[asterisk-users] Multiple lines on Linksys/Sipura phones
I'm going to be deploying around 30 IP phones with Asterisk in the near future. I've tentatively settled on the Linksys/Sipura SPA9xx family. I am unclear on the notion of lines in the context of SIP phones like these. The SPA942 model has a 2-line-to-4-line upgrade available, but I don't know why I'd need to purchase it. I have tested a SPA942 with Asterisk, and even without the upgrade, I can easily send/receive/hold four separate calls at a time, using the four line keys. What am I missing? Thanks. -- Kevin DeGraaf ___ --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
Re: [asterisk-users] Multiple lines on Linksys/Sipura phones
Kevin DeGraaf wrote: What am I missing? Thanks. Nothing. Keep your money in your wallet. Your users will never need or understand using more than 2 different calls appearances atone time. Even I, with several PBXs to mointor, I never us all the call appearances I have on a 601. -- Chris Mason (264) 497-5670 http://www.snapanumber.com/ Fax: (264) 497-8463 http://www.snapanumber.com/ Int: (305) 704-7249 http://www.snapanumber.com/ Fax: (815)301-9759 http://www.snapanumber.com/ UK 44.207.183.0271 http://www.snapanumber.com/ Cell: 264-235-5670 http://www.snapanumber.com/ Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, 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
[asterisk-users] Multiple lines in body of UserEvent
Hi everybody, I'm trying to send a user event from the dialplan like this: exten = s,n,UserEvent(EventName|var1:value1^var2:value2) The event is sent just fine, but the body is not split in two lines as it should be according to http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+UserEvent. Using 1.2.9.1 here, someone knows whether the split character has changed or the feature has been removed and it is simply not possible (anymore? Didn't need to try with previous versions...) to split the body argument of user events? Gratefull for any hint, Florian Müllner ___ --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
[Asterisk-Users] Multiple lines
Hi, Question... Is there a way to receive two phone calls on the same phone, or, for example to receive a phone call, put the call in stand-by and then make another call and finally, why not put them all together in conference... Thanks David Masure ___ 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
David, please search the wiki for meetme rooms; this is a standard feature. If you want to be able to the control those calls from a web interface do a search for meetme2 If you are only new to asterisk go and download [EMAIL PROTECTED] http://asteriskathome.sourceforge.net/ It's a iso you can download that does all of the configuring and setup for you automatically. Cheers dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Masure Sent: Wednesday, March 02, 2005 9:58 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Multiple lines Hi, Question... Is there a way to receive two phone calls on the same phone, or, for example to receive a phone call, put the call in stand-by and then make another call and finally, why not put them all together in conference... Thanks David Masure ___ 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
Dean, Thank you for your answer but fromwhat I know meetme is able to solde the conference problem, but how can I for example receive 2 phone calls at the same time on 1 phone and just switching from one line to another ? In my current config, I make a phone call and the SIP phone is answering, when trying to make a second call, I've got the music to hold me till first conversation has ended. Meanwhile, the sip phone user doesn't know there is a call waiting and so, he won't answer the line Is there a solution to that problem ? Thanks David -Message d'origine-De: dean collins [mailto:[EMAIL PROTECTED]Envoyé: mercredi 2 mars 2005 17:02À: Asterisk Users Mailing List - Non-Commercial DiscussionObjet: RE: [Asterisk-Users] Multiple lines David, please search the wiki for meetme rooms; this is a standard feature. If you want to be able to the control those calls from a web interface do a search for meetme2 If you are only new to asterisk go and download [EMAIL PROTECTED] http://asteriskathome.sourceforge.net/ It's a iso you can download that does all of the configuring and setup for you automatically. Cheers dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David MasureSent: Wednesday, March 02, 2005 9:58 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Multiple lines Hi, Question... Is there a way to receive two phone calls on the same phone, or, for example to receive a phone call, put the call in stand-by and then make another call and finally, why not put them all together in conference... Thanks David Masure ___ 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
Thats a typical situation for me on a Cisco 7940 - I'm sure its the same as any phone. When a second call comes in I can put the first on hold (and they hear MOH) and if I want I can blind-transfer both parties into a meetme room and then go join them there if I want. I'm not sure how the other phone brands handle multiple incoming calls but they must do. Derek David Masure wrote: Hi, Question... Is there a way to receive two phone calls on the same phone, or, for example to receive a phone call, put the call in stand-by and then make another call and finally, why not put them all together in conference... Thanks David Masure ___ 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 -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823 (International) +353 86 856 3823 Main Line: (Local Ireland) 1890 45 70 74 (International) +353 1 201 0180 Fax: (Local Ireland) 01 201 0085 (International) +353 1 201 0085 Email: [EMAIL PROTECTED] Web: www.rivertowerhosting.com ___ 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
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
[Asterisk-Users] Multiple lines on Cisco 7960
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: scott line1_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. -- 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
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
[Asterisk-Users] multiple lines with SIP like MGCP?
We have a Dlink DVG-1120M and were surprised that it was able to handle 2 simultaneous conversations to 2 seperate phones using only 1 MAC address and 1 IP address. So we asked ourselves..why can't other 1 MAC/1IP devices do this as well? I have a Grandstream 486 that has 1IP and 1MAC. But I don't see anywhere in sip.conf to add a second line to a device. Is this possible? Can this only be done with an MGCP device? Thanks, Matthew ___ 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
Re: [Asterisk-Users] multiple lines with SIP like MGCP?
The HT486 is a single-line device with a PSTN pass-thru. The only multiline IADs I know of are the SIPURAs and the Cisco ATA-186... What you do is you create 2 contexts, 1 for each line of the device and you set the host name to the IP address (or host name if applicable) of the IAD. Set the username of each context to the line's respective extension in Asterisk. Then in the web setup for the IAD, there should be a place to put the username for each line as well as the password... I have not tried this but it should work, SIP is not IP/MAC based it's more like SMTP, it's user based... -Chris ___ 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
Re: [Asterisk-Users] multiple lines with SIP like MGCP?
We have a Dlink DVG-1120M and were surprised that it was able to handle 2 simultaneous conversations to 2 seperate phones using only 1 MAC address and 1 IP address. So we asked ourselves..why can't other 1 MAC/1IP devices do this as well? I have a Grandstream 486 that has 1IP and 1MAC. But I don't see anywhere in sip.conf to add a second line to a device. Is this possible? Can this only be done with an MGCP device? I don't have a Granstream, but the Cisco and Snom does that. There are no standards that dictate an IP per line. ___ 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
[Asterisk-Users] multiple lines on 7960's
I'm assuming this works on the 7960's with * from looking at the wiki and reading other posts. (user has primary ext. for themselves, but can pick up and dial multiple other lines on the 7960, place these lines on hold, transfer, etc.) Can someone verify ? How does this look in extensions.conf ? Thanks, Chris Clifton ___ 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
RE: [Asterisk-Users] multiple lines on 7960's
Not if I understand but with the 7960 you can have one exten for the primary line and then have the other 5 softkeys register different sip extensions. Then just choose that extension and dial out. We do this for people you share 1 phone. -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Netlabz - Chris Clifton Posted At: Friday, February 20, 2004 2:39 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] multiple lines on 7960's Subject: [Asterisk-Users] multiple lines on 7960's I'm assuming this works on the 7960's with * from looking at the wiki and reading other posts. (user has primary ext. for themselves, but can pick up and dial multiple other lines on the 7960, place these lines on hold, transfer, etc.) Can someone verify ? How does this look in extensions.conf ? Thanks, Chris Clifton ___ 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 ___ 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
