SIP Set-up

[aluro]

Hello, 

I have an Cisco IP-Phone 7962G and have tried getting it to work with Sipgate as my VOIP-Provider. 

I first of all downloaded the latest SIP firmware for the phone(cmterm-7942_7962-sip.9-4-2SR3-1) and managed to succesfully flash the new firmware to the phone using TFTPD64. Then i created the two needed files SIPDefault.cnf and SIPXXXXXX.cnf. (The x-es are my MAC-Adress!)

If needed I can post the contents of the files.

I included the SIPDefault.cnf file with the other files from the firmware and rebooted the phone. After that the software installed just fine.

I then deleted all contents from my root directory of my TFTP-Server and only had the SIPXXXXXXX.cnf file and restarted the phone. 

I checked the log and realised it has an issue with the SIPXXXXXX.cnf file. 

Heres what he says: 

File <CTLSEPXXXXXXXXX.tlv> : error 2 in system call CreateFile The system cannot find the file specified. [26/09 15:32:53.474]

I have read through all the disussions but it seems like I am doing something wrong. In general a big part of what I am doing seems wrong, but the problem is everyone has a different way of doing it. 

Can someone maybe give me a proper guide. 

One problem I ran into was finding theese POS files like this one: P0S3-08-10-00.zip . 

I looked everywhere but just cant find them. 

Maybe someone can help me. 

Help much appreciated!

Kind regards. 

[Rajan]

Hi Aluro,

I dont think you need P0S3-08-10-00.zip file for 7962/7942 phones. These are for 7960/40 series phones.

https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/7960g_7940g/firmware/sip/8_11/english/release/notes/796040sip_811.pdf

You need to check your config file and whether the required PBX information and other details are specified. IF all these are correct, just check whether the phone has got all these details.

HTH
Rajan
Please rate all useful posts by clicking the star below and mark solutions as accepted wherever applicable

[aluro]

Thanks, but the link you gave me is giving instructions for an 7960/40 series phones. 

Also what do you mean under PBX information and other details?

Is it possible that you give me an direct guide?

Thanks in forward!

Kind regards!

[Rajan]

Hi Aluro,

Yes, I mentioned that only for 7940/60 you need that P0S3-08-10-00.zip file, hence given that link for your reference. By PBX information what I meant was, you need to provide the details of the server to which you are trying to register your 7942/62 phone to in the SEPmacaddress.cnf.xml file.

You can follow the link shared by Leo to get a sample config for the same.

HTH
Rajan

[aluro]

Ok, thanks.
I will try that as soon as I have some time and report back to you.

Kind regards.

[Leo Laohoo · Accepted solution]

---

@aluro wrote:

SIPDefault.cnf 

---

Phones running 9.X.X firmware (and later) don't need SIPDefault.cnf file. 

I want to see the SEPmacaddress.cnf.xml.

[aluro]

# SIP Default Generic Configuration File

# Image Version
image_version: P0S3-08-10-00

# Proxy Server
proxy1_address: "sipgate.de" ; 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: 5061
proxy2_port: 5062
proxy3_port: 5063
proxy4_port: 5060
proxy5_port: 5060
proxy6_port: 5060

# Proxy Registration (0-disable (default), 1-enable)
proxy_register: 1

# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: 500

# 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: "192.53.103.104" ; SNTP Server IP Address
sntp_mode: unicast ; unicast, multicast, anycast, or directedbroadcast (default)
time_zone: CET ; 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: "217.10.79.9" ; Dotted IP of Backup Proxy
proxy_backup_port: 5061 ; Backup Proxy port (default is 5060)

# Emergency Proxy Support
proxy_emergency: "sipgate.de" ; 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: 1 ; 0-Disabled (default), 1-Enabled
nat_address: "hostname.dyndns.org" ; WAN IP address of NAT box (dotted IP or DNS A record only)
voip_control_port: 5061 ; 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: 1 ; 0-Disabled (default), 1-Enabled

# Outbound Proxy Support
outbound_proxy: "sipgate.de" ; restricted to dotted IP or DNS A record only
outbound_proxy_port: 5061 ; 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: 2 ; 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

[Leo Laohoo]

The configuration you've posted is for a 7940/7960 and it is invalid for a 7962.

[aluro]

So how do I have to configure it?

Is there any documentation I missed?

Kind reagrds

[Leo Laohoo]

A known working configuration can be found HERE.

[aluro]

I apologize for not replying for such a long time. 

I have actually written a reply but it somehow didnt work. 

Now to come back to my problem:

I downloaded the configuration but stil cant get it to work. 

I configured everything as written there. 

But apparently I am missing something out.

[Leo Laohoo]

---

@aluro wrote:
I downloaded the configuration but stil cant get it to work. 
I configured everything as written there. 
But apparently I am missing something out.

---

That's nice, but with this limited information there is not a lot anyone can do to help.  

[aluro]

Alright,
Thank you anyway I will try and maybe I’ll get it working.

[aluro]

Ok, so after a long time I have an update. I have more knowledge in Cisco phone systems and managed to set up several phones without any issues. However my 7962 still has a problem. The phone gets all the configuration like the proxy or time server but not the proxy port. I have had that issue on several phones. I've tried different config files but none of them seems to work. I am going to post my main configuration file here in case something is missing or mistyped. I checked the log and theres nothing like "error validating config info" so the file isn't broken or has wrong parameters.

Here's my config file:

<?xml version="1.0" encoding="UTF-8"?>
<device>

<deviceProtocol>SIP</deviceProtocol>

<sshUserId>cisco</sshUserId>
<sshPassword>cisco</sshPassword>

<devicePool>
<dateTimeSetting>
<dateTemplate>D.M.YY</dateTemplate>
<timeZone>Central Europe Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>130.149.17.21</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>

<callManagerGroup>
<tftpDefault>true</tftpDefault>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<processNodeName>sipgate.de</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>

<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>

<loadInformation></loadInformation>

<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<pcPort>0</pcPort>
<settingsAccess>1</settingsAccess>
<garp>0</garp>
<voiceVlanAccess>0</voiceVlanAccess>
<videoCapability>0</videoCapability>
<autoSelectLineEnable>0</autoSelectLineEnable>
<sshAccess>0</sshAccess>
<sshPort>22</sshPort>
<webAccess>0</webAccess>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
<daysDisplayNotActive></daysDisplayNotActive>
<displayOnTime>06:00</displayOnTime>
<displayOnDuration>18:00</displayOnDuration>
<displayIdleTimeout>00:30</displayIdleTimeout>
<displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
</vendorConfig>

<deviceSecurityMode>1</deviceSecurityMode>

<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>

<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL>http://www.eurobilltracker.eu/?referer=80060</servicesURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>

<transportLayerProtocol>2</transportLayerProtocol>

<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
</capf>
</capfList>

<certHash></certHash>
<encrConfig>false</encrConfig>

<sipProfile>
<sipProxies>
<backupProxy></backupProxy>
<backupProxyPort></backupProxyPort>
<emergencyProxy></emergencyProxy>
<emergencyProxyPort></emergencyProxyPort>
<outboundProxy></outboundProxy>
<outboundProxyPort></outboundProxyPort>
<registerWithProxy>true</registerWithProxy>
</sipProxies>

<sipCallFeatures>
<cnfJoinEnabled>true</cnfJoinEnabled>
<callForwardURI>x--serviceuri-cfwdall</callForwardURI>
<callPickupURI>x-cisco-serviceuri-pickup</callPickupURI>
<callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI>
<callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI>
<meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI>
<abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI>
<rfc2543Hold>false</rfc2543Hold>
<callHoldRingback>2</callHoldRingback>
<localCfwdEnable>true</localCfwdEnable>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>2</anonymousCallBlock>
<callerIdBlocking>2</callerIdBlocking>
<dndControl>0</dndControl>
<remoteCcEnable>true</remoteCcEnable>
</sipCallFeatures>

<sipStack>
<sipInviteRetx>6</sipInviteRetx>
<sipRetx>10</sipRetx>
<timerInviteExpires>180</timerInviteExpires>
<timerRegisterExpires>3600</timerRegisterExpires>
<timerRegisterDelta>5</timerRegisterDelta>
<timerKeepAliveExpires>120</timerKeepAliveExpires>
<timerSubscribeExpires>120</timerSubscribeExpires>
<timerSubscribeDelta>5</timerSubscribeDelta>
<timerT1>500</timerT1>
<timerT2>4000</timerT2>
<maxRedirects>70</maxRedirects>
<remotePartyID>false</remotePartyID>
<userInfo>None</userInfo>
</sipStack>

<autoAnswerTimer>1</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>false</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>none</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>

<natEnabled>false</natEnabled>
<natAddress></natAddress>

<stutterMsgWaiting>0</stutterMsgWaiting>

<callStats>false</callStats>

<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>

<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>

<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate></dialTemplate>

<phoneLabel>Kevin Office</phoneLabel>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>username</featureLabel>
<name>username</name>
<displayName>Urosevic</displayName>
<contact>uswername</contact>

<proxy>sipgate.de</proxy>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>

<authName>username</authName>
<authPassword>pass</authPassword>

<sharedLine>true</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>**600</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>

<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button="2">
<featureID>2</featureID>
<featureLabel>Telekom</featureLabel>
<speedDialNumber>speedDialNumber</speedDialNumber>
</line>
<line button="3">
<featureID>9</featureID>
<featureLabel>Telekom</featureLabel>
<name>username</name>
<displayName>username</displayName>
<contact>username</contact>

<proxy>tel.t-online.de</proxy>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>

<authName>username</authName>
<authPassword>pass</authPassword>

<sharedLine>true</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber></messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>

<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button="4">
<featureID>9</featureID>
<featureLabel>624</featureLabel>
<name>624</name>
<displayName>624</displayName>
<contact>624</contact>

<proxy>192.168.178.1</proxy>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>

<authName>624</authName>
<authPassword>123456</authPassword>

<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>**600</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>

<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
</sipProfile>
</device>

Kind regards