Re: [Asterisk-Users] Multiple lines on Cisco 7960
There should be no quotes after the : in the cisco SIPmacaddress.cnf files. Change it from: # Line 1 line1_name: Scott line1_authname: scott line1_password: scott # Line 2 line2_name: Scott1 line2_authname: scott1 line2_password: scott1 To: # Line 1 line1_name: Scott line1_authname:scott line1_password:scott # Line 2 line2_name: Scott1 line2_authname:scott1 line2_password:scott1 this works for me hope it works for you too. On Sat, 08 Jan 2005 02:38:00 +0800, Nathan Alberti [EMAIL PROTECTED] wrote: Theres your problem right there; All of them say line2_X Nathan. # 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 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) #
Re: [Asterisk-Users] Multiple lines on Cisco 7960
I did set these to the correct poxy serveras well in the SIPDefault.cnf file. This is very frustrating problem, I have ready dozens of posts that refer to how to set this up and I see mto have followed all the suggestions. 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 Scott Nathan Alberti wrote: Do you have: # Proxy Server proxy1_address: x.x.x.x proxy2_address: x.x.x.x Unsure if this is required, does your phone list the correct server ? (settings | 4 | 2 | 6) Nathan. Scott Henderson wrote: I have been trying to get multiple lines on the 7960 to work for several days. i have read all the posts I can find and have run multiple sip debug and have gotten no place on this. Here are the relevant section of the config files: sip.conf [scott] type=friend host=dynamic username=scott secret=scott context=default mailbox=6101 callerid=Scott Henderson [scott1] type=friend host=dynamic username=scott1 secret=scott1 context=default mailbox=6101 callerid=Scott Henderson 1 macaddress.cnf # Line 1 line1_name: Scott line1_authname: scottline1_password: scott # Line 2 line2_name: Scott1 line2_authname: scott1 line2_password: scott1 sip debug output from resetting the phone: Sip read: REGISTER sip:192.168.17.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK61a1a63a From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: CSCO/7 Contact: sip:[EMAIL PROTECTED]:5060 Content-Length: 0 Expires: 3600 10 headers, 0 lines Using latest request as basis request Sending to 192.168.17.114 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK61a1a63a From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as00424045 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.17.114:5060 Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK61a1a63a From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as00424045 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=0045611f Content-Length: 0 to 192.168.17.114:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms argon*CLI Sip read: REGISTER sip:192.168.17.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK49f2aa87 From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: CSCO/7 Contact: sip:[EMAIL PROTECTED]:5060 Authorization: Digest username=scott,realm=asterisk,uri=sip:192.168.17.13,response=7b9f392d15161ef76ae35f283e876497,nonce=0045611f,algorithm=md5 Content-Length: 0 Expires: 3600 11 headers, 0 lines Using latest request as basis request Sending to 192.168.17.114 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK49f2aa87 From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as00424045 Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.17.114:5060 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK49f2aa87 From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as00424045 Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: sip:[EMAIL PROTECTED]:5060;expires=3600 Date: Fri, 07 Jan 2005 02:56:25 GMT Content-Length: 0 to 192.168.17.114:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.17.13:5060;branch=z9hG4bK29419b41 From: asterisk sip:[EMAIL PROTECTED];tag=as42c5efcf To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 36 Messages-Waiting: no Voicemail: 0/0 (no NAT) to 192.168.17.114:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms argon*CLI Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.17.13:5060;branch=z9hG4bK29419b41 From: asterisk sip:[EMAIL PROTECTED];tag=as42c5efcf To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] Date: Fri, 07 Jan 2005 02:56:26 GMT CSeq: 102 NOTIFY Content-Length: 0 8 headers, 0 lines Destroying call '[EMAIL PROTECTED]' Destroying call '[EMAIL PROTECTED]' argon*CLI The result of this configuration is that I always get the first line line_1 but never the second
RE: [Asterisk-Users] Multiple lines on Cisco 7960
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 SIPMAC.cnf file. If that works, then your .cnf file is wrong. -- Nabeel Jafferali tel: 416.491.9136 (toronto) 646.225.7426 (new york) fwd: 46990 email/msn : nabeelatjafferali.net ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Multiple lines on Cisco 7960
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) #
Re: [Asterisk-Users] Multiple lines on Cisco 7960
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) ###
Re: [Asterisk-Users] Multiple lines on Cisco 7960
Theres your problem right there; All of them say line2_X Nathan. # 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 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
Re: [Asterisk-Users] Multiple lines on Cisco 7960
Do you have: # Proxy Server proxy1_address: x.x.x.x proxy2_address: x.x.x.x Unsure if this is required, does your phone list the correct server ? (settings | 4 | 2 | 6) Nathan. Scott Henderson wrote: I have been trying to get multiple lines on the 7960 to work for several days. i have read all the posts I can find and have run multiple sip debug and have gotten no place on this. Here are the relevant section of the config files: sip.conf [scott] type=friend host=dynamic username=scott secret=scott context=default mailbox=6101 callerid=Scott Henderson [scott1] type=friend host=dynamic username=scott1 secret=scott1 context=default mailbox=6101 callerid=Scott Henderson 1 macaddress.cnf # Line 1 line1_name: Scott line1_authname: scottline1_password: scott # Line 2 line2_name: Scott1 line2_authname: scott1 line2_password: scott1 sip debug output from resetting the phone: Sip read: REGISTER sip:192.168.17.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK61a1a63a From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: CSCO/7 Contact: sip:[EMAIL PROTECTED]:5060 Content-Length: 0 Expires: 3600 10 headers, 0 lines Using latest request as basis request Sending to 192.168.17.114 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK61a1a63a From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as00424045 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.17.114:5060 Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK61a1a63a From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as00424045 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=0045611f Content-Length: 0 to 192.168.17.114:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms argon*CLI Sip read: REGISTER sip:192.168.17.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK49f2aa87 From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: CSCO/7 Contact: sip:[EMAIL PROTECTED]:5060 Authorization: Digest username=scott,realm=asterisk,uri=sip:192.168.17.13,response=7b9f392d15161ef76ae35f283e876497,nonce=0045611f,algorithm=md5 Content-Length: 0 Expires: 3600 11 headers, 0 lines Using latest request as basis request Sending to 192.168.17.114 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK49f2aa87 From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as00424045 Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.17.114:5060 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK49f2aa87 From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as00424045 Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: sip:[EMAIL PROTECTED]:5060;expires=3600 Date: Fri, 07 Jan 2005 02:56:25 GMT Content-Length: 0 to 192.168.17.114:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.17.13:5060;branch=z9hG4bK29419b41 From: asterisk sip:[EMAIL PROTECTED];tag=as42c5efcf To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 36 Messages-Waiting: no Voicemail: 0/0 (no NAT) to 192.168.17.114:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms argon*CLI Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.17.13:5060;branch=z9hG4bK29419b41 From: asterisk sip:[EMAIL PROTECTED];tag=as42c5efcf To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] Date: Fri, 07 Jan 2005 02:56:26 GMT CSeq: 102 NOTIFY Content-Length: 0 8 headers, 0 lines Destroying call '[EMAIL PROTECTED]' Destroying call '[EMAIL PROTECTED]' argon*CLI The result of this configuration is that I always get the first line line_1 but never the second line. From what I can tell the phone never even tries to register the second line. ___ Asterisk-Users mailing list Asterisk-Users@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
