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

gossa1973
Level 1
Level 1

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>

19 Replies 19

gmgarrian
Level 4
Level 4

Is that the entire CME config?  I'm not seeing any telephony configuration.

Unless things have changed drastically since 8.0, you need to specify what load to use for each phone type.

Example:

!
telephony-service
 no auto-reg-ephone
 load 7960-7940 P00307020300
 max-ephones 144
 max-dn 500
 ip source-address 172.22.1.107 port 2000
 max-redirect 15
 service phone videoCapability 1
 dialplan-pattern 1 5123781291 extension-length 4
 voicemail 2000
 max-conferences 8 gain -6
 transfer-system full-consult
 secondary-dialtone 9

Config example:

http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_configuration_example09186a008037f2a9.shtml

Compatibility matrix:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/requirements/guide/cme85spc.htm.html

I left that part of configuration, I was thinking that was not important for registering SIP phones

here is it:

telephony-service
no auto-reg-ephone

max-ephones 24
max-dn 72
ip source-address 192.168.5.251 port 2000
calling-number initiator
service phone videoCapability 1
timeouts interdigit 2
timeouts busy 5
system message ELTOSAN
cnf-file location flash:
load 7906 SCCP11.8-4-2S.loads
time-zone 26
time-format 24
date-format dd-mm-yy
voicemail 555

mwi relay

max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
moh music-on-hold.au
web admin system name admin password xxxxxx
dn-webedit
time-webedit
transfer-system full-consult dss
transfer-pattern 9.T
create cnf-files version-stamp Jan 01 2002 00:00:00

You need to add a load statement for the 9971 phones.  You have one for the 7906 model but you need to remove the .loads at the end so it reads:

load 7906 SCCP11.8-4-2S

You'll need to add:

load 9971 sip9971.9-1-1SR1

Once you go into telephony config mode type "load ?" and it will list all the supported phone types.

Edit:  You may want to do a no cnf files/create cnf files once you make the change.

I do not have in "Telephony Service" loads for 9971 phone because phone is SIP not SCCP  phone, in "Voice register global" i have option for loads for 9971 phones.

My bad.  I haven't used SIP phone on CME so I assumed (bad idea!) the config was the same.  I did some research and it looks like the same rule applies and you need to remove the .loads from your load statement uner voice register global.

Also, is there are reason all your tftp entries have an alias listed with the same name?  You may want to clean that up to make sure.  Other than that I'm not sure.

I remove loads from "Voice Register Global " , same problem , I also try with reseting to factory setting phone. Same issue .

Is it possible to have some kind debug on the phone?

Which files phone need to download from TFTP if I have load statment ?

I am confused. When I reset phone to factory and I look in the tftp debug i can see that only three files are downloaded.

*Aug 20 08:06:11.055: TFTP: Looking for CTLSEP04C5A4B03B0D.tlv

*Aug 20 08:06:11.159: TFTP: Looking for ITLSEP04C5A4B03B0D.tlv

*Aug 20 08:06:11.255: TFTP: Looking for ITLFile.tlv

*Aug 20 08:06:11.395: TFTP: Looking for SEP04C5A4B03B0D.cnf.xml

*Aug 20 08:06:11.399: TFTP: Opened flash:/SEP04C5A4B03B0D.cnf.xml, fd 14, size 4579 for process 93

*Aug 20 08:06:11.463: TFTP: Finished flash:/SEP04C5A4B03B0D.cnf.xml, time 00:00:00 for process 93v

*Aug 20 08:06:14.771: TFTP: Looking for English_United_States/gd-sip.jar

*Aug 20 08:06:14.779: TFTP: Opened flash0:English_United_States/gd-sip.jar, fd 14, size 75616 for process 93

*Aug 20 08:06:15.159: TFTP: Finished flash0:English_United_States/gd-sip.jar, time 00:00:00 for process 93

*Aug 20 08:06:17.995: TFTP: Looking for United_States/g4-tones.xml

*Aug 20 08:06:18.003: TFTP: Opened flash0:United_States/g4-tones.xml, fd 14, size 1915 for process 93

*Aug 20 08:06:18.015: TFTP: Finished flash0:United_States/g4-tones.xml, time 00:00:00 for process 93

None of the *.loads and *.sebn files was requested by the phone. I presume that phone see in configurationfile  that 9.1.1 firmware is loaded on CME and compare with his firmware version, if are the same version, phone does not try to download them.

Do you have any idea how to generete this TLV files

Today I will try to two things.

first I wil try newes firmware 9.2.1 and debug whole process

second I will convert one of the 7906 to SIP protocol to see does it work with 7906 over SIP..

If you have any idea , I will be very grateful.

10x in advanced.

I'm not sure on this one.  Those .tlv files seem to have the phone name in them (SEP04C5A4B03B0D) so I'm assuming it's looking for specific config info.

Donwload and install new locale. This issue is related with the process:

CME SIP configured -> CME SIP removed -> CME SIP configured again

copy tftp://x.x.x.x/CME-locale-xx_XX-Xxxxxx.tar flash:/its/

I have the same Problem. Reinstalling the same locale as bevor? My 9951 dont Register after adding a sip-ua trunk.

HELP!

Your configuration is correct.

You do not need to have telephony-service for SIP phones as it is used for SCCP Phones only.

You need to use voice register global instead which is the SIP subsysted for CME.

Try binding SIP to correct interface for Voice VLAN and take debug ccsip messages. You should see SIP Register message from the phone.

Please provide the debug output.

--

Udit

One stupid question . How to bind SIP to interface. Is it command in "voice register global"  "source-address 192.168.5.251 port 5060"

I try to debug ccsip messages but i do not receive any message for phone 9971 , I receive only messeges for XLITE client on data VLAN (2). I tryed to convert one 7906 from SCCP to SIP and now i get similar behvor with message on display "unprovisioned". I think that something is wrong in my configuration but i do not know what is it.

probably you are right i did not bin SIP to corect VLAN 5 interface but i do not know how I did it for data VLAN 2 interface becouse XLITE work over SIP

please help me

10x in advanced

Use the following command to bind SIP to an interfaace -

!

voice service voip

sip

  bind all source-interface

!

HTH

Udit

Binding SIP to interface did not help , but I solved problem.. I start from scratch,  I reset to factory router , then manualy configure whole CME. After adding phone 9971, phone register without problem . I compare configurations . I  saw my mistake .

in first configuration i did not configure line 1

voice register pool 3

id mac 04C5.A4B0.3B0D

type 9971

number 3 dn 3 <- mistake

second time i configure

voice register pool 3

id mac 04C5.A4B0.3B0D

type 9971

number 1 dn 3

one stupid mistake

10x any way

It's great you got it working!

You helped me learn a bit about SIP in an CME environment.

Thanks!

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