German menu language on 7960G SIP Phone
Hi All
I have flashed a 7960G SCCP Phone to SIP Firmware.
Is there any possibility to give this phone another menu language, like german?
Or is this running only on SCCP Firmware?
Thank you
Hi,
thank you for the help.
Now I have these files in my TFTP Server.
These are my files:
OS79XX.TXT
P0S3-08-12-00
SEP0014A8924D6D.CNF.XML
<device>
<devicePool>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
</ports>
<processNodeName> </processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<versionStamp>{Jan 01 2005 00:00:00}</versionStamp>
<loadInformation>P0S3-08-12-00</loadInformation>
<userLocale>
<name>German_Germany</name>
<langCode>de</langCode>
</userLocale>
<networkLocale>Germany</networkLocale>
<idleTimeout>0</idleTimeout>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL></servicesURL>
</device>
SIP0014A8924D6D.cnf
# SIP Configuration Generic File
# Line 1 appearance
line1_name: 01010101001
# Line 1 Registration Authentication
line1_authname: "UNPROVISIONED"
# Line 1 Registration Password
line1_password: "UNPROVISIONED"
# Line 2 appearance
line2_name: football
# Line 2 Registration Authentication
line2_authname: "UNPROVISIONED"
# Line 2 Registration Password
line2_password: "UNPROVISIONED"
####### New Parameters added in Release 2.0 #######
# All user_parameters have been removed
# Phone Label (Text desired to be displayed in upper right corner)
phone_label: "" ; Has no effect on SIP messaging
# Line 1 Display Name (Display name to use for SIP messaging)
line1_displayname: "User ID"
# Line 2 Display Name (Display name to use for SIP messaging)
line2_displayname: ""
####### New Parameters added in Release 3.0 ######
# Phone Prompt (The prompt that will be displayed on console and telnet)
phone_prompt: "SIP Phone" ; Limited to 15 characters (Default - SIP Phone)
# Phone Password (Password to be used for console or telnet login)
phone_password: "cisco" ; Limited to 31 characters (Default - cisco)
# User classifcation used when Registering [ none(default), phone, ip ]
user_info: none
SIPDefault.cnf
# SIP Default Generic Configuration File
# Image Version
image_version: P0S381200
language: german
# Proxy Server
proxy1_address: "" ; Can be dotted IP or FQDN
proxy2_address: "" ; Can be dotted IP or FQDN
proxy3_address: "" ; Can be dotted IP or FQDN
proxy4_address: "" ; Can be dotted IP or FQDN
proxy5_address: "" ; Can be dotted IP or FQDN
proxy6_address: "" ; Can be dotted IP or FQDN
# Proxy Server Port (default - 5060)
proxy1_port: 5060
proxy2_port: 5060
proxy3_port: 5060
proxy4_port: 5060
proxy5_port: 5060
proxy6_port: 5060
# Proxy Registration (0-disable (default), 1-enable)
proxy_register: 0
# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: 3600
# Codec for media stream (g711ulaw (default), g711alaw, g729a)
preferred_codec: g711ulaw
# TOS bits in media stream [0-5] (Default - 5)
tos_media: 5
# Inband DTMF Settings (0-disable, 1-enable (default))
dtmf_inband: 1
# Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt )
dtmf_outofband: avt
# DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up)
dtmf_db_level: 3
# SIP Timers
timer_t1: 500 ; Default 500 msec
timer_t2: 4000 ; Default 4 sec
sip_retx: 10 ; Default 10
sip_invite_retx: 6 ; Default 6
timer_invite_expires: 180 ; Default 180 sec
####### New Parameters added in Release 2.0 #######
# Dialplan template (.xml format file relative to the TFTP root directory)
dial_template: dialplan
# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: "" ; Example: ./sip_phone/
# Time Server (There are multiple values and configurations refer to Admin Guide for Specifics)
sntp_server: "" ; SNTP Server IP Address
sntp_mode: directedbroadcast ; unicast, multicast, anycast, or directedbroadcast (default)
time_zone: EST ; Time Zone Phone is in
dst_offset: 1 ; Offset from Phone's time when DST is in effect
dst_start_month: April ; Month in which DST starts
dst_start_day: "" ; Day of month in which DST starts
dst_start_day_of_week: Sun ; Day of week in which DST starts
dst_start_week_of_month: 1 ; Week of month in which DST starts
dst_start_time: 02 ; Time of day in which DST starts
dst_stop_month: Oct ; Month in which DST stops
dst_stop_day: "" ; Day of month in which DST stops
dst_stop_day_of_week: Sunday ; Day of week in which DST stops
dst_stop_week_of_month: 8 ; Week of month in which DST stops 8=last week of month
dst_stop_time: 2 ; Time of day in which DST stops
dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) DST automatic adjustment
time_format_24hr: 1 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr)
# Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no user control)
dnd_control: 0 ; Default 0 (Do Not Disturb feature is off)
# Caller ID Blocking (0-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
callerid_blocking: 0 ; Default 0 (Disable sending all calls as anonymous)
# Anonymous Call Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
anonymous_call_block: 0 ; Default 0 (Disable blocking of anonymous calls)
# DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127)
dtmf_avt_payload: 101 ; Default 101
# Sync value of the phone used for remote reset
sync: 1 ; Default 1
####### New Parameters added in Release 2.1 #######
# Backup Proxy Support
proxy_backup: "" ; Dotted IP of Backup Proxy
proxy_backup_port: 5060 ; Backup Proxy port (default is 5060)
# Emergency Proxy Support
proxy_emergency: "" ; Dotted IP of Emergency Proxy
proxy_emergency_port: 5060 ; Emergency Proxy port (default is 5060)
# Configurable VAD option
enable_vad: 0 ; VAD setting 0-disable (Default), 1-enable
####### New Parameters added in Release 2.2 ######
# NAT/Firewall Traversal
nat_enable: 0 ; 0-Disabled (default), 1-Enabled
nat_address: "" ; WAN IP address of NAT box (dotted IP or DNS A record only)
voip_control_port: 5060 ; UDP port used for SIP messages (default - 5060)
start_media_port: 16384 ; Start RTP range for media (default - 16384)
end_media_port: 32766 ; End RTP range for media (default - 32766)
nat_received_processing: 0 ; 0-Disabled (default), 1-Enabled
# Outbound Proxy Support
outbound_proxy: "" ; restricted to dotted IP or DNS A record only
outbound_proxy_port: 5060 ; default is 5060
####### New Parameter added in Release 3.0 #######
# Allow for the bridge on a 3way call to join remaining parties upon hangup
cnf_join_enable : 1 ; 0-Disabled, 1-Enabled (default)
####### New Parameters added in Release 3.1 #######
# Allow Transfer to be completed while target phone is still ringing
semi_attended_transfer: 1 ; 0-Disabled, 1-Enabled (default)
# Telnet Level (enable or disable the ability to telnet into the phone)
telnet_level: 1 ; 0-Disabled (default), 1-Enabled, 2-Privileged
####### New Parameters added in Release 4.0 #######
# XML URLs
services_url: "" ; URL for external Phone Services
directory_url: "" ; URL for external Directory location
logo_url: "" ; URL for branding logo to be used on phone display
# HTTP Proxy Support
http_proxy_addr: "" ; Address of HTTP Proxy server
http_proxy_port: 80 ; Port of HTTP Proxy Server (80-default)
# Dynamic DNS/TFTP Support
dyn_dns_addr_1: "" ; restricted to dotted IP
dyn_dns_addr_2: "" ; restricted to dotted IP
dyn_tftp_addr: "" ; restricted to dotted IP
# Remote Party ID
remote_party_id: 0 ; 0-Disabled (default), 1-Enabled
####### New Parameters added in Release 4.4 #######
# Call Hold Ringback (0-off, 1-on, 2-off with no user control, 3-on with no user control)
call_hold_ringback: 0 ; Default 0 (Call Hold Ringback feature is off)
####### New Parameters added in Release 6.0 #######
# Dialtone Stutter for MWI
stutter_msg_waiting: 0 ; 0-Disabled (default), 1-Enabled
# RTP Call Statistics (SIP BYE/200 OK message exchange)
call_stats: 0 ; 0-Disabled (default), 1-Enabled
xmlDefault.CNF.XML
<?xml version="1.0"?>
-<Default>
-<callManagerGroup>
-<members>
-<member priority="0">
-<callManager>
-<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
</ports>
<processNodeName/>
</callManager>
</member>
</members>
<loadInformation6 model="IP Phone 7910"/>
<loadInformation124 model="Addon 7914"/>
<loadInformation9 model="IP Phone 7935"/>
<loadInformation8 model="IP Phone 7940"/>
<loadInformation7 model="IP Phone 7960">P0S3-8-12-00</loadInformation7>
<loadInformation20000 model="IP Phone 7905"/>
<loadInformation30008 model="IP Phone 7902"/>
<loadInformation30002 model="IP Phone 7920"/>
<loadInformation30019 model="IP Phone 7936"/>
<loadInformation30006 model="IP Phone 7970"/>
<loadInformation30018 model="IP Phone 7961"/>
<loadInformation30007 model="IP Phone 7912"/>
</callManagerGroup>
</Default>
The folders "German_Germany"; "germany" and the file German_Germany.aar are in TFTP folder,too. But my phone is doing nothing with this files.
This is the log of TFTP Server:
Rcvd DHCP inform Msg for IP 192.168.0.9, Mac 02:70:7E:7F:09:01 [04/03 13:21:54.559]
Rcvd DHCP inform Msg for IP 192.168.0.9, Mac 02:70:7E:7F:09:01 [04/03 13:21:58.562]
Rcvd DHCP Rqst Msg for IP 0.0.0.0, Mac 00:14:A8:92:4D:6D [04/03 13:22:45.892]
Previously allocated address 192.168.0.6 acked [04/03 13:22:45.893]
Connection received from 192.168.0.6 on port 50798 [04/03 13:22:46.037]
Read request for file <CTLSEP0014A8924D6D.tlv>. Mode octet [04/03 13:22:46.038]
File <CTLSEP0014A8924D6D.tlv> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:22:46.038]
Connection received from 192.168.0.6 on port 50798 [04/03 13:22:47.036]
Read request for file <CTLSEP0014A8924D6D.tlv>. Mode octet [04/03 13:22:47.036]
File <CTLSEP0014A8924D6D.tlv> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:22:47.036]
Connection received from 192.168.0.6 on port 50798 [04/03 13:22:51.035]
Read request for file <CTLSEP0014A8924D6D.tlv>. Mode octet [04/03 13:22:51.036]
File <CTLSEP0014A8924D6D.tlv> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:22:51.036]
Connection received from 192.168.0.6 on port 50799 [04/03 13:22:51.056]
Read request for file <SEP0014A8924D6D.cnf.xml>. Mode octet [04/03 13:22:51.057]
File <SEP0014A8924D6D.cnf.xml> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:22:51.057]
Connection received from 192.168.0.6 on port 50800 [04/03 13:22:51.085]
Read request for file <SIP0014A8924D6D.cnf>. Mode octet [04/03 13:22:51.086]
File <SIP0014A8924D6D.cnf> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:22:51.086]
Connection received from 192.168.0.6 on port 50801 [04/03 13:22:51.105]
Read request for file <MGC0014A8924D6D.cnf>. Mode octet [04/03 13:22:51.105]
File <MGC0014A8924D6D.cnf> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:22:51.105]
Connection received from 192.168.0.6 on port 50802 [04/03 13:22:51.124]
Read request for file <XMLDefault.cnf.xml>. Mode octet [04/03 13:22:51.127]
Using local port 65426 [04/03 13:22:51.127]
<XMLDefault.cnf.xml>: sent 3 blks, 1077 bytes in 0 s. 0 blk resent [04/03 13:22:51.135]
Connection received from 192.168.0.6 on port 50803 [04/03 13:22:51.180]
Read request for file <P0S3-8-12-00.loads>. Mode octet [04/03 13:22:51.180]
Using local port 65427 [04/03 13:22:51.180]
<P0S3-8-12-00.loads>: sent 1 blk, 458 bytes in 0 s. 0 blk resent [04/03 13:22:51.183]
Rcvd DHCP inform Msg for IP 192.168.0.9, Mac 02:70:7E:7F:09:01 [04/03 13:24:17.151]
Rcvd DHCP inform Msg for IP 192.168.0.9, Mac 02:70:7E:7F:09:01 [04/03 13:24:20.150]
Rcvd DHCP Rqst Msg for IP 0.0.0.0, Mac 00:14:A8:92:4D:6D [04/03 13:24:34.219]
Previously allocated address 192.168.0.6 acked [04/03 13:24:34.220]
Connection received from 192.168.0.6 on port 50823 [04/03 13:24:34.258]
Read request for file <SIPDefault.cnf>. Mode octet [04/03 13:24:34.259]
Using local port 64133 [04/03 13:24:34.259]
Connection received from 192.168.0.6 on port 50823 [04/03 13:24:35.250]
Read request for file <SIPDefault.cnf>. Mode octet [04/03 13:24:35.250]
Using local port 64134 [04/03 13:24:35.250]
Connection received from 192.168.0.6 on port 50823 [04/03 13:24:39.250]
Read request for file <SIPDefault.cnf>. Mode octet [04/03 13:24:39.250]
Using local port 64135 [04/03 13:24:39.250]
<SIPDefault.cnf>: sent 13 blks, 6203 bytes in 0 s. 0 blk resent [04/03 13:24:39.284]
Connection received from 192.168.0.6 on port 50824 [04/03 13:24:39.485]
Read request for file <SIP0014A8924D6D.cnf>. Mode octet [04/03 13:24:39.485]
File <SIP0014A8924D6D.cnf> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:24:39.485]
TIMEOUT waiting for Ack block #1 [04/03 13:24:49.260]
TIMEOUT waiting for Ack block #1 [04/03 13:24:50.251]
Rcvd DHCP Rqst Msg for IP 0.0.0.0, Mac 00:14:A8:92:4D:6D [04/03 13:26:50.610]
Previously allocated address 192.168.0.6 acked [04/03 13:26:50.611]
Connection received from 192.168.0.6 on port 50857 [04/03 13:26:50.649]
Read request for file <SIPDefault.cnf>. Mode octet [04/03 13:26:50.649]
Using local port 64137 [04/03 13:26:50.649]
Connection received from 192.168.0.6 on port 50857 [04/03 13:26:51.641]
Read request for file <SIPDefault.cnf>. Mode octet [04/03 13:26:51.642]
Using local port 64138 [04/03 13:26:51.642]
Connection received from 192.168.0.6 on port 50857 [04/03 13:26:55.641]
Read request for file <SIPDefault.cnf>. Mode octet [04/03 13:26:55.641]
Using local port 64139 [04/03 13:26:55.642]
<SIPDefault.cnf>: sent 13 blks, 6203 bytes in 0 s. 0 blk resent [04/03 13:26:55.676]
Connection received from 192.168.0.6 on port 50858 [04/03 13:26:55.874]
Read request for file <SIP0014A8924D6D.cnf>. Mode octet [04/03 13:26:55.875]
File <SIP0014A8924D6D.cnf> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:26:55.875]
TIMEOUT waiting for Ack block #1 [04/03 13:27:05.651]
TIMEOUT waiting for Ack block #1 [04/03 13:27:06.645]
Rcvd DHCP Rqst Msg for IP 0.0.0.0, Mac 00:14:A8:92:4D:6D [04/03 13:28:40.670]
Previously allocated address 192.168.0.6 acked [04/03 13:28:40.671]
Connection received from 192.168.0.6 on port 50797 [04/03 13:28:40.813]
Read request for file <CTLSEP0014A8924D6D.tlv>. Mode octet [04/03 13:28:40.813]
File <CTLSEP0014A8924D6D.tlv> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:28:40.814]
Connection received from 192.168.0.6 on port 50797 [04/03 13:28:41.803]
Read request for file <CTLSEP0014A8924D6D.tlv>. Mode octet [04/03 13:28:41.804]
File <CTLSEP0014A8924D6D.tlv> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:28:41.804]
Connection received from 192.168.0.6 on port 50797 [04/03 13:28:45.803]
Read request for file <CTLSEP0014A8924D6D.tlv>. Mode octet [04/03 13:28:45.803]
File <CTLSEP0014A8924D6D.tlv> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:28:45.804]
Connection received from 192.168.0.6 on port 50798 [04/03 13:28:45.824]
Read request for file <SEP0014A8924D6D.cnf.xml>. Mode octet [04/03 13:28:45.824]
File <SEP0014A8924D6D.cnf.xml> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:28:45.824]
Connection received from 192.168.0.6 on port 50799 [04/03 13:28:45.853]
Read request for file <SIP0014A8924D6D.cnf>. Mode octet [04/03 13:28:45.853]
File <SIP0014A8924D6D.cnf> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:28:45.853]
Connection received from 192.168.0.6 on port 50800 [04/03 13:28:45.874]
Read request for file <MGC0014A8924D6D.cnf>. Mode octet [04/03 13:28:45.876]
File <MGC0014A8924D6D.cnf> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:28:45.876]
Connection received from 192.168.0.6 on port 50801 [04/03 13:28:45.895]
Read request for file <XMLDefault.cnf.xml>. Mode octet [04/03 13:28:45.895]
Using local port 64147 [04/03 13:28:45.895]
<XMLDefault.cnf.xml>: sent 3 blks, 1077 bytes in 0 s. 0 blk resent [04/03 13:28:45.898]
Connection received from 192.168.0.6 on port 50802 [04/03 13:28:45.935]
Read request for file <P0S3-8-12-00.loads>. Mode octet [04/03 13:28:45.936]
Using local port 64148 [04/03 13:28:45.936]
<P0S3-8-12-00.loads>: sent 1 blk, 458 bytes in 0 s. 0 blk resent [04/03 13:28:45.937]
Rcvd DHCP Rqst Msg for IP 0.0.0.0, Mac 00:14:A8:92:4D:6D [04/03 13:30:28.597]
Previously allocated address 192.168.0.6 acked [04/03 13:30:28.597]
Connection received from 192.168.0.6 on port 50787 [04/03 13:30:28.636]
Read request for file <SIPDefault.cnf>. Mode octet [04/03 13:30:28.636]
Using local port 64149 [04/03 13:30:28.636]
Connection received from 192.168.0.6 on port 50787 [04/03 13:30:29.627]
Read request for file <SIPDefault.cnf>. Mode octet [04/03 13:30:29.628]
Using local port 64150 [04/03 13:30:29.628]
Connection received from 192.168.0.6 on port 50787 [04/03 13:30:33.627]
Read request for file <SIPDefault.cnf>. Mode octet [04/03 13:30:33.628]
Using local port 64151 [04/03 13:30:33.628]
<SIPDefault.cnf>: sent 13 blks, 6203 bytes in 0 s. 0 blk resent [04/03 13:30:33.658]
Connection received from 192.168.0.6 on port 50788 [04/03 13:30:33.856]
Read request for file <SIP0014A8924D6D.cnf>. Mode octet [04/03 13:30:33.856]
File <SIP0014A8924D6D.cnf> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:30:33.857]
Rcvd DHCP inform Msg for IP 192.168.0.9, Mac 02:70:7E:7F:09:01 [04/03 13:30:41.756]
TIMEOUT waiting for Ack block #1 [04/03 13:30:43.637]
TIMEOUT waiting for Ack block #1 [04/03 13:30:44.629]
Rcvd DHCP inform Msg for IP 192.168.0.9, Mac 02:70:7E:7F:09:01 [04/03 13:30:44.759]
Rcvd DHCP inform Msg for IP 192.168.0.9, Mac 02:70:7E:7F:09:01 [04/03 13:32:11.253]
Rcvd DHCP inform Msg for IP 192.168.0.9, Mac 02:70:7E:7F:09:01 [04/03 13:32:14.253]
Rcvd DHCP Rqst Msg for IP 0.0.0.0, Mac 00:14:A8:92:4D:6D [04/03 13:33:50.655]
Previously allocated address 192.168.0.6 acked [04/03 13:33:50.656]
Connection received from 192.168.0.6 on port 50795 [04/03 13:33:50.696]
Read request for file <SIPDefault.cnf>. Mode octet [04/03 13:33:50.696]
Using local port 62124 [04/03 13:33:50.696]
Connection received from 192.168.0.6 on port 50795 [04/03 13:33:51.685]
Read request for file <SIPDefault.cnf>. Mode octet [04/03 13:33:51.686]
Using local port 62125 [04/03 13:33:51.686]
Connection received from 192.168.0.6 on port 50795 [04/03 13:33:55.685]
Read request for file <SIPDefault.cnf>. Mode octet [04/03 13:33:55.685]
Using local port 62126 [04/03 13:33:55.685]
<SIPDefault.cnf>: sent 13 blks, 6203 bytes in 0 s. 0 blk resent [04/03 13:33:55.723]
Connection received from 192.168.0.6 on port 50796 [04/03 13:33:55.915]
Read request for file <SIP0014A8924D6D.cnf>. Mode octet [04/03 13:33:55.915]
File <SIP0014A8924D6D.cnf> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:33:55.915]
TIMEOUT waiting for Ack block #1 [04/03 13:34:05.698]
TIMEOUT waiting for Ack block #1 [04/03 13:34:06.691]
Is there anybody, who can help me with this problem.
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Is it really so hard to understand, that an ULTIMATE BUSINESS SOLUTION as E90 definitely would be preferred by the people, who really use it - really intensive - as a communication device?! Is it really hard to understand, that thousands of them would get e-mails and documents while on the road that definitely need to be urgent answered some other language than 3 preinstalled?! Should I change the firmware of my phone every time I need to answer e-mail in French or edit the financial contract for my Russian partner?!
Ridiculous and disappointing - what else should I say?! Especially because I know, that the situation as easy to fix from the Nokia side as a single mouse click.
I am sure there are people reading and writing to this Forum who needs for example Chinese Mandarin and German, or Hindi and Farsi, or Arabic and Spanish simultaniously. There is no way to understand why it is impossible to choose the INPUT languages - I do NOT speak about the MENU languages and never pretend to.
So, because this Forum is Nokia's piece of land and Nokia has kindly allowed us to cry out here what I do really appreciate, I want definitely get the answer to following questions:
1) Is it possible to add some INPUT (once again - NOT menu, just INPUT) languages to E90 firmware?
2) If yes, when?
3) If not, what is the reason? The reason - please, no "technical excuses", because I am not that stupid, OK? ;-)
We are living in such a damn globalized world, aren't we? So let's be really globalized and let's help us ourselves to get out the most from this.
Thanks to all for listening.Since Windows 98 is absolutely not a problem to add ANY input language in ANY localized version of the OS. I do know that Symbian OS has at least the same flexibility options.
Once again: I do not pretend to have any other MENU language - I simply do not need it. But the possibility to choose more INPUT language is simply and quickly to add, that is what I am asking for.
Maybe your statistic research is right, I do not want to argue with you. More than that, this is exactly I expected as an excuse option: "Marginal request". But no, it is not. The device would probably have higher customer interest when the free input would be represented.
Your advice to use laptop instead of E90 is highly appreciated, but it is not unique enough to be accepted "as is". ;-) The idea of E90 as well as previous communicators was to give the customer an option to leave her/his laptop at home and make a 1 or 2 day business trip as lite as possible in every meaning of the word "lite". I have 2 laptops already, 1 Panasonic for traveling and IBM Z61p Workstation as a desktop replacement. I should say that Panasonic, while being one of the best travel-oriented devices on the market, with its almost 2 Kilo travel weight and about 1 minute power-up/power-down time is completely insufficient as a communicator replacement. In addition to it you need a businesscard-scanner and a digicam, you are constantly out of power and so on. It is or will be the same story with UMPC. The only option until now was HTC/Qtek 9000, the most powerful and well-equipped device on the market. But it is completely useless as a phone and its camera is a disaster, so I switched to E90 because of its awesome usability as a phone, a digicam and as a communicator simultaneously just because I HATE to carry on 12 hours a day here and there a 5 Kilo bag with laptop, phone, camera, cables, batteries and other damned stuff around. I am just SICK and TIRED of this and I dreamed about the device that can make free from all that - for YEARS!!!
I remember my conversation with Nokia people at its CeBIT Stand in late 1990-es - the concept was already here, but the materialization of my dreams took Nokia R&D almost a decade. And now, when it is already in my hands, it lacks such a damn little thing to be as perfect as possible! ;-)
I am not that crazy - "perfect" does not mean really perfect. Of course there are and there would be minor and major bugs with later fixes and improvements. Because of the possibility of fixes and improvements I insist on one of the most critical for me. I understand as well, that it needs neither additional research nor hardware changes. All you need to to do is a little software module to be added and/or replaced. The Occam Razor Rule is still in my mind. ;-) -
Change Menu Language to English?
Hello! My hubby bought me a Nokia 7373 as a gift - we are currently located in Germany. When I put my sim card in the phone, the menu displayed in German _ is there a way to change the menu language to english??
You are going to have to find your way through the menus (sorry, I speak no Spanish):
-Menu
-Settings (spanner icon)
-Phone (phone and spanner together icon)
-Language settings
-Phone Language
You will probably find it set to 'Automatic', simply choose English from this list. -
Although I installed SQL Developer on an english WinXP the language of the SQL developer tool is german.
Is there an option where I can later (after installation) switch the menu language?
Or do I have to download explicitely another setup package?
From where exactly for english?
PeterHi,
This thread should help you out:
Re: Can I get english interface back?
- Rakesh -
Does anyone know how I can retrieve sip firmware and load it on the 7960G Cisco phones? I have never done it before. Also I am looking for CCVP workbooks/labbooks does anyone know how I can get them? I ahve already bought all the Cisco Press books?
What is your current firmware version? Some firmware might require extra work...
-
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# -
What are the mandatory fields needed to setup/register the SIP phone manually in CUCM
What are the mandatory fields needed to setup/register the SIP phone manually.Also, if someone can let me know the mandatory fields for Cisco based SIP phone and also the third party SIP hard phones like Avaya or any other Third party SIP phones both Soft phone and physical phone requirements...in CUCM
Please suggest...I need to know if MAC address is mandatory for all Cisco SIP phone to setuphttp://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/7_1_2/ccmcfg/bccm-712-cm/b09sip3p.html
http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-version-70/112110-phone-add-00.html -
Cisco UC5xx 8.6 Support for 99xx SIP phones using CCA 3.2.1
Hi Friends,
I was looking through posts here in the SMB commuity, the SWP 8.6 for UC5xx, the RN for CCA 3.2.1, and the OLH for CCA 3.2.1, and found a nice thread that will help anyone wanting to get a 9951 or 9971 SIP phone to operate on the UC5xx after upgrade to the Cisco IOS {15.1(4)M4} bringing CME 8.6 to Telephony Services and selection of the SIP 9-2-2 phone loads (included in the SWP) for 99xx.
My only inquiry for Cisco to check, would be why isnt this documented in the release notes,
https://supportforums.cisco.com/servlet/JiveServlet/previewBody/26979-102-1-67567/cca_3_2_2_relnotes.pdf
since CCA doesnt seem to add the 'load 99xx sip99xx.9-2-2' statement, the 'tftp-path flash:', followed by the 'create profile' under VOICE REGISTER GLOBAL?
If it is supposed to work, then I would alert you that it did not. After Upgrading CCA to 3.2.1 and then upgrading the UC5xx to SWP 8.6, the adminisrtrator manually adds the 99xx phone by entering MAC and Type under Configure> Telephony> Users/Extensions> Users and Phones: ADD button. Nothing special, just a normal extension on one button, Video Enabled, and a VM box created. This allows the phone to register just fine, but it doesnt automatically upgrade to 9-2-2 due to the missing bold commands above.
I think if this were a known defect, it would have been documented in the RNs, so I raise it to your attention.
Which operation should have added these commands?
Can you let us know if this is an anomaly or if everyone will encounter this?
Thanks kindly,
SteveYeah uh beleive me Steven, I have tried everything and every location to get these phones working and nothing did. Other people have the same issue (thread here somewhere) luckily mine are only out 1 hour others are out many hours.
Thanks,
Bob James -
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 *1VPN 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.
-
Cisco SIP Phone 9971 won't register on CME 8.6
Hello,
I'm facing a very strange problem:
a Cisco SIP Phone 9971 won't register on CME 8.6 running on a 2811
I have read all the related-postings to this and other Forum, but I have not been able to solve it.
One of the "potential solutions" was to make sure that the Phone had a Line configured.
But I think that the commands voice register dn and voice register pool are properly configured (see config below)
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#Thank you for your reply.
I did some debugs and the results are very strange!
This is what I got:
Feb 24 18:01:12.219: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 172.35.140.12:5060;branch=z9hG4bK08011844
From: ;tag=189c5db6bd09000260cf3daf-289a76d1
To: ;tag=52488-160A
Date: Mon, 24 Feb 2014 18:01:12 GMT
Call-ID: [email protected]
CSeq: 1000 REFER
Content-Length: 0
Contact:
Feb 24 18:01:12.291: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
REGISTER sip:172.25.140.1 SIP/2.0
Via: SIP/2.0/UDP 172.35.140.12:5060;branch=z9hG4bK1e9ad079
From: ;tag=189c5db6bd0900032df02e9c-25d79707
To:
Call-ID: [email protected]
Max-Forwards: 70
Date: Fri, 01 Jan 1982 00:02:41 GMT
CSeq: 101 REGISTER
User-Agent: Cisco-CP9971/9.4.1
Contact: ;+sip.instance="
000000-0000-0000-0000-189c5db6bd09>";+u.sip!devicename.ccm.cisco.com="SEP189C5DB
6BD09";+u.sip!model.ccm.cisco.com="493";video
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-
cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-
cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.2,X-cisco-xsi-
8.0.1
Content-Length: 0
Reason: SIP;cause=200;text="cisco-alarm:22 Name=SEP189C5DB6BD09 ActiveLoad=sip99
71.9-4-1-9.loads InactiveLoad=sip9971.9-3-2SR1-1.loads Last=reset-reset"
Expires: 3600
Feb 24 18:01:12.395: voice_reg_get_reg_expires_timer: no voice register pool found
Feb 24 18:01:12.395: VOICE_REG_POOL: Register request for (1010) from (172.35.140.12)
Feb 24 18:01:12.395: VOICE_REG_POOL: Contact matches pool 1 number list 1
Feb 24 18:01:12.395: VOICE_REG_POOL: No entry for (172.35.140.12) found in srst contact table
Feb 24 18:01:12.395: VOICE_REG_POOL: key(1010) contact(172.35.140.12:5060) add to contact table
Feb 24 18:01:12.395: VOICE_REG_POOL: No entry for (1010) found in contact table
Feb 24 18:01:12.399: VOICE_REG_POOL: key(1010) contact(172.35.140.12) added to contact table
Feb 24 18:01:12.399: VOICE_REG_POOL: key(172.35.140.12) contact(1010) add to srst contact table
Feb 24 18:01:12.399: VOICE_REG_POOL: No entry for (172.35.140.12) found in srst contact table
Feb 24 18:01:12.399: VOICE_REG_POOL: key(172.35.140.12) contact(1010) added to srst contact table
Feb 24 18:01:12.399: VOICE_REG_POOL pool->tag(1), dn->tag(1), submask(1)
But right after these errors, I get the following:
Feb 24 18:01:12.399: VOICE_REG_POOL: Creating param container for dial-peer 4000
1.VOICE_REG_POOL pool->tag(1), dn->tag(1), submask(1)
VOICE_REG_POOL pool_tag(1), dn_tag(1)
Feb 24 18:01:12.399: VOICE_REG_POOL: Created dial-peer entry of type 0
Feb 24 18:01:12.399: VOICE_REG_POOL: Registration successful for 1010, registration id is 1
Feb 24 18:01:12.411: VOICE_REG_POOL: Contact matches pool 1 number list 1
Feb 24 18:01:12.411: VOICE_REG_POOL: GW SIS: X-cisco-cme-sis-1.0.0
Feb 24 18:01:12.411: VOICE REGISTER POOL-1 has registered.
Name:SEP189C5DB6BD09 IP:172.35.140.12 DeviceType:Phone
Feb 24 18:01:12.411: VOICE_REG_POOL: Pool[1]: service-control (reset type: 2) message sent to sip:[email protected]
Feb 24 18:01:12.411: voice_reg_privacy_update_to_phone: delay sending privacy update during bulk registration
Feb 24 18:01:12.415: //1/7B0070C28003/SIP/Msg/ccsipDisplayMsg:
====================
And when I do a sh voice register pool, I get the following:
C2811#sh voice register pool 1
Pool Tag 1
Config:
Mac address is 189C.5DB6.BD09
Type is 9971
Number list 1 : DN 1
Proxy Ip address is 0.0.0.0
Current Phone load version is Cisco-CP9971/9.4.1
DTMF Relay is enabled, rtp-nte
Call Waiting is enabled
DnD is disabled
Video is enabled
Camera is enabled
Busy trigger per button value is 0
call-forward b2bua busy 68600
keep-conference is enabled
registration expires timer max is 3600 and min is 120
username adm password adm
kpml signal is enabled
Lpcor Type is none
blf call list is enabled
Transport type is udp
service-control mechanism is supported
registration Call ID is [email protected]
Registration method: per line
Privacy feature is not configured.
Privacy button is disabled
active primary line is: 1010
contact IP address: 172.35.140.12 port 5060
Phone SIS Version: 6.0.2
GW SIS Version: 1.0.0
Dialpeers created:
Dial-peers for Pool 1:
dial-peer voice 40001 voip
destination-pattern 1010
session target ipv4:172.35.140.12:5060
session protocol sipv2
dtmf-relay rtp-nte
digit collect kpml
codec g711ulaw bytes 160
no vad
call-fwd-busy 68600
after-hours-exempt FALSE
Statistics:
Active registrations : 4
Total SIP phones registered: 1
Total Registration Statistics
Registration requests : 4
Registration success : 4
Registration failed : 0
unRegister requests : 0
unRegister success : 0
unRegister failed : 0
Attempts to register
after last unregister : 0
Last register request time : 18:11:43.551 UTC Mon Feb 24 2014
Last unregister request time :
Register success time : 18:11:43.551 UTC Mon Feb 24 2014
Unregister success time :
C2811#
So apparently the Phone is actually registered!
However, the Phone screens still shows this message: Phone Not Registered.
So frankly I don't understand what's going on!
I really hope somebody can help. Thanks! -
CUCM 8.6 Dropped call transfers involving SIP phones
Hi All,
I am a developer who has been tasked with figuring out why call transfers are being dropped by Cisco CUCM when the original call comes from a SIP phone. This scenario works:
Cisco phone calls another Cisco phone, which transfers the original call to a SIP phone
These scenarios do not work:
SIP phone calls Cisco phone, which transfers the original call to another Cisco phone
SIP phone calls Cisco phone, which transfers the original call to another SIP phone
I have researched the Call Manager traces to the best of my ability, and I see some info in there that could potentially point to the source of the problem. I am just unable to understand what the trace means:
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/active_CcDisconnReq: ccDisconnReq.onBehalfOf=Media : ccDisconnReq.s.sv=2 : ccDisconnReq.c.cv=47 |1,100,63,1.93259^10.10.10.85^*
10:23:08.672 |//SIP/Stack/Info/0x0/sipConstructContainerContext #### Created container=0xb0b42f58|1,100,71,1.1^*^*
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/appendReasonHdr: appendReasonHdr - Invalid Disconnect Cause(cause=47), No Reason Header Appended|1,100,63,1.93259^10.10.10.85^*
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/appendRPIDHdrForOriginalCalledParty: SIP device does not Support Orig Dialled Phone nego: 0|1,100,63,1.93259^10.10.10.85^*
I have been wondering whether this could be a codec issue, however the SIP phones we are using are configured with the following codecs:
G711U
G711A
G722
ILBC
GSM
and our SIP software is also set to accept the first codec offered by the remote side. It seems from the SIP client logs that G722 is being used as the codec to communicate with the Cisco phones, but perhaps I'm misinterpreting.
I have attached a CUCM trace of a call from a SIP phone (ext. 491) to a Cisco handset (ext. 170) where the Cisco handset attempts to transfer the call to another SIP phone (ext. 492). The trace snippet shown above is from this log.
I would really appreciate it if someone more experienced with VoIP/SIP/CUCM could take a look and offer any ideas on what the issue might be, and also how we might be able to address it. I can try to provide more info about our CUCM configuration if needed.
Thanks in advance!Leslie, so here is what I found from the traces....
To understand the difference we need to understand how cucm performs call transfers from a sccp signalling point and a sip signalling point
SCCP
When the transfer key is pressed
1. CUCM sends a CloseReceiveChannel and StopMediaTransmission to the IP phone involved in active media (referenced by the callids)
NB, here CUCM updates the call state on the phone to a call state of 8 which is "Hold"
2.CUCM tells the held party to listen MOH from MOH server
3.CUCM establishes newcall leg with the intended transfered destination..Once this call is connected
4.CUCM receives a new Transfer instruction from the transferring phone to connect the held party
5..CUCM sends a CloseReceiveChannel between the held phone and MOH server (to tear down the media)
6. Next CUCM sends a CloseReceiveChannel and StopMediaTransmission to the transfering party & transfered party to remove Xferring party from call
7. finally CUCM sends OpenReceiveChannel between the original called party and the transfered party..and call is done
For SIP signalling. when the first transfer key is pressed
1. CUCM sends invite (re-invite) with an inactive SDP (a=inactive) to indicate a break in media path
2. CUCM sends a Delayed offer to insert MOH or to resume Media stream
NB: CUCM expects a sendrecv offer with SDP to the DO. (NB:if cucm gets an inactive offer SDP in the 200 OK instead of providing a send-recv offer SDP, the media path remains in an inactive state and causes calls to dropcall will drop),CUCM sends an ACK with sendonly to the 200 OK
3.CUCM establishes newcall leg with the intended transfered destination..Once this call is connected
4.CUCM receives a new Transfer instruction from the transferring phone to connect the held party
5. Next CUCM sends a re-invite with an inactive SDP to indicate a break in media path to MOH (in attempt to complete transfer)
6.Next CUCM sends an inactive SDP to indicate a break in media path between transfering party & transfered party to remove Xferring party from call
7. Next CUCM sends a DO re-invite to connect the transfered party. The far end then sends 200 OK with the required SDP to connect the call
Now having explained all of these, we need to look at where the call is failing for SIP-----SCCP----SIP calls without MTP
lets look at succesful SCCP-----SCCP-----SIP without MTP
Point 4 above
++++++++Extension 170 presses the transfer button to connect the two calls (Callid=24378483)+++++++++++++
(0003395) SoftKeyEvent softKeyEvent=4(Trnsfer) lineInstance=1 callReference=24378483
Point 5 above
++++Next CUCM closed the media between extension 160 and MOH server callid=24378480(this is the only active call on this callid)+++
(0003396) CloseReceiveChannel conferenceID=24378480 passThruPartyID=16777845. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
Point 6 Above
+++++Next cucm closes the call between extension 170 and 490 callid=(24378483)++++++++
(0003395) CloseReceiveChannel conferenceID=24378483 passThruPartyID=16777847. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a8b(10.10.10.139)
(0003395) StopMediaTransmission conferenceID=24378483 passThruPartyID=16777847. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a8b(10.10.10.139)
Point 6 above for the sip side (since the destination is SIP, to tear down media to SCCP phone, so as to connect the caller to the xfered party)
+++++++Next CUCM sends a re-invite with a=inactive SDP to the sip phone ++++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885626,NET]
INVITE sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK23332dbee978
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
o=CiscoSystemsCCM-SIP 192115 2 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 0.0.0.0
m=audio 24560 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=ptime:20
a=inactive-----------------------------------------------------Inactive
Still part of Point 6 for SIP signalling
++++++++++++Next sip phone responds with a 200 OK recevonly SDP +++++++++++++++++++
//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 62220 index 1890 with 683 bytes:
[885628,NET]
SIP/2.0 200 OK
v=0
o=- 18077 11099 IN IP4 10.10.10.104
s=yasdjip
c=IN IP4 10.10.10.104
t=0 0
a=ptime:20
a=recvonly-------------------------------------a=recvonly
Finally Point 7 above..
+++++++++++++=Next cucm sends a DO re-invite to extension 492-sip phone++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885630,NET]
INVITE sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK233534ffec4a
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
To: ;tag=5B0E9816C2CA6D70F3166FB972EDE4C2
+++++++Next we get a 200 OK from sip phone with sdp=sendrecv+++++++++=
/SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 62220 index 1890 with 683 bytes:
[885634,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK233534ffec4a
Contact:
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
Call-ID: [email protected]
v=0
o=- 18077 11099 IN IP4 10.10.10.104
s=yasdjip
c=IN IP4 10.10.10.104
t=0 0
m=audio 16574 RTP/AVP 9 101
a=rtpmap:101 TELEPHONE-EVENT/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
+Now CUCM sends an OpenReceiveChannel and start media xmission to sccp phone (callid=24378480) with media parameters of sip phone++++++
(0003396) OpenReceiveChannel conferenceID=24378480 passThruPartyID=16777848 millisecondPacketSize=20 compressionType=6(Media_Payload_G722_64k) RFC2833PayloadType=101 qualifierIn=? sourceIpAddr=IpAddr.type:0 ipAddr:0x0a0a0a68000000000000000000000000(10.10.10.104). myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
(0003396) startMediaTransmission conferenceID=24378480 passThruPartyID=16777848 remoteIpAddress=IpAddr.type:0 ipAddr:0x0a0a0a68000000000000000000000000(10.10.10.104)
remotePortNumber=16574 milliSecondPacketSize=20 compressType=6(Media_Payload_G722_64k) RFC2833PayloadType=101 qualifierOut=?. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
+++++++++++=Next Phone sends its ACK+++++++++++++++
(0003396) OpenReceiveChannelAck Status=0, IpAddr=IpAddr.type:0 ipAddr:0x0a0a0a89000000000000000000000000(10.10.10.137), Port=20352, PartyID=16777848
+++++++++++=Next CUCM sends ACK to 200 OK from SIP Phone+++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885635,NET]
ACK sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK23366067b8c0
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
To: ;tag=5B0E9816C2CA6D70F3166FB972EDE4C2
Date: Tue, 19 Feb 2013 21:44:45 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 237
v=0
o=CiscoSystemsCCM-SIP 192115 3 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.137
b=TIAS:64000
b=AS:64
t=0 0
m=audio 20352 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Now at this point all is well...and the call is connected....
Now here is where the call is failing on the SIP-SCCP-SIP call without MTP
From Point 2 above, CUCM sends a DO to insert MOH, and then gets response, then sends an ACK to 200 Ok back to SIP Phone..
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 53361 index 1810
[881160,NET]
ACK sip:[email protected]:53361;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22035ecc1fcb
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Date: Tue, 19 Feb 2013 17:38:50 GMT
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
Max-Forwards: 70
CSeq: 102 ACK
o=CiscoSystemsCCM-SIP 190666 3 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195---------------------------------------IP address of MOH server
t=0 0
m=audio 4000 RTP/AVP 0--------------------------------MOH port 4000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendonly---------------------------------------------------------sendonly
+++++NOW Point 6 above (SIP) CUCM sends a=inactive to break media path to MOH server to connect caller and xfered party++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 53361 index 1810
[881161,NET]
INVITE sip:[email protected]:53361;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22045bb7f918
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Date: Tue, 19 Feb 2013 17:39:04 GMT
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 164
v=0
o=CiscoSystemsCCM-SIP 190666 4 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 0.0.0.0--------------------------------------------------------------------Media IP is 0.0.0.0
t=0 0
m=audio 4000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=inactive---------------------------------------------------------------------media inactive
At this point, we should get a response back from the sip phone...
and here is what we got..
++Trying which is expected++++
//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 53361 index 1810 with 331 bytes:
[881162,NET]
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22045bb7f918
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
CSeq: 103 INVITE
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Content-Length: 0
++++++++Then we get a BYE+++++++++++++++
/SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 53361 index 1810 with 576 bytes:
[881163,NET]
BYE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.104:53361;branch=z9hG4bKa2vdQvR7J9OiMyjU;rport
Contact:
Max-Forwards: 70
From: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
User-Agent: Acrobits Softphone Business/2.4.8
To: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
CSeq: 3 BYE
Content-Length: 0
So this is the root cause of the problem. Your SIP phone does not know how to respond to multiple media break between it and the MOH server.
The difference between this and the succesful SCCP-SIP--SCCP, is that the held party was a sccp phone, hence the sip phone only has to process one a=inactive SDP message, where as in the SIP-SCCP-SIP, the help party was sip, so the sip phone has to process two a=inactive SDP messages
Now what is the difference when MTP is involved! A Big difference. MTP stays in the media path. So there is never a break in media and no inactive SDP attribute is sent. The flow looks like below:
for the initial call...The SIP phone sends its media to MTP and likewise the SCCP phone
SIP------Media------MTP------------Media-------SCCP Phone
When the new destination is dialled and transfer is commited,
SIP-------------media----MTP--------media---------MTP
The final invoite sent to connect 492 and 491 has MTP as the IP address to connect Media to.
++++++++Ivite to 492 ++++++++++++++
INVITE sip:[email protected]:61303;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK231a3b24b862
From: ;tag=192048~d8e94532-127d-4dca-bba0-64b1675da032-24378472
To: ;tag=78FF5BF6C019A55EA020B69BB6A767E2
Date: Tue, 19 Feb 2013 21:24:59 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 214
v=0
o=CiscoSystemsCCM-SIP 192048 1 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195---------------------------------------------------------------the MTP ip address
t=0 0
m=audio 25038 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
+++++++Invite to 491 +++++++++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.94 on port 50376 index 1887
[885429,NET]
INVITE sip:[email protected]:50376;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK231b78d1b56
From: ;tag=192046~d8e94532-127d-4dca-bba0-64b1675da032-24378467
To: "491" ;tag=F13CE94DE942C47680356A647DC7F916
Date: Tue, 19 Feb 2013 21:24:59 GMT
Call-ID: AE7045FFB2D8D9C28D54651473A14A5D41B5B93C
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: "Leslie2" ;party=calling;screen=no;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 237
v=0
o=CiscoSystemsCCM-SIP 192046 1 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195----------------------------------------MTP
b=TIAS:64000
b=AS:64
t=0 0
m=audio 25030 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Wao! That was a long one isnt it...It was fun too.
So now you can look at your sip phones and see if they can accept two inactive sdp messages within the same call. That way you can remove MTP. otherwise you will have MTP involved in every single call involving a sip phone, even if they do not involve transfers
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared" -
Cisco SIP Phone 9971 will not register on CME 8.6
Hello,
I'm trying to configure a Cisco SIP Phone 9971,
but it won't register on CME 8.6, which is running on a 2811
The Phone shows this error message: Phone Not Registered.
And when I check the the Status Messages in the Phone, I see the following:
VPN Error: vpn is not configured
Actually, it shows all these 4 messages in a constant Loop:
12:01:59a SEP189C5DB6BD09.cnf.xml (TFTP)
12:01:59a No Trust List instaled
12:01:59a Updating Trust list
12:02:00a VPN Error: VPN is not Configured
It seems that this VPN Error is keeping the Phone from registering.
This is repeated for ever and the Phone never registers; at least that's what it appears.
However, when I do a sh voice register pool, I get the following:
C2811#sh voice register pool 1
Pool Tag 1
Config:
Mac address is 189C.5DB6.BD09
Type is 9971
Number list 1 : DN 1
Proxy Ip address is 0.0.0.0
Current Phone load version is Cisco-CP9971/9.4.1
DTMF Relay is enabled, rtp-nte
Call Waiting is enabled
DnD is disabled
Video is enabled
Camera is enabled
Busy trigger per button value is 0
call-forward b2bua busy 68600
keep-conference is enabled
registration expires timer max is 3600 and min is 120
username adm password adm
kpml signal is enabled
Lpcor Type is none
blf call list is enabled
Transport type is udp
service-control mechanism is supported
registration Call ID is [email protected]
Registration method: per line
Privacy feature is not configured.
Privacy button is disabled
active primary line is: 1010
contact IP address: 172.35.140.12 port 5060
Phone SIS Version: 6.0.2
GW SIS Version: 1.0.0
Dialpeers created:
Dial-peers for Pool 1:
dial-peer voice 40001 voip
destination-pattern 1010
session target ipv4:172.35.140.12:5060
session protocol sipv2
dtmf-relay rtp-nte
digit collect kpml
codec g711ulaw bytes 160
no vad
call-fwd-busy 68600
after-hours-exempt FALSE
Statistics:
Active registrations : 4
Total SIP phones registered: 1
Total Registration Statistics
Registration requests : 4
Registration success : 4
Registration failed : 0
unRegister requests : 0
unRegister success : 0
unRegister failed : 0
Attempts to register
after last unregister : 0
Last register request time : 18:11:43.551 UTC Mon Feb 24 2014
Last unregister request time :
Register success time : 18:11:43.551 UTC Mon Feb 24 2014
Unregister success time :
C2811#
This sh voice register pool seems to indicate that the Phone has actually registered.
But I still get the Phone Not Registered message on the screen!
I did some Debugs and they also seem to indicate that the Phone has indeed registered:
Feb 24 18:01:12.399: VOICE_REG_POOL: Creating param container for dial-peer 4000
1.VOICE_REG_POOL pool->tag(1), dn->tag(1), submask(1)
VOICE_REG_POOL pool_tag(1), dn_tag(1)
Feb 24 18:01:12.399: VOICE_REG_POOL: Created dial-peer entry of type 0
Feb 24 18:01:12.399: VOICE_REG_POOL: Registration successful for 1010, registration id is 1
Feb 24 18:01:12.411: VOICE_REG_POOL: Contact matches pool 1 number list 1
Feb 24 18:01:12.411: VOICE_REG_POOL: GW SIS: X-cisco-cme-sis-1.0.0
Feb 24 18:01:12.411: VOICE REGISTER POOL-1 has registered.
Name:SEP189C5DB6BD09 IP:172.35.140.12 DeviceType:Phone
So frankly, I have no idea why the Phone keeps showing the Phone Not Registered message.
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#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.
-
CME SIP Phone Calls in one-way (inside local network)
Hello everyone, first time here, need a little help.
I'm having some trouble to find a solution to the following problem.
Recently I've installed CME 9.1 using the router 2921. Most of the phones are SIPs, model 3905 (around 20 of them), with the last firmware updated.
Some users are complaining one way audio issue in internal calls, from a extension to another (only in sip phones)
With Wireshark capture I could see that RTP packets are being sent and receive by the router and not directly trough the phones. Is this normal in CME? When a call with problems occours (one way audio) there is no audio in one way, but router still sends confort noise packets.
Here is my config.
Thanks for any help.
Martin
##################################################################################33
System returned to ROM by power-on
System restarted at 11:29:23 BR Tue Jan 29 2013
System image file is "flash0:c2900-universalk9-mz.SPA.152-4.M2.bin"
Last reload type: Normal Reload
Last reload reason: power-on
voice service voip
no ip address trusted authenticate
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server expires max 3600 min 120
voice register global
mode cme
source-address 10.3.245.1 port 5060
max-dn 60
max-pool 70
load ATA-187 ATA187.9-2-3-1
load 3905 CP3905.9-2-1-0
authenticate realm all
timezone 17
time-format 24
date-format D/M/Y
tftp-path flash:
file text
create profile sync 0094230880392697
network-locale U1
user-locale U1 load /CME-locale-pt_BR-Portuguese-8.8.2.5.tar
ntp-server 10.3.244.7 mode directedbroadcast
voice register dn 1
number 9006
name Sala_Reuniao_02
label Sala de Reuniao 2
voice register dn 2
number 9007
name Sala_Reuniao_03
voice register dn 3
number 9008
name Sala Reuniao 04
voice register pool 1
id mac 8478.ACE6.09A2
type 3905
number 1 dn 1
template 1
codec g711ulaw
voice register pool 2
id mac 8478.ACE6.0573
type 3905
number 1 dn 2
codec g711ulaw
voice register pool 3
id mac 5897.1ECD.8F8D
type 3905
number 1 dn 3
codec g711ulaw
interface GigabitEthernet0/0
no ip address
duplex auto
speed auto
interface GigabitEthernet0/0.220
encapsulation dot1Q 220
ip address 10.3.245.1 255.255.255.0
ip helper-address 10.3.244.71
h323-gateway voip bind srcaddr 10.3.245.1
telephony-service
max-ephones 5
max-dn 5 no-reg both
ip source-address 10.3.245.1 port 2000
timeouts interdigit 5
timeouts busy 12
system message XXXXXXXX
cnf-file location flash:
cnf-file perphone
user-locale U2 load CME-locale-pt_BR-Portuguese-8.8.2.5.tar
user-locale 2 PT
network-locale U2
load 7925 CP7925G-1.4.1SR1.LOADS
load 6941 SCCP69xx.9-2-1-0.loads
time-zone 17
time-format 24
date-format dd-mm-yy
max-conferences 8 gain -6
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 1 dual-line
number 9001
ephone 1
mac-address D867.D9E6.F57F
ephone-template 1
type 6941
button 1:1Hi ,
We have upgarded the the firmware to the 3905.9-2-2ES2 , but show voice register pool phone-load still shows the old firmware, but the phoen itself is showing the new upgraded version on the dsiplay ...any advice is highly appricated,
ADM-CME9#show voice register pool phone-load
Pool Device Name Current-Version Previous-Version
==== =============== =========================== ===========================
1 SEP7081053DE72F Cisco/SPA502G-7.4.8a
3 SEP34BDC8C6C412 Cisco-CP3905/9.2.1
4 SEP34BDC8C64561 Cisco-CP3905/9.2.1
5 SEP54781AE1F531 Cisco-CP3905/9.2.1
6 SEP54781AE171D2 Cisco-CP3905/9.2.1
10 SEP54781AE1F544 Cisco-CP3905/9.2.1
15 SEP1CE6C77323CD Cisco-CP3905/9.2.1
16 SEP58971E282A23 Cisco-CP3905/9.2.1
17 SEP58971E2822A8 Cisco-CP3905/9.2.1
19 SEP1CE6C77321F3 Cisco-CP3905/9.2.1
30 SEP54781AE171E2 Cisco-CP3905/9.2.1
31 SEP54781AE16FD4 Cisco-CP3905/9.2.1
32 SEP54781AE16F2F Cisco-CP3905/9.2.1
33 SEP54781A1C77FD Cisco-CP3905/9.2.1
34 SEP54781A1C77DC Cisco-CP3905/9.2.1
35 SEP54781AE17527 Cisco-CP3905/9.2.1
36 SEP54781AE17766 Cisco-CP3905/9.2.1
37 SEP54781AE1731A Cisco-CP3905/9.2.1
38 SEP54781AE08B8D Cisco-CP3905/9.2.1
39 SEP54781AE123B1 Cisco-CP3905/9.2.1 -
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
thanksHi 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, -
Incoming calls issue in Third Party SIP Phone
Hi,
Yesterday I configured my third party sip phone which is yealink in this case on cucm and successfully registered it with cucm, despite of registration i have some calling issue in this phone. I am able to make outbound calls from this phone to any other phone however issue is related to inbound calls.I tried calling its DN from anywhere but call disconnect after sometime. Also didnt get any proper sip session trace in RTMT. Kindly suggest some step to sortout this issue.
ThanksDear Manish,
Call normally dicsonnected after 30-40 sec with termination code 102 in session trace. PFB SDI trace with 5030 is Thirdparty sip phone and 5033 is c7945. Looking forward for your suggestion.
CallingPartyNumber=5033
|DialingPartition=
|DialingPattern=5030
|FullyQualifiedCalledPartyNumber=5030
|DialingPatternRegularExpression=(5030)
|DialingWhere=
|PatternType=Enterprise
|PotentialMatches=NoPotentialMatchesExist
|DialingSdlProcessId=(0,0,0)
|PretransformDigitString=5030
|PretransformTagsList=SUBSCRIBER
|PretransformPositionalMatchList=5030
|CollectedDigits=5030
|UnconsumedDigits=
|TagsList=SUBSCRIBER
|PositionalMatchList=5030
|VoiceMailbox=
|VoiceMailCallingSearchSpace=PT-LHR-LOCAL:PT-Local:Unityvmpt:PT-F6-Local:PT-ISL-LOCAL:PT-KHI-LOCAL:PT_Operator_LHR:PT_Operator_KHI:PT_Operator_ISL
|VoiceMailPilotNumber=7103
|RouteBlockFlag=RouteThisPattern
|RouteBlockCause=0
|AlertingName=Syed Ahmer
|UnicodeDisplayName=Syed Ahmer
|DisplayNameLocale=1
|OverlapSendingFlagEnabled=0
12:17:38.028 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 172.16.200.21:[5062]:
[23928282,NET]
INVITE sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.11:5060;branch=z9hG4bK1ca0cc6e317649
From: "Syed Ahmer" ;tag=8787406~039e2a80-8561-4586-8954-d01ed2aa12c8-246211918
To:
Date: Thu, 30 Jan 2014 07:17:38 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Send-Info: conference, x-cisco-conference
Alert-Info:
Contact:
Remote-Party-ID: "Syed Ahmer" ;party=calling;screen=yes;privacy=off
Max-Forwards: 70
Content-Length: 0
|14,100,50,1.14103336^10.163.14.4^SEP00230432C828
12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 0|14,100,50,1.14103336^10.163.14.4^SEP00230432C828
12:17:38.028 |EnvProcessUdpHandler::fireSignal - SEND: index = 0, handler = 0xaf299320|*^*^*
12:17:38.028 |EnvProcessUdpPort::fireSignal - SEND, destination = 172.16.200.21:5062|*^*^*
12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 850, 172.16.200.21:5062)|*^*^*
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