ala cisco 7960 -----Original Message----- From: Matt Schulte Sent: Thursday, December 16, 2004 10:34 AM To: 'Paul A Brown' Subject: RE: [Asterisk-Users] Cisco 7960 (SIP) hold problems
Sure thing, the biggest problem I had was getting the SIP filenames working correctly for updating the firmware (blah, I love Cisco but these phones are a joke for support). This works for me! Good luck. sip.conf [107] host=dynamic type=friend context=default username=107 secret=blahblah mailbox=107 canreinvite=no disallow=all allow=all -------------------------- -sipMACADDRESS.cnf- image_version: P0S3-07-3-00 line1_name: 107 # Line 1 Registration Authentication line1_authname: "107" # Line 1 Registration Password line1_password: "elblahblah" --snip-- ####### 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: "Matt S 107" ; Has no effect on SIP messaging # Line 1 Display Name (Display name to use for SIP messaging) line1_displayname: "Matt S" # 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: "blahblahblah" ; Limited to 31 characters (Default - cisco) # User classifcation used when Registering [ none(default), phone, ip ] user_info: none ------------------------- sipdefault.cnf # Image Version image_version: "P0S3-07-3-00" # Proxy Server # Note: I put the proxy server information in the individual conf files # for each machine, since each box has different configs. You could, of course, # put all of them here in the Default file... proxy1_address: "192.168.1.17" #proxy2_address: "192.168.117.4" # Proxy Server Port (default - 5061) #proxy1_port:"5060" # Emergency Proxy info proxy_emergency: "192.168.1.17" proxy_emergency_port: "5060" # Backup Proxy info proxy_backup: "192.168.1.17" proxy_backup_port: "5060" # Outbound Proxy info outbound_proxy: "192.168.1.17" outbound_proxy_port: "5060" # NAT/Firewall Traversal nat_enable: "0" nat_address: "" voip_control_port: "5061" start_media_port: "16384" end_media_port: "32766" nat_received_processing: "0" # Proxy Registration (0-disable (default), 1-enable) proxy_register: "1" # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: "120" # Codec for media stream (g711ulaw (default), g711alaw, g729) preferred_codec: "none" # TOS bits in media stream [0-5] (Default - 5) tos_media: "5" # Enable VAD (0-disable (default), 1-enable) enable_vad: "0" # Allow for the bridge on a 3way call to join remaining parties upon hangup cnf_join_enable: "1" ; 0-Disabled, 1-Enabled (default) # Allow Transfer to be completed while target phone is still ringing semi_attended_transfer: "0" ; 0-Disabled, 1-Enabled (default) # Telnet Level (enable or disable the ability to telnet into this phone telnet_level: "1" ; 0-Disabled (default), 1-Enabled, 2-Privileged # 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 11 sip_invite_retx: "6" ; Default 7 timer_invite_expires: "180" ; Default 180 sec # Setting for Message speeddial to UOne box messages_uri: "8500" #********* Release 2 new config parameters ********** # TFTP Phone Specific Configuration File Directory tftp_cfg_dir: "./" # Time Server sntp_mode: "directedbroadcast" sntp_server: "17.254.0.49" time_zone: "CST" dst_offset: "1" dst_start_month: "April" dst_start_day: "" dst_start_day_of_week: "Sun" dst_start_week_of_month: "1" dst_start_time: "02" dst_stop_month: "Oct" dst_stop_day: "" dst_stop_day_of_week: "Sunday" dst_stop_week_of_month: "8" dst_stop_time: "2" dst_auto_adjust: "1" # 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-disabled, 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-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control) anonymous_call_block: "0" ; Default 0 (Disable blocking of anonymous calls) # Call Waiting (0-disabled, 1-enabled, 2-disabled with no user control, 3-enabled with no user control) call_waiting: "1" ; Default 1 (Call Waiting enabled) # DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127) dtmf_avt_payload: "101" ; Default 100 # XML file that specifies the dialplan desired dial_template: "dialplan" # Network Media Type (auto, full100, full10, half100, half10) network_media_type: "auto" #Autocompletion During Dial (0-off, 1-on [default]) autocomplete: "1" #Time Format (0-12hr, 1-24hr [default]) time_format_24hr: "1" ####### New Parameters added in Release 4.0 ####### # XML URLs services_url: "http://who.cares/blah/"; ; test url directory_url: "http://192.168.1.17/directories.xml"; # URL for external Directory location logo_url: "http://192.168.1.17/n2voiplogo.bmp"; ; URL for branding logo to be used on phone display # put your own logo in the logo_url location; I include the 10-20.com one for reference in building your own # HTTP Proxy Support http_proxy_addr: "" ; Address of HTTP Proxy server http_proxy_port: 80 ; Port of HTTP Proxy Server (80-default) # The dynamic tftp server should be set to whatever your TFTP server is. This way, it # keeps the tftp server setting even though you might be using DHCP (default behavior # is to use the DHCP server as a tftp server, which is rarely correct.) # Remote Party ID remote_party_id: 1 ; 0-Disabled (default), 1-Enabled -----Original Message----- From: Paul A Brown [mailto:[EMAIL PROTECTED] Sent: Thursday, December 16, 2004 10:07 AM To: Matt Schulte Subject: Re: [Asterisk-Users] Cisco 7960 (SIP) hold problems Hi Matt, Seems you have gone one step further than me, Can you possibly let em see your sip configs for the cisco 7960 and also the configs in the phone. I am having trouble getting it to talk to asterisk Thanks Paul ----- Original Message ----- From: "Matt Schulte" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, December 16, 2004 3:01 PM Subject: [Asterisk-Users] Cisco 7960 (SIP) hold problems > Has anyone had problems with using hold on a 7960 SIP firmware? The > problem is when the 7960 puts a call on hold and you take it off hold > again, the 7960 outbound audio is delayed on the other end. Sometimes > up to a few seconds. I've tried a couple different things, making the > "other end" a diff type of trunk ie: > > 7960sip --> asterisk --> IAX2 --> PRI > > 7960sip --> asterisk --> SER --> SIP proxy > > Anyone have a clue? The 7960 has the latest firmware, 7.3 or > something. Could this be a (the?) problem? Thanks! > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > No virus found in this incoming message. > Checked by AVG Anti-Virus. > Version: 7.0.296 / Virus Database: 265.5.4 - Release Date: 15/12/2004 > > -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.5.4 - Release Date: 15/12/2004 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
![http://192.168.1.17/n2voiplogo.bmp"](http://192.168.1.17/n2voiplogo.bmp"){kind=link}