[Asterisk-Users] Multiple lines on Cisco 7960
Scott Henderson
scott at finite-tech.com
Fri Jan 7 11:20:13 MST 2005
Someone on the list spotted the problem, there is a typo in my line
definitions.
Thanks all
Scott Henderson wrote:
> I set this up manually on the phone and it works just fine so config
> files ... I attached the complete config files so maybe someone can
> see what I am missing.
>
> ============
> argon:/tftpboot# cat SIPDefault.cnf
> # SIP Default Generic Configuration File
>
> # Image Version
> image_version: P0S3-07-3-00 ;
>
> # Proxy Server
> proxy1_address: "192.168.17.13" ; Can be dotted IP or FQDN
> proxy2_address: "192.168.17.13" ; Can be dotted IP or FQDN
> proxy3_address: "192.168.17.13" ; Can be dotted IP or FQDN
> proxy4_address: "192.168.17.13" ; Can be dotted IP or FQDN
> proxy5_address: "192.168.17.13" ; Can be dotted IP or FQDN
> proxy6_address: "192.168.17.13" ; 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: 1
>
> # Phone Registration Expiration [1-3932100 sec] (Default - 3600)
> timer_register_expires: 3600
>
> # Codec for media stream (g711ulaw (default), g711alaw, g729a)
> preferred_codec: none
>
> # 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: "192.168.17.11" ; SNTP Server IP Address
> sntp_mode: directedbroadcast ; unicast, multicast, anycast, or
> directedbroadcast (default)
> time_zone: YST ; 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: 0 ; 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: "http://192.168.17.11/asterisk-tux.bmp" ; 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-disabled, 1-enabled, 2-disabled no user
> control, 3-enabled no user control)
> call_hold_ringback: 0 ; Default 0 (Disable ringback of held
> call)
>
> ========================
> argon:/tftpboot# cat SIP00115C407FA3.cnf
> # SIP Configuration Generic File
>
> # Line 1
> line1_name: Scott
> line1_authname: "scott"
> line1_password: "scott"
>
> # Line 2
> line2_name: Scott1
> line2_authname: "scott1"
> line2_password: "scott1"
>
> # Line 3
> line2_name: "Line 2"
> line2_authname: "UNPROVISIONED"
> line2_password: "UNPROVISIONED"
>
> # Line 4
> line2_name: "Line 4"
> line2_authname: "UNPROVISIONED"
> line2_password: "UNPROVISIONED"
>
> # Line 5
> line2_name: "Line 5"
> line2_authname: "UNPROVISIONED"
> line2_password: "UNPROVISIONED"
>
> # Line 6
> line2_name: "Line 6"
> line2_authname: "UNPROVISIONED"
> 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: "User ID"
>
>
> ####### 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
>
> messages_uri: "_6101"
> argon:/tftpboot#
>
> Nabeel Jafferali wrote:
>
>>>I had not looked at the phones settings yet, thanks for the
>>>suggestion. The setting indicate that there is no configuration on the
>>>second line it is listed as "UNPROVISIONED"
>>>
>>>
>>
>>Go into the phone and program Line 2 Settings directly, without using
>>the SIP<MAC>.cnf file. If that works, then your .cnf file is wrong.
>>
>>
>>
>
>--
>Scott Henderson
>============================================================================
>Finite Technologies Incorporated
>3763 Image Drive, Anchorage, Alaska 99504
>Phone: 907.339.8085 ext 6101, Fax: 907.333.4482
>http://www.finite-tech.com
>http://www.chillywall.com
>http://www.virtuale.cc
>http://www.mphage.com
>Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK
>============================================================================
>
>
>------------------------------------------------------------------------
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Scott Henderson
============================================================================
Finite Technologies Incorporated
3763 Image Drive, Anchorage, Alaska 99504
Phone: 907.339.8085 ext 6101, Fax: 907.333.4482
http://www.finite-tech.com
http://www.chillywall.com
http://www.virtuale.cc
http://www.mphage.com
Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK
============================================================================
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050107/eed2af19/attachment.htm
More information about the asterisk-users
mailing list