register dn 1
number 207
allow watch
name GossaVM
label 207
voice register dn 3
number 101
name Dejan
label 101
mwi
voice register pool 1
id mac 000C.29C5.0011
number 1 dn 1
dtmf-relay sip-notify
username testvm password testera
codec g711alaw
voice register pool 3
id mac 04C5.A4B0.3B0D
type 9971
number 3 dn 3
presence call-list
dtmf-relay rtp-nte
username dejan password 1234
codec g711alaw
no vad
license udi pid CISCO2901/K9 sn xxxxxxxxxxxx
hw-module ism 0
hw-module pvdm 0/0
redundancy
interface GigabitEthernet0/0
description INTERFACE INTERNAL
no ip address
no ip route-cache
duplex auto
speed auto
no mop enabled
interface GigabitEthernet0/0.2
description LAN DATA
encapsulation dot1Q 2
ip address 192.168.2.251 255.255.255.0
no ip route-cache
interface GigabitEthernet0/0.5
description LAN VOICE
encapsulation dot1Q 5
ip address 192.168.5.251 255.255.255.0
no ip route-cache
interface ISM0/0
no ip address
no ip route-cache
shutdown
!Application: SRSV-CUE Running on ISM
interface GigabitEthernet0/1
no ip address
no ip route-cache
shutdown
duplex auto
speed auto
interface ISM0/1
description Internal switch interface connected to Internal Service Module
shutdown
interface Vlan1
no ip address
no ip route-cache
shutdown
ip forward-protocol nd
no ip http server
no ip http secure-server
snmp-server community public RO
tftp-server flash:dkern9971.100609R2-9-1-1SR1.sebn alias dkern9971.100609R2-9-1-1SR1.sebn
tftp-server flash:kern9971.9-1-1SR1.sebn alias kern9971.9-1-1SR1.sebn
tftp-server flash:rootfs9971.9-1-1SR1.sebn alias rootfs9971.9-1-1SR1.sebn
tftp-server flash:sboot9971.031610R1-9-1-1SR1.sebn alias sboot9971.031610R1-9-1-1SR1.sebn
tftp-server flash:skern9971.022809R2-9-1-1SR1.sebn alias skern9971.022809R2-9-1-1SR1.sebn
tftp-server flash:sip9971.9-1-1SR1.loads alias sip9971.9-1-1SR1.loads
tftp-server flash:United_States/g4-tones.xml
tftp-server flash:English_United_States/gd-sip.jar
control-plane
voice-port 0/0/0
voice-port 0/0/1
voice-port 0/0/2
voice-port 0/0/3
voice-port 0/1/0
voice-port 0/1/1
voice-port 0/1/2
voice-port 0/1/3
mgcp profile default
gatekeeper
shutdown
line con 0
line aux 0
line 67
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
password jebiga
login
transport input all
end
I did not have any kind of problem with X-LITE to register to CME. also try with few SCCP phones 7940 and I did not any kind of problem .
this is content of SEP....xml file for 9971
<device>
<deviceProtocol>SIP</deviceProtocol>
<devicePool>
<dateTimeSetting>
<dateTemplate>M/D/YA</dateTemplate>
<timeZone>Pacific Standard/Daylight Time</timeZone>
<ntps>
<ntp priority="0">
<name>0.0.0.0</name>
<ntpMode>unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<sipPort>5060</sipPort>
</ports>
<processNodeName>192.168.5.251</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<sipProfile>
<sipProxies>
<registerWithProxy>true</registerWithProxy>
</sipProxies>
<sipCallFeatures>
<cnfJoinEnabled>true</cnfJoinEnabled>
<localCfwdEnable>true</localCfwdEnable>
<callForwardURI>service-uri-cfwdall</callForwardURI>
<callPickupURI>service-uri-pickup</callPickupURI>
<callPickupGroupURI>service-uri-gpickup</callPickupGroupURI>
<callHoldRingback>2</callHoldRingback>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>2</anonymousCallBlock>
<callerIdBlocking>2</callerIdBlocking>
<dndControl>2</dndControl>
<remoteCcEnable>true</remoteCcEnable>
</sipCallFeatures>
<sipStack>
<remotePartyID>true</remotePartyID>
</sipStack>
<sipLines>
<line button="1" lineIndex="1">
<featureID>9</featureID>
<featureLabel></featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name></name>
<displayName></displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>1</callWaiting>
<authName>dejan</authName>
<authPassword>1234</authPassword>
<sharedLine>false</sharedLine>
<messagesNumber></messagesNumber>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>true</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button="2" lineIndex="2">
<featureID>9</featureID>
<featureLabel>101</featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>101</name>
<displayName>Dejan Rakic</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>1</callWaiting>
<authName>dejan</authName>
<authPassword>1234</authPassword>
<sharedLine>false</sharedLine>
<messagesNumber></messagesNumber>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>true</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
<enableVad>true</enableVad>
<preferredCodec>g711alaw</preferredCodec>
<dialTemplate></dialTemplate>
<kpml>1</kpml>
<phoneLabel></phoneLabel>
<stutterMsgWaiting>2</stutterMsgWaiting>
<disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig>
<dscpForAudio>184</dscpForAudio>
<dscpVideo>136</dscpVideo>
</sipProfile>
<commonProfile>
<phonePassword>1234</phonePassword>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<featurePolicyFile>featurePolicyDefault.xml</featurePolicyFile>
<loadInformation>sip9971.9-1-1SR1.loads</loadInformation>
<vendorConfig>
</vendorConfig>
<commonConfig>
<videoCapability>0</videoCapability>
<ciscoCamera>0</ciscoCamera>
</commonConfig>
<sshUserId>dejan</sshUserId>
<sshPassword>1234</sshPassword>
<userId></userId>
<phoneServices>
<provisioning>2</provisioning>
<phoneService type="1" category="0">
<name>Missed Calls</name>
<phoneLabel></phoneLabel>
<url>Application:Cisco/MissedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Received Calls</name>
<phoneLabel></phoneLabel>
<url>Application:Cisco/ReceivedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Placed Calls</name>
<phoneLabel></phoneLabel>
<url>Application:Cisco/PlacedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="2" category="0">
<name>Voicemail</name>
<phoneLabel></phoneLabel>
<url>Application:Cisco/Voicemail</url>
<vendor></vendor>
<version></version>
</phoneService>
</phoneServices>
<versionStamp>0131511014412102</versionStamp>
<userLocale>
<name>English_United_States</name>
<langCode>en</langCode>
</userLocale>
<networkLocale>United_States</networkLocale>
<networkLocaleInfo>
<name>United_States</name>
</networkLocaleInfo>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<servicesURL>http://192.168.5.251:80/CMEserverForPhone/serviceurl</servicesURL>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>2</transportLayerProtocol>
</device>

Hello,
I'm facing exactly the same problem, that is:
a Cisco SIP Phone 9971 won't register on CME 8.6 running on a 2811
I have read all the postings to this Forum, but I have not been able to solve it.
In my case the commands voice register dn and voice register pool are OK.
So frankly, I have no idea what I could be missing.
I'm pasting the Router's config.
I hope somebody is able to point me in the right direction.
Here is the config. Thank you!
C2811#sh run
Building configuration...
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname C2811
no aaa new-model
dot11 syslog
ip source-route
ip cef
ip dhcp excluded-address 172.25.140.1 172.25.140.10
ip dhcp excluded-address 172.35.140.1 172.35.140.10
ip dhcp pool Data
network 172.25.140.0 255.255.255.0
default-router 172.25.140.1
option 150 ip 172.25.140.1
dns-server 172.25.140.1
ip dhcp pool Voice
network 172.35.140.0 255.255.255.0
default-router 172.35.140.1
option 150 ip 172.35.140.1
dns-server 172.35.140.1
no ip domain lookup
no ipv6 cef
multilink bundle-name authenticated
voice service voip
allow-connections sip to sip
sip
registrar server expires max 3600 min 120
voice register global
mode cme
source-address 172.25.140.1 port 5060
max-dn 40
max-pool 42
load 9971 sip9971.9-4-1-9.loads
authenticate register
authenticate realm cisco
tftp-path flash:
create profile sync 0004820400584603
voice register dn 1
number 1010
allow watch
name Phone10
label Phone10
mwi
voice register pool 1
id mac 189C.5DB6.BD09
type 9971
number 1 dn 1
presence call-list
dtmf-relay rtp-nte
username adm password adm
call-forward b2bua busy 68600
codec g711ulaw
no vad
camera
video
voice-card 0
crypto pki token default removal timeout 0
crypto pki trustpoint TP-self-signed-1879153754
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-1879153754
revocation-check none
rsakeypair TP-self-signed-1879153754
crypto pki certificate chain TP-self-signed-1879153754
certificate self-signed 01
(details ommited)
license udi pid CISCO2811 sn FTX1146A44H
username admin privilege 15 password 0 admin
redundancy
interface FastEthernet0/0
no ip address
duplex auto
speed auto
interface FastEthernet0/0.25
description Data VLAN
encapsulation dot1Q 25
ip address 172.25.140.1 255.255.255.0
interface FastEthernet0/0.35
description Voice VLAN
encapsulation dot1Q 35
ip address 172.35.140.1 255.255.255.0
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 600 life 86400 requests 10000
tftp-server flash:P00308010200.bin
tftp-server flash:P00308010200.sbn
tftp-server flash:P00308010200.sb2
tftp-server flash:P00308010200.loads
tftp-server flash:SCCP42.9-3-1SR3-1S.loads
tftp-server flash:apps42.9-3-1ES19.sbn
tftp-server flash:cnu42.9-3-1ES19.sbn
tftp-server flash:cvm42sccp.9-3-1ES19.sbn
tftp-server flash:dsp42.9-3-1ES19.sbn
tftp-server flash:jar42sccp.9-3-1ES19.sbn
tftp-server flash:term42.default.loads
tftp-server flash:term62.default.loads
tftp-server flash:SCCP45.9-3-1SR3-1S.loads
tftp-server flash:apps45.9-3-1ES19.sbn
tftp-server flash:cnu45.9-3-1ES19.sbn
tftp-server flash:cvm45sccp.9-3-1ES19.sbn
tftp-server flash:dsp45.9-3-1ES19.sbn
tftp-server flash:jar45sccp.9-3-1ES19.sbn
tftp-server flash:term45.default.loads
tftp-server flash:term65.default.loads
tftp-server flash:/Ringtones/Ringlist.xml alias Ringlist.xml
tftp-server flash:/Ringtones/DistinctiveRingList.xml alias DistinctiveRingList.x
ml
tftp-server flash:sip9971.9-4-1-9.loads
tftp-server flash:kern9971.9-4-1-9.sebn
tftp-server flash:rootfs9971.9-4-1-9.sebn
tftp-server flash:dkern9971.100609R2-9-4-1-9.sebn
tftp-server flash:sboot9971.031610R1-9-4-1-9.sebn
tftp-server flash:skern9971.022809R2-9-4-1-9.sebn
tftp-server flash:/g4-tones.xml alias United_States/g4-tones.xml
tftp-server flash:/gd-sip.jar alias English_United_States/gd-sip.jar
control-plane
mgcp profile default
telephony-service
max-ephones 24
max-dn 48
ip source-address 172.25.140.1 port 2000
cnf-file location flash:
load 7960-7940 P00308010200
load 7942 SCCP42.9-3-1SR3-1S.loads
load 7945 SCCP45.9-3-1SR3-1S.loads
load 7962 SCCP42.9-3-1SR3-1S.loads
load 7965 SCCP45.9-3-1SR3-1S.loads
max-conferences 8 gain -6
dn-webedit
transfer-system full-consult
create cnf-files version-stamp 7960 Feb 11 2014 07:18:32
ephone-dn 1
number 1001
description Phone 1
name Phone 1
hold-alert 30 originator
ephone-dn 2
number 1002
description Phone 2
name Phone 2
hold-alert 30 originator
ephone-dn 3
number 1003
description Phone 3
name Phone 3
hold-alert 30 originator
ephone 1
device-security-mode none
mac-address 001C.58FB.6E0F
button 1:1
ephone 2
device-security-mode none
mac-address 0014.A981.7F8A
button 1:2
ephone 3
device-security-mode none
mac-address 0006.5356.A4B8
button 1:3
alias exec con conf t
alias exec sib show ip int brief
alias exec srb show run | b
alias exec sri show run int
line con 0
exec-timeout 0 0
logging synchronous
line aux 0
line vty 0 4
privilege level 15
login local
transport input telnet ssh
transport output telnet ssh
line vty 5 15
privilege level 15
login local
transport input telnet ssh
transport output telnet ssh
scheduler allocate 20000 1000
ntp master 1
end
C2811#

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#

CME SIP issue - Cisco 7821 phone not registering

Hi
I am having issues with getting a Cisco 7821 phone to register.
Current deployment is with Cisco 6921 phones SCCP registration
SIP integration with CUE
SIP integration with Mitel system
c2951-universalk9-mz.SPA.154-3.M1.bin (CME 10.5)
In flash:
rootfs78xx.10-1-1SR1-4.sbn
kern78xx.10-1-1SR1-4.sbn
sboot78xx.10-1-1SR1-4.sbn
sip78xx.10-1-1SR1-4.loads
The 7821 phone gets IP address but fails to register. Please could somebody let me know why phone is not registering.
Configuration below (10.245.226.132 is CME address) .
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol pass-through g711ulaw
modem passthrough nse codec g711ulaw redundancy maximum-sessions 5
h323
sip
registrar server expires max 600 min 60
options-ping 90
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
voice register global
mode cme
source-address 10.245.226.132 port 5060
max-dn 30
max-pool 10
load 7821 sip78xx.10-1-1SR1-4
authenticate register
authenticate realm all
timezone 22
date-format D/M/Y
voicemail 590
tftp-path flash:
create profile sync 0061443538560005
network-locale GB
voice register dn 1
number 1010
name user1
label user1
mwi
voice register pool 1
busy-trigger-per-button 2
id mac F09E.636E.63F2
type 7821
number 1 dn 1
presence call-list
dtmf-relay rtp-nte
username 1010 password 123
codec g711ulaw
no vad
dial-peer voice 391 voip
description *** Auto Attendant ***
destination-pattern 399
session protocol sipv2
session target ipv4:10.245.226.131
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 392 voip
description *** Administration Via Telephone ***
destination-pattern 392
session protocol sipv2
session target ipv4:10.245.226.131
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 393 voip
description *** Extension Assigner ***
service ea out-bound
destination-pattern 393
session target ipv4:10.245.226.132
dial-peer voice 590 voip
description *** Voice Mail Pilot ***
destination-pattern 590
b2bua
session protocol sipv2
session target ipv4:10.245.226.131
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 1 pots
description ** Match all incoming POTS calls **
translation-profile incoming IncomingPSTNcalls
incoming called-number .
direct-inward-dial
dial-peer voice 899 voip
description Call to Mitel
translation-profile incoming Prefix9
translation-profile outgoing rem44
destination-pattern [23]..
session protocol sipv2
session target ipv4:192.168.114.2
voice-class codec 1
dtmf-relay rtp-nte
no vad
interface GigabitEthernet0/0
description *** Connection to Mitel Phone System ***
ip address 192.168.114.5 255.255.255.248
duplex auto
speed auto
interface ISM0/0
description *** Connection to Cisco Unity Express ***
ip unnumbered GigabitEthernet0/1
service-module ip address 10.245.226.131 255.255.255.128
!Application: CUE Running on ISM
service-module ip default-gateway 10.245.226.132
interface GigabitEthernet0/1
description *** Connection to IP Phone LAN ***
ip address 10.245.226.132 255.255.255.128
duplex auto
speed auto
ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip http path flash:
ip route 0.0.0.0 0.0.0.0 10.245.226.129
ip route 10.245.226.131 255.255.2
tftp-server flash:apps37sccp.1-4-4-0.bin
tftp-server flash:sip78xx.10-1-1SR1-4.loads
tftp-server flash:rootfs78xx.10-1-1SR1-4.sbn
tftp-server flash:sboot78xx.10-1-1SR1-4.sbn
sip-ua
mwi-server ipv4:10.245.226.131 expires 3600 port 5060 transport udp
registrar ipv4:10.245.226.132 expires 600
gatekeeper
shutdown
telephony-service
authentication credential cmeadmin c4p1ta2012
xml user xmladmin password xmladmin 15
extension-assigner tag-type provision-tag
max-ephones 104
max-dn 299
ip source-address 10.245.226.132 port 2000
auto assign 101 to 105
no service directed-pickup
timeouts interdigit 5
system message CFGS
url services http://10.245.226.131/voiceview/common/login.do
url authentication http://10.245.226.132/CCMCIP/authenticate.asp
cnf-file location flash:
cnf-file perphone
load 7931 SCCP31.9-2-1S
load 6921 SCCP69xx.9-2-1-0
time-zone 22
date-format dd-mm-yy
voicemail 590
max-conferences 8 gain -6
call-forward pattern .T
moh enable-g711 "music-on-hold.au"
web admin system name cmeadmin secret 5 $1$QmIK$46fDKVSudMxzI2bRp/Ef7/
time-webedit
transfer-system full-consult
transfer-pattern .T
secondary-dialtone 9
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 298
number 598...
mwi on
ephone-dn 299
number 599...
mwi off

Page 7 of the following link recommends that you use option 150 with the Cisco 7800 series phones and use option 66 if you cannot use option 150
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/7821_7841_7861/10_1/english/admin_guide/PA2D_BK_AB3F74DA_00_admin-7821-7841-7861-10_0/PA2D_BK_AB3F74DA_00_admin-7821-7841-7861-10_0_chapter_01.pdf
Dynamic Host Configuration Protocol (DHCP)
DHCP dynamically allocates and assigns an IP address to network devices.
DHCP enables you to connect an IP phone into the network and have the phone become operational without your needing to manually assign an IP address or to configure additional network parameters.
DHCP is enabled by default. If disabled, you must manually configure the IP address, subnet mask, gateway, and a TFTP server on each phone locally.
Cisco recommends that you use DHCP custom option 150. With this method, you configure the TFTP server IP address as the option value. For additional supported DHCP configurations, go to the "Dynamic Host Configuration Protocol" chapter and the "Cisco TFTP" chapter in the Cisco Unified Communications Manager System Guide.
Note
If you cannot use option 150, you may try using DHCP option 66.

Cisco CP-78XX SIP Phone Pickup Not Work on CME

Hi,
I configured some SIP phones (CP-7821, CP-7841) with pickup function. Is it the Pickup / GPickup soft keys not function as the SIP phone? If yes, then I can use the FAC to access that? And I tried the FAC std. / custom as the pickup / gpickup .. both not work ... I don't know how to use the FAC on CME? As the FAC std., if I pickup local, that I should press (**3) > call?
Ref.:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm/cmecover.html#45535
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm/cmefacs.html#30064
This is the configuration:
CME-SIP-Phone#sh run
Building configuration...
Current configuration : 5413 bytes
! Last configuration change at 11:06:12 UTC Fri Nov 28 2014 by mtlops
version 15.4
no service pad
service tcp-keepalives-in
service tcp-keepalives-out
service timestamps debug datetime msec localtime show-timezone
service timestamps log datetime msec localtime show-timezone
service password-encryption
service sequence-numbers
hostname CME-SIP-Phone
boot-start-marker
boot system flash:c2900-universalk9-mz.SPA.154-2.T1.bin
boot-end-marker
! card type command needed for slot/vwic-slot 0/0
enable secret 5 $XXXXXXXXXXXXXXXXXXXXXXXX
aaa new-model
aaa authentication login default local
aaa authorization console
aaa authorization exec default local
aaa session-id common
ip cef
no ipv6 cef
multilink bundle-name authenticated
stcapp feature access-code
voice-card 0
dspfarm
dsp services dspfarm
voice service pots
voice service voip
ip address trusted list
ipv4 10.118.0.0 255.255.255.0
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 h225-notify cid-update
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
no h225 timeout keepalive
call preserve
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
registrar server expires max 600 min 60
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
voice class h323 1
h225 timeout tcp establish 3
call preserve
voice class custom-cptone ABC-Company
dualtone disconnect
frequency 425
cadence 500 500
voice register pool-type 7821
description Cisco IP Phone 7821
reference-pooltype 6921
voice register pool-type 7841
description Cisco IP Phone 7841
reference-pooltype 6941
voice register global
mode cme
source-address 10.118.0.10 port 5060
timeouts interdigit 2
max-dn 200
max-pool 100
authenticate register
authenticate realm all
timezone 42
time-format 24
date-format D/M/Y
mwi stutter
mwi reg-e164
voicemail 5000
call-feature-uri pickup http://10.118.0.10/pickup
call-feature-uri gpickup http://10.118.0.10/gpickup
tftp-path flash:
file text
create profile sync 0001170446349417
ntp-server 10.118.0.10 mode unicast
ip qos dscp af11 media
ip qos dscp cs2 signal
ip qos dscp af43 video
ip qos dscp 25 service
camera
video
voice register dn 2
number 1000
pickup-call any-group
pickup-group 1
name BB Leung
label BB Leung
voice register dn 3
number 1001
pickup-call any-group
pickup-group 1
name CC Chan
label CC Chan
voice register dn 4
number 1002
pickup-call any-group
pickup-group 1
name DD Leung
label DD Leung
voice register dn 50
mwi
voice register template 1
softkeys hold Newcall Resume
softkeys idle Newcall Redial Gpickup Pickup Cfwdall DND
softkeys seized Cfwdall Endcall Redial
softkeys connected Confrn Endcall Hold Trnsfer
voice register pool 1
busy-trigger-per-button 1
id mac A8XX.XXXX.XXXX
type 7841
number 1 dn 2
template 1
dtmf-relay sip-notify
username 1001 password 112233
codec g711ulaw
no vad
voice register pool 2
busy-trigger-per-button 1
id mac 50XX.XXXX.XXXX
type 7841
number 1 dn 3
template 1
dtmf-relay sip-notify
username 1002 password 112233
codec g711ulaw
no vad
voice register pool 3
busy-trigger-per-button 1
id mac 00XX.XXXX.XXXX
type 7821
number 1 dn 4
template 1
dtmf-relay sip-notify
username 1003 password 112233
codec g711ulaw
no vad
license udi pid CISCO2921/K9 sn FHK1407F25D
license accept end user agreement
license boot c2900 technology-package uck9
hw-module pvdm 0/0
hw-module sm 1
username mtlops privilege 15 secret 5 $1$0qqx$1WGdfRW.flJrwmY7k8eUy0
redundancy
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
ip address 10.118.0.10 255.255.255.0
duplex auto
speed auto
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
interface GigabitEthernet0/2
no ip address
shutdown
duplex auto
speed auto
interface SM1/0
no ip address
shutdown
service-module fail-open
interface SM1/1
no ip address
interface Vlan1
no ip address
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 0.0.0.0 0.0.0.0 10.118.0.1
control-plane
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
mgcp profile default
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 7
associate application SCCP
shutdown
gatekeeper
shutdown
telephony-service
max-conferences 8 gain -6
transfer-system full-consult
fac standard
line con 0
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line 67
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
transport input all
scheduler allocate 20000 1000
end
CME-SIP-Phone#sh telephony-service fac
telephony-service fac standard
callfwd all **1
callfwd cancel **2
pickup local **3
pickup group **4
pickup direct **5
park **6
dnd **7
redial **8
voicemail **9
ephone-hunt join *3
ephone-hunt cancel #3
ephone-hunt hlog *4
ephone-hunt hlog-phone *5
trnsfvm *6
dpark-retrieval *0
cancel call waiting *1

VPN is not Configured prints on all phones now with the built-in VPN client if VPN isn't configured. That's normal and is just cosmetic. That should not be causing your registration issues.

SIP to SIP call on CME 8.6

Hi all, I'm trying to setup a video call between 9951 and IP door station 2N Helios IP that support H264 over SIP.
The audio call is working well but I see only black screen on my 9951, I don't see video also with other SIP client connected on the same CME 8.6.
This is my config:
voice service voip
no notify redirect ip2ip
clid network-provided
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback cisco
modem passthrough nse codec g711alaw
sip
registrar server
asymmetric payload full
voice register global
mode cme
source-address 192.168.99.204 port 5060
bandwidth video tias-modifier 512000 negotiate end-to-end
max-dn 20
max-pool 10
load 9951 sip9951.9-1-1SR1
authenticate presence
authenticate register
authenticate realm cme
timezone 23
date-format D/M/Y
tftp-path flash:
file text
create profile sync 0010244609221862
network-locale IT
user-locale IT
ntp-server 192.168.99.254 mode directedbroadcast
camera
video
voice register dn 1
number 200
allow watch
name 200
no-reg
label 200
voice register dn 2
number 201
allow watch
name 201
no-reg
label 201
voice register dn 3
number 202
allow watch
name 202
no-reg
label 202
voice register pool 1
id mac 0000.0000.0000
number 1 dn 1
presence call-list
dtmf-relay rtp-nte
username 200 password xxxxx
codec g711ulaw
camera
video
voice register pool 2
id mac 0000.0000.0000
type 9951
number 1 dn 2
presence call-list
dtmf-relay sip-notify
username 201 password xxxxx
codec g711ulaw
camera
video
voice register pool 3
id mac 0000.0000.0000
number 1 dn 3
presence call-list
dtmf-relay rtp-nte
username 202 password xxxxx
codec g711ulaw
camera
video
Anyone have a CME with SIP video call working that can help me to debug my problem?
Thanks
Enrico.

Hi William
Thanks for the response.
i have attached screen shots from 2N Admin page.
can you please check and let me know the settings are correct.
door is not opening by dialing 11 during the call or after call.
Regards
shameer

CME SIP phone outside call issue

Dear all,
i have cme version 9.1 on router 2921 with 7962 sccp phones and 3905 sip phone.
when i place outside call ( to pstn) using the below dial peer, call is processed.
when the call is answered by the autoattendent of the called company ( assume i called x company) , i cant press any other numbers using the sip phones.
i mean if i want to press zero for help or internal extension of the x company, these pressed numbered are not recognized by the analog panasonic PBX of the x company.
Sccp phones works well.
Any help please and below is the dial-peer.
dial-peer voice 1003 pots
trunkgroup 1
corlist outgoing CITIES
description CALLING CITIES
destination-pattern 90[1-9]......
forward-digits 8
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 handle-replaces
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface GigabitEthernet0/2.10
bind media source-interface GigabitEthernet0/2.10
registrar server expires max 36000 min 600
voice class codec 5
codec preference 1 g729r8
codec preference 2 g711ulaw
voice register global
mode cme
source-address 10.100.4.20 port 5060
max-dn 200
max-pool 100
load 3905 CP3905.9-2-1-0.loads
authenticate register
timezone 31
date-format D/M/Y
voicemail 177
tftp-path flash:
create profile sync 000473524028932A
conference hardware
voice register dn 1
number 109
allow watch
pickup-call any-group
pickup-group 170
shared-line max-calls 3
voice register pool 1
id mac 6C99.8984.9678
type 3905
number 1 dn 1
template 1
dtmf-relay sip-notify
voice-class codec 5
username SFD1 password SFD1
thanks

Hi Yahsiel,
firstly thanks for help, secondly if you don't mind i want to ask you the below if possible:
1- in my cme, is there a way when i call an internal extension (e.g 110) from an internal phone it rings normally but when i call from outside-->autoattendent answers-->when i press 110 it get transferred to another phone (e.g 111)....????
2- when i call from outside(pstn) to the cme -->when the plar command is directly to the internal extension the caller id appears but when the autoattendent answers and then transfer to the operator (by pressing zero) the caller id appears as unknown number ??????
3- is the 3905 sip phone support 1Gbps when connected to the PC, as after connecting the phones to the PCs the speed decreased up to 100Mbps?? or it is another matter?
(poe switches is cisco SG200)
regards,

Third Party SIP Phone Config

Good afternoon all,
I have a client that I am trying to program a Gai-Tronic Titan Phone on thier Cisco phone system. The system has the following:
Cisco 2851 Revision 53.51
IOS Version 12.4(24)T2
CME Version 7.1
Unity Express 7.0.3
Anything I do in CME is by CLI.
After some trial and error, I was able to get the phone to show up in Unity Express. When picking up the reciever, we get nothing but a fast busy signal. I am sure I am missing something somewhere.
Is there anyone out there that has setup one of these phones that might be able to help out?
Here is the programming I have so far for the Titan Phone:
ephone-dn 37 dual-line
number 700 secondary 1234567890
label Titan Phone
description Titan Phone
name Titan
call-forward busy 1000
call-forward noan 1000 timeout 15
hold-alert 30 originator
ephone 70
mac-address 0017.AE01.02AB
username "titan" password ********
Again, any help would be appreciated.
Thanks
Mike

I ended up programming the system based off of your above post. Here is what I added to the conifg....
voice register global
mode cme
source-address 10.170.130.250 port 5060
max-dn 10
max-pool 10
hold-alert
voice register dn 1
number 700
name titan password *******
no-reg
label Titan
voice register pool 1
id mac 0017.AE01.02AB
type 7912
number 1 dn 1
dtmf-relay sip-notify
codec g711ulaw
The problem is, when I do a sh voice register pool 1, it doesn't show any phones registered. Am I missing something? I am not currently on-site. I am going to have someone at the facility check the phone tomorrow morning first thing to see if it's working. If there is something in the code I am missing, please let me know.
Thanks
Mike

CUCM not communicating with SIP Gateway

This is my lab environment:
VOIP.MS --> 2811 CUBE--> CUCM 10.5.2
Attached is my configuration on both my cube and publisher
What is going on is if I make a call from my IP Communicator phone it does not call out. Also if I do a debug ccsip all, I get no output from the calls I make going out. I have not even attempted the incoming since I believe that my cucm server and gateway are not even communicating with each other. Please let me know if I am doing something wrong.
Thanks

Results Summary
Calling Party Information
Calling Party = 462
Partition = 10Digit:7Digit:Internal:LongDistance
Device CSS =
Line CSS = CSSLongDistance
AAR Group Name =
AAR CSS =
Dialed Digits = 918042221111
Match Result = RouteThisPattern
Matched Pattern Information
Pattern = 9.1[2-9]XXXXXXXXX
Partition = LongDistance
Time Schedule =
Called Party Number = 18042221111
Time Zone = Etc/GMT
End Device = WANLIST
Call Classification = OffNet
InterDigit Timeout = NO
Device Override = Disabled
Outside Dial Tone = NO
Call Flow
Route Pattern :Pattern= 9.1[2-9]XXXXXXXXX
Positional Match List = 18042221111
DialPlan =
Route Filter
Filter Name =
Filter Clause =
Require Forced Authorization Code = No
Authorization Level = 0
Require Client Matter Code = No
Call Classification =
PreTransform Calling Party Number = 462
PreTransform Called Party Number = 918042221111
Calling Party Transformations
External Phone Number Mask = NO
Calling Party Mask =
Prefix =
CallingLineId Presentation = Default
CallingName Presentation = Default
Calling Party Number = 462
ConnectedParty Transformations
ConnectedLineId Presentation = Default
ConnectedName Presentation = Default
Called Party Transformations
Called Party Mask =
Discard Digits Instruction = PreDot
Prefix =
Called Number = 18042221111
Route List :Route List Name= WANLIST
RouteGroup :RouteGroup Name= WANGROUP
PreTransform Calling Party Number = 462
PreTransform Called Party Number = 918042221111
Calling Party Transformations
External Phone Number Mask = Default
Calling Party Mask =
Prefix =
Calling Party Number = 462
Called Party Transformations
Called Party Mask =
Discard Digits Instructions =
Prefix =
Called Number = 918042221111
Device :Type= SIPTrunk
End Device Name = SIP2800
PortNumber = 0
Device Status = UnKnown
AAR Group Name =
AAR Calling Search Space =
AAR Prefix Digits =
Call Classification = Use System Default
Calling Party Selection = Originator
CallingLineId Presentation = Default
CallerID DN =
Alternate Matches
Note: Information Not Available
I also adjusted my dial-peer to be:
dial-peer voice 10 voip
destination-pattern 1[2-9].........
session protocol sipv2
session target dns:newyork4.voip.ms
voice-class codec 1
dtmf-relay sip-notify rtp-nte
I also verified I can ping newyork4.voip.ms

Dtmf-relay sip-notify

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