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Beginner

CISCO IP TELEPHONY CME 7821 PHONES XML PARSING ERROR

Dear All,

                   my cme IP address is 10.0.40.1 and all phones are registered and working fine. To provide telephone lines to another partner company I configured another lan IP 10.10.20.1 on another interface and configure dhcp in that range . Now the sccp phones are connected and working well, but the 7821 sip phones are showing xml parsing error and when I checked the debug messages , I can see  connection  "unauthorized" for the 7821 phones . So trusted the subnet in trusted list, but still the same error exist. Can anybody suggest me a solution

Sameer 

6 REPLIES 6
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VIP Collaborator

Please try this

https://supportforums.cisco.com/discussion/12747636/phone-reg-issue-cme-105

Phones in the same LAN of the CME. If yes then remove then apply these commands and restart the phones

voice service voip
 sip
  registrar server expires max 60 min 3600
voice register global
no authenticate register
no authenticate realm all
 create profile 

If the phones are in remote LAN, then apply these commands and restart the phones

voice service voip
 sip
  registrar server expires max 60 min 3600
voice register pool  1
username phone1 password phone1
!
voice register pool  2
username phone2 password phone2
!
voice register pool  3
username phone3 password phone3
voice register global
no authenticate realm all
create profile
Highlighted
VIP Advisor

Please post your config to see where is the problem.

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Dear Baqari , 

   Now the phone is registered in the cme status , but physically phone is hanging in registering and incoming call ringing , but cannot dial any extension and there is no dial tone . I tried by giving as per comment given by CISCO RATH , but same 

voice service voip
ip address trusted list
ipv4 10.0.40.0
ipv4 10.0.40.0 255.255.254.0
ipv4 10.10.11.0
ipv4 10.10.11.0 255.255.255.0
ipv4 192.168.50.0
ipv4 192.168.50.0 255.255.255.0
ipv4 10.10.20.0 255.255.255.0
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 sip moved-temporarily
no supplementary-service sip refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server expires max 86400 min 3600
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
!
voice class codec 10
codec preference 1 g711ulaw
codec preference 2 g729r8
!
voice class codec 3
codec preference 1 g711ulaw
codec preference 2 g711alaw
!
!
!
voice class custom-cptone jointone
dualtone conference
frequency 600 900
cadence 300 150 300 100 300 50
!
voice class custom-cptone leavetone
dualtone conference
frequency 400 800
cadence 400 50 200 50 200 50
!
!
voice register global
mode cme
source-address 10.0.40.1 port 5060
max-dn 200
max-pool 165
olsontimezone Asia/Bahrain version 2012d
timezone 31
time-format 24
date-format D/M/Y
hold-alert
voicemail 9751
tftp-path flash:
file text
create profile sync 0376432345128429
ntp-server 10.0.40.1 mode unicast
conference hardware
camera

interface GigabitEthernet0/0
ip address 10.0.40.1 255.255.254.0
ip virtual-reassembly in
duplex auto
speed auto

!
interface GigabitEthernet0/2
ip address 10.10.20.1 255.255.255.0 DHCP CONFIGURED FOR REMOTE NETWORK ON ANOTHER SWITCH
ip virtual-reassembly in
duplex auto
speed auto


telephony-service
sdspfarm conference mute-on 111 mute-off 222
sdspfarm units 4
sdspfarm transcode sessions 30
sdspfarm tag 1 TRANSCODE
sdspfarm tag 2 CONFPROF1
conference hardware
conference transfer-pattern
video
no auto-reg-ephone
authentication credential xxxx xxxx
xml user xxxx password xxx 15
max-ephones 105
max-dn 105
ip source-address 10.0.40.1 port 2000

 

Highlighted

When there is no dial tone, can you share the output of the command show voice register poo all brief.

Seems that the phone isn't registered and thats the reason you aren't getting dial tone.

Is this the case for one phone or all phones.

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Dear Mohammed,

          When I check 

 sh voice register pool telephone number 9882  I get the output as registered , but physically phones in registering mode . 

All the sccp phones are connected in the same lan 

All other sccp and sip phones in my ip address 10.0.40.1 255.255.254.0  lan working perfectly 

  Now am away from site , so I cannot paste the output 

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In my case, setting source interface to be loopback solved SIP phone registration issue.

I do not know why, but there will be a bug.