[Asterisk-Users] Converting 7905G to SIP

Greg Thu, 24 Mar 2005 23:34:27 -0800

I am trying to convert my 7905G to be SIP based and seem to be running into a few hassles. Below are all the config files and logs from the server. I have tried to follow the pdf's from cisco and some posts from other mailing lists that google turnedup, but it seems that nothing is working. Am I somehow missing a fundamental step in trying to upgrade from Call Manager to SIP?

Any help is greatly appreciated.

Regards,
Greg


Host Name           SEP001193508864
Phone DN
App Load ID         CP7905010300SCCP040312B
Boot Load ID        LD0100BOOT021112A
Software Version    1.03.00(040312B)
Hardware Revision   0x0005 0x0000
Serial Number       INM08291F2V
Product ID          CP-7905G
H/W Features        0x00000002
BTXML Cards Version LD04-25-2002#0
Message Waiting

Here is the tftpd log on the asterisk box

Mar 25 17:25:35 pbx in.tftpd[14814]: RRQ from 192.168.2.200 filename SEP001193508864.cnf.xml Mar 25 17:25:35 pbx in.tftpd[14815]: RRQ from 192.168.2.200 filename ld001193508864 Mar 25 17:25:40 pbx in.tftpd[14815]: tftpd: read(ack): Connection refused

Here is the SEP001193508864.cnf.xml

<Default>
    <callManagerGroup>
        <members>
            <member  priority="0">
                <callManager>
                    <ports>
                        <ethernetPhonePort>2000</ethernetPhonePort>
                    </ports>
                    <processNodeName>192.168.2.1</processNodeName>
                </callManager>
            </member>
        </members>
    </callManagerGroup>

Here is the ld0011... file

# SIP Configuration Generic File (start)
image_version: P0S3-07-3-00
preferred_codec: g711ulaw
#preferred_codec: g729a

# Proxy Server
proxy1_address: "192.168.2.1"
proxy2_address: ""
proxy3_address: ""
proxy4_address: ""
proxy5_address: ""
proxy6_address: ""

# Line 1 Settings
line1_name: "2001"                        ; Line 1 Extension\User ID
line1_displayname: "Line1"    ; Line 1 Display Name
line1_shortname: "2001"
line1_authname: "2001"; Line 1 Registration Authentication
line1_password: "phone"         ; Line 1 Registration Password

# Line 2 Settings line2_name: "2001" ; Line 2 Extension\User ID line2_displayname: "Line2" ; Line 2 Display Name line2_shortname: "2001" line2_authname: "2001" ; Line 2 Registration Authentication line2_password: "phone" ; Line 2 Registration Password

# Line 3 Settings line3_name: "2001" ; Line 3 Extension\User ID line3_displayname: "Line3" ; Line 3 Display Name line3_shortname: "2001" line3_authname: "2001" ; Line 3 Registration Authentication line3_password: "phone" ; Line 3 Registration Password

# Line 4 Settings line4_name: "2001" ; Line 4 Extension\User ID line4_displayname: "Line4" ; Line 4 Display Name line4_shortname: "2001" line4_authname: "phone" ; Line 4 Registration Authentication line4_password: "101" ; Line 4 Registration Password

# Line 5 Settings line5_name: "2001" ; Line 5 Extension\User ID line5_displayname: "Line5" ; Line 5 Display Name line5_shortname: "2001" line5_authname: "2001" ; Line 5 Registration Authentication line5_password: "phone" ; Line 5 Registration Password

# Line 6 Settings line6_name: "2001" ; Line 6 Extension\User ID line6_displayname: "Line6" ; Line 6 Display Name line6_shortname: "2001" line6_authname: "2001" ; Line 6 Registration Authentication line6_password: "phone" ; Line 6 Registration Password

# Emergency Proxy info
proxy_emergency: ""
proxy_emergency_port: "5060"

# Backup Proxy info
proxy_backup: "192.168.2.1"
proxy_backup_port: "5060"

# Outbound Proxy info
outbound_proxy: "192.168.2.1"
outbound_proxy_port: "5060"

proxy_register: 1
timer_register_expires : 120
# NAT/Firewall Traversal
nat_enable: "1"
nat_address: ""
voip_control_port: "5060"
start_media_port: "16384"
end_media_port:  "32766"
nat_received_processing: "1"

# Phone Label (Text desired to be displayed in upper right corner)
phone_label: "Cisco - "            ; Has no effect on SIP messaging

# Time Zone phone will reside in time_zone: EST sntp_server: "136.159.2.254" ; SNTP Server IP Address sntp_mode: directedbroadcast ; unicast, multicast, anycast, or directedbroadcast (default)

# Telnet Level (enable or disable the ability to telnet into this phone
telnet_level: "2"      ; 0-Disabled (default), 1-Enabled, 2-Privileged

# Phone prompt/password for telnet/console session phone_prompt: "admin" ; Telnet/Console Prompt phone_password: "admin" ; Telnet/Console Password

# Enable_VAD (1-enabled, 0-disabled)
enable_vad: "0"

# Network Media Type (auto, full100, full10, half100, half10)
network_media_type: "auto"
user_info: phone
tftp_cfg_dir: "/"
sync: 1

# URL for external Directory location
#logo_url: http://www.domain.com/asterisk/tux.bmp
#directory_url: http://www.bkw.org/directory.cgi
#services_url: http://phone-xml.berbee.com/menu.xml
#messages_uri: 300
# SIP Configuration Generic File (stop)


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[Asterisk-Users] Converting 7905G to SIP

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