FW: [asterisk-users] Cisco 7960
Softphone Eyebeam v 1.5.2 _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mohamed A. Gombolaty Sent: Tuesday, February 27, 2007 2:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 Dear Khaled, What is the softphone u r using? Thx MAG Khaled wrote: I am using firmware version pos3-07-500 Kindly can you provide me with the basic configuration for cisco ip phone and asterisk config file *I have nat=never at my asterisk config file and nat enabled N0 at cisco phone *I have an out bound proxy ip and port 5060 at cisco phone *Voip control port is 5061 My problem is my soft phone can call the cisco phone with normal RTP and Bye message,but my cisco phone cant dial my soft phone. Asterisk sends bye message for my soft phone. Thanks _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Wireless Sent: Tuesday, February 27, 2007 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 can you give a bit more info? I know that you need nat=never for example - Original Message - From:Khaled mailto:[EMAIL PROTECTED] To:'Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com Non-Commercial Discussion' Cc:[EMAIL PROTECTED] Sent: Tuesday, February 27, 2007 10:03 AM Subject: [asterisk-users] Cisco 7960 Hi I have cisco 7960 connected to asterisk ,using tftp xml config file,my problem is it can receive any call but it cant call any extension. Please can you send me ,how to solve this issue Regards Khaled Chehab System Integration Engineer Xplorium Offshore. Sakiet Al Janzir Postal Code: 1102-2080 Tel: (961) 1- 868 686 Fax :(961) 1-808 810 GSM: (961) 3-979 343 _ * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. _ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * _ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the
Re: FW: [asterisk-users] Cisco 7960
Dear All, Please send the sip configuration for both phones along with a debug from asterisk when you try to call from cisco to the eyebeam? also are you trying to make them call peer to peer or not? What I am suspecting is that there must be something mismatching when the cisco phone tries to call the softphone you just need to focus on the debug and check the configuration. Thx MAG Khaled wrote: Softphone Eyebeam v 1.5.2 --- From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mohamed A. Gombolaty Sent: Tuesday, February 27, 2007 2:03 PM To:Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 Dear Khaled, What is the softphone u r using? Thx MAG Khaled wrote: I am using firmware version pos3-07-500 Kindly can you provide me with the basic configuration for cisco ip phone and asterisk config file *I have nat=never at my asterisk config file and nat enabled N0 at cisco phone *I have an out bound proxy ip and port 5060 at cisco phone *Voip control port is 5061 My problem is my soft phone can call the cisco phone with normal RTP and Bye message,but my cisco phone cant dial my soft phone. Asterisk sends bye message for my soft phone. Thanks - From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Wireless Sent: Tuesday, February 27, 2007 12:48 PM To:Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 can you give a bit more info? I know that you need nat=never for example - Original Message - From:Khaled To:'Asterisk Users Mailing List - Non-Commercial Discussion' Cc:[EMAIL PROTECTED] Sent: Tuesday, February 27, 2007 10:03 AM Subject: [asterisk-users] Cisco 7960 Hi I have cisco 7960 connected to asterisk ,using tftp xml config file,my problem is it can receive any call but it cant call any extension. Please can you send me ,how to solve this issue Regards Khaled Chehab System Integration Engineer Xplorium Offshore. Sakiet Al Janzir Postal Code: 1102-2080 Tel: (961) 1- 868 686 Fax :(961) 1-808 810 GSM: (961) 3-979 343 --- * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of
FW: [Asterisk-Users] Cisco 7960 (SIP) hold problems
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) #
