Re: [asterisk-users] Cisco 7960 SIP Upgrade
As expected, Jim took care of me WRT the Cisco upgrade. It is now far more usable than when it was SCCP... I gave up on trying to get SCCP working in Asterisk after upgrading to 1.4 from 1.0. Due to his generosity, I feel I owe him to recommend his termination\origination services. The one or two times I've had any issue, he has been quick to respond and took care of me. -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Sigma Networks [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 04, 2008 12:34 PM Subject: Re: [asterisk-users] Cisco 7960 SIP Upgrade Mike Hammett wrote: I couldn't figure it out on my own. I tried to purchase a Smartnet for the phone, but the original 7960 is not supported. Is it technically possible and if so, what would it cost me to have someone remote into my network and upgrade my SCCP 7960 to the latest SIP firmware? -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Mike, I know you are a very happy customer of Sigma Networks ( :-) )... I'd be happy to upgrade the phone to 8.3.3SR2 for you. Jim ph: 408-701-9929 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7960 SIP Upgrade
I couldn't figure it out on my own. I tried to purchase a Smartnet for the phone, but the original 7960 is not supported. Is it technically possible and if so, what would it cost me to have someone remote into my network and upgrade my SCCP 7960 to the latest SIP firmware? -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 SIP Upgrade
Mike Hammett wrote: I couldn't figure it out on my own. I tried to purchase a Smartnet for the phone, but the original 7960 is not supported. Is it technically possible and if so, what would it cost me to have someone remote into my network and upgrade my SCCP 7960 to the latest SIP firmware? -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Mike, I know you are a very happy customer of Sigma Networks ( :-) )... I'd be happy to upgrade the phone to 8.3.3SR2 for you. Jim ph: 408-701-9929 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 SIP Upgrade
That I am. I'll contact you off list. -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Sigma Networks [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 04, 2008 12:34 PM Subject: Re: [asterisk-users] Cisco 7960 SIP Upgrade Mike Hammett wrote: I couldn't figure it out on my own. I tried to purchase a Smartnet for the phone, but the original 7960 is not supported. Is it technically possible and if so, what would it cost me to have someone remote into my network and upgrade my SCCP 7960 to the latest SIP firmware? -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Mike, I know you are a very happy customer of Sigma Networks ( :-) )... I'd be happy to upgrade the phone to 8.3.3SR2 for you. Jim ph: 408-701-9929 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 / SIP tftp configs
On Wed, 2005-08-24 at 12:44 -0400, Asterisk User Group wrote: I have three questions about my 7960 phone that I can't discern from the docs/wiki. 1st - If I change the SIPxx.cnf file to change registrations it sets up new lines as expected. If I delete a line it doesn't get removed when I reboot the phone. I have to go to the phone, unlock it, and reset the SIP parameters. How do I make it forget what it has programmed and listen only to the download? Change it to UNPROVISIONED 2nd - Has anyone figured out how to get the Message button to launch a dial to VoicemailMain? messages_uri: 3rd - How do I display on the LCD an alias to the registered line? line1_name: 2000 line1_authname: 2000 line1_password: ** line1_shortname: Home The doc seems to suggest that line1_name is what it registers with and line1_authname is what it uses if challenged during the authentication. This doesn't make any sense to me. I am looking for the line to be 2000 but the display to say Home or Business, etc. Thanks, dbc. ___ 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 ___ --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] Cisco 7960 SIP 8-3-0
Are you using the Non-CallManager version? _ Mobilcom http://www.mobilcom.net - Original Message - From: Tong [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, July 17, 2006 8:56 PM Subject: Re: [asterisk-users] Cisco 7960 SIP 8-3-0 if you don't report it to cisco they won't know that bug exisit. - Original Message - From: Daryl Johnson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, July 17, 2006 4:05 PM Subject: Re: [asterisk-users] Cisco 7960 SIP 8-3-0 Tim, I have seen the same 400 errors and the broken MWI... I backed up to 7.3... We'll see if Cisco corrects these in the next release... Daryl - Original Message - From: Tim Connolly [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, July 17, 2006 12:06 PM Subject: [asterisk-users] Cisco 7960 SIP 8-3-0 Looks like the MWI broke on 8-3 also... ___ --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] Cisco 7960 SIP 8-3-0 getting Got SIP response 400
After upgrading my phones I now see routine error messages: -- Got SIP response 400 Bad Request back from 10.5.1.94 Asterisk SVN-trunk-r7230 Cisco 7960 SIP version 8-3-0. Sip show peer: * Name : 14012 Secret : Set MD5Secret: Not set Context : labcm33 Subscr.Cont. : Not set Language : AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : 14012 VM Extension : asterisk LastMsgsSent : 0 Call limit : 0 Dynamic : Yes Callerid : removed Expire : 272931 Insecure : no Nat : Always ACL : No CanReinvite : Yes PromiscRedir : No User=Phone : No Trust RPID : No Send RPID: No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 10.5.1.94 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 14012 SIP Options : (none) Codecs : 0x8000e (gsm|ulaw|alaw|h263) Codec Order : (ulaw,alaw,gsm) Status : Unmonitored Useragent: Cisco-CP7960G/8.0 Reg. Contact : sip:[EMAIL PROTECTED]:5060;transport=udp Any ideas? The phones seem to work fine other than the annoying console message. Is there some secret setting I can add to my config to stop this? ___ --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] Cisco 7960 SIP 8-3-0
Looks like the MWI broke on 8-3 also... ___ --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] Cisco 7960 SIP 8-3-0
Tim, I have seen the same 400 errors and the broken MWI... I backed up to 7.3... We'll see if Cisco corrects these in the next release... Daryl - Original Message - From: Tim Connolly [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, July 17, 2006 12:06 PM Subject: [asterisk-users] Cisco 7960 SIP 8-3-0 Looks like the MWI broke on 8-3 also... ___ --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 ___ --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] Cisco 7960 SIP 8-3-0
if you don't report it to cisco they won't know that bug exisit. - Original Message - From: Daryl Johnson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, July 17, 2006 4:05 PM Subject: Re: [asterisk-users] Cisco 7960 SIP 8-3-0 Tim, I have seen the same 400 errors and the broken MWI... I backed up to 7.3... We'll see if Cisco corrects these in the next release... Daryl - Original Message - From: Tim Connolly [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, July 17, 2006 12:06 PM Subject: [asterisk-users] Cisco 7960 SIP 8-3-0 Looks like the MWI broke on 8-3 also... ___ --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 ___ --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 virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.1/390 - Release Date: 7/17/2006 ___ --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] Cisco 7960 SIP - Displaying Time
Yes it does display caller id as callingnumber@ip of calling party but that does not interfere with me hitting dial from missed calls. Seems the Cisco phone sends the sip INVITE as callingnumber@ip of calling party rather than callingnumber@ip address of defaultproxyserver but asterisk ignores the info after the @? Chris - Original Message - From: Omar A. Sabek [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 13, 2006 11:45 PM Subject: Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time The 8.2 firmware displays Caller ID as callingnumber@proxyaddr... this becomes problematic for users that want to dial from their 'Missed Calls' log. Omar On 3/13/06, Nathan Bowyer [EMAIL PROTECTED] wrote: On 3/13/06, Chris Stenton [EMAIL PROTECTED] wrote: I have had no issues with 8.2 so far! Chris Except the Caller ID issue reported in another thread? This issue has been fixed in SIP firmware 7.5 Omar A. Sabek Yes, and I read that SIP 7.5 firmware have some other issues. They recommend using 7.4 firmware. I'm not sure how good in new 8.2 firmware. Tomislav ___ --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 ___ --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 ___ --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] Cisco 7960 SIP - Displaying Time
Chris, you may have a different and simpler setup. Internal calls work fine here, since the proxy server on the CallerID is the same proxy server used for all internal users. I was referring to calls that originate outside of the enterprise. I should have been more clear. Omar On 3/14/06, Chris Stenton [EMAIL PROTECTED] wrote: Yes it does display caller id as callingnumber@ip of calling party but that does not interfere with me hitting dial from missed calls. Seems the Cisco phone sends the sip INVITE as callingnumber@ip of calling party rather than callingnumber@ip address of defaultproxyserver but asterisk ignores the info after the @? Chris - Original Message - From: Omar A. Sabek [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 13, 2006 11:45 PM Subject: Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time The 8.2 firmware displays Caller ID as callingnumber@proxyaddr... this becomes problematic for users that want to dial from their 'Missed Calls' log. Omar On 3/13/06, Nathan Bowyer [EMAIL PROTECTED] wrote: On 3/13/06, Chris Stenton [EMAIL PROTECTED] wrote: I have had no issues with 8.2 so far! Chris Except the Caller ID issue reported in another thread? This issue has been fixed in SIP firmware 7.5 Omar A. Sabek Yes, and I read that SIP 7.5 firmware have some other issues. They recommend using 7.4 firmware. I'm not sure how good in new 8.2 firmware. Tomislav ___ --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 ___ --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 ___ --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 ___ --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] Cisco 7960 SIP - Displaying Time
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tue, 14 Mar 2006, Omar A. Sabek wrote: Chris, you may have a different and simpler setup. Internal calls work fine here, since the proxy server on the CallerID is the same proxy server used for all internal users. I was referring to calls that originate outside of the enterprise. I should have been more clear. Omar Do you have canreinvite=no in the phone definition in sip.conf? I am running our Cisco 7960s that way and under v8.2 the CallerID always shows the IP of the local asterisk server. This way hitting the Dial softkey works perfectly wherever the call originated. - -- Ron Wellsted [EMAIL PROTECTED] http://www.wellsted.org.uk N 52.567623, W 2.137621 Linux Counter No. 202120 FWD:519961 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2.2 (GNU/Linux) iQEVAwUBRBa/qUtP/KMNOfRbAQKxGAf8DNDFTudN+rKXVVyhUyAJ2X9Ku9oZYg0F 3EzcyMqDG1Fgly4IxRFpML480TFN+cxqZAflFB92cwECO980y/geGN3XZA6izHKK 4PC+90iWCjhXFUR7aJo+wJ2jkCA/BozAQiGDA2wtkctRy0OQEdaAsxiRt5gY/Sm7 9xSz82KNXp0HM/InBK1abwd4n0UQ9Wm+v+3wrdD3XL0elp0FFQaaesSZS2PDMWCT JSdDPfDoWN7t+VeDEeA+qugTYvt3HBJF8pDOzogg8Tnw1hhFXIYeATe8p2XNypkN cJ/YshMWxi5/sLSyc8musc8t4UzBcIYB/Cdqm8s55+oBPziuhRgPHw== =3cdx -END PGP SIGNATURE- ___ --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] Cisco 7960 SIP - Displaying Time
I have had no issues with 8.2 so far! Chris - Original Message - From: Tomislav Parcina [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 13, 2006 7:10 AM Subject: RE: [Asterisk-Users] Cisco 7960 SIP - Displaying Time -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Omar A. Sabek Sent: 9. ozujak 2006 18:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time This issue has been fixed in SIP firmware 7.5 Omar A. Sabek Yes, and I read that SIP 7.5 firmware have some other issues. They recommend using 7.4 firmware. I'm not sure how good in new 8.2 firmware. Tomislav ___ --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 ___ --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] Cisco 7960 SIP - Displaying Time
On 3/13/06, Chris Stenton [EMAIL PROTECTED] wrote: I have had no issues with 8.2 so far! Chris Except the Caller ID issue reported in another thread? This issue has been fixed in SIP firmware 7.5 Omar A. Sabek Yes, and I read that SIP 7.5 firmware have some other issues. They recommend using 7.4 firmware. I'm not sure how good in new 8.2 firmware. Tomislav ___ --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] Cisco 7960 SIP - Displaying Time
The 8.2 firmware displays Caller ID as callingnumber@proxyaddr... this becomes problematic for users that want to dial from their 'Missed Calls' log. Omar On 3/13/06, Nathan Bowyer [EMAIL PROTECTED] wrote: On 3/13/06, Chris Stenton [EMAIL PROTECTED] wrote: I have had no issues with 8.2 so far! Chris Except the Caller ID issue reported in another thread? This issue has been fixed in SIP firmware 7.5 Omar A. Sabek Yes, and I read that SIP 7.5 firmware have some other issues. They recommend using 7.4 firmware. I'm not sure how good in new 8.2 firmware. Tomislav ___ --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 ___ --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] Cisco 7960 SIP - Displaying Time
To be totally honest, I have 7.5 running on many phones and I have yet to receive a report on a firmware related issue. Omar On 3/13/06, Tomislav Parcina [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Omar A. Sabek Sent: 9. ozujak 2006 18:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time This issue has been fixed in SIP firmware 7.5 Omar A. Sabek Yes, and I read that SIP 7.5 firmware have some other issues. They recommend using 7.4 firmware. I'm not sure how good in new 8.2 firmware. Tomislav ___ --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 ___ --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] Cisco 7960 SIP - Displaying Time
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Omar A. Sabek Sent: 9. ozujak 2006 18:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time This issue has been fixed in SIP firmware 7.5 Omar A. Sabek Yes, and I read that SIP 7.5 firmware have some other issues. They recommend using 7.4 firmware. I'm not sure how good in new 8.2 firmware. Tomislav ___ --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] Cisco 7960 SIP - Displaying Time
Is there a way to display the time of the 7960 running firmware 7.4? Im unable to find any information. Add the following to SIPDefault.cnf or SIPMAC.cnf: sntp_server: time.nrc.ca sntp_mode: unicast time_zone: EST You should of course change your NTP server and/or time zone. Nabeel ___ --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] Cisco 7960 SIP - Displaying Time
On Thu, 2006-03-09 at 10:36 -0500, Nabeel Jafferali wrote: Is there a way to display the time of the 7960 running firmware 7.4? Im unable to find any information. Add the following to SIPDefault.cnf or SIPMAC.cnf: sntp_server: time.nrc.ca sntp_mode: unicast time_zone: EST You should of course change your NTP server and/or time zone. On my 7960 with 7.4 firmware, the time automagically disappears for some unknown reason. The phone still functions, but the time goes away until I reboot it. Not a big deal to me, so I have not investigated it further. -Greg ___ --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] Cisco 7960 SIP - Displaying Time
We had that problem for a while. You have to configure the ntp server in the phone so it'll pull the time otherwise it just randomly loses it. Aaron Greg Oliver wrote: On Thu, 2006-03-09 at 10:36 -0500, Nabeel Jafferali wrote: Is there a way to display the time of the 7960 running firmware 7.4? Im unable to find any information. Add the following to SIPDefault.cnf or SIPMAC.cnf: sntp_server: time.nrc.ca sntp_mode: unicast time_zone: EST You should of course change your NTP server and/or time zone. On my 7960 with 7.4 firmware, the time automagically disappears for some unknown reason. The phone still functions, but the time goes away until I reboot it. Not a big deal to me, so I have not investigated it further. -Greg ___ --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 ___ --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] Cisco 7960 SIP - Displaying Time
On 3/9/06, Greg Oliver [EMAIL PROTECTED] wrote: On Thu, 2006-03-09 at 10:36 -0500, Nabeel Jafferali wrote: Is there a way to display the time of the 7960 running firmware 7.4? Im unable to find any information. Add the following to SIPDefault.cnf or SIPMAC.cnf: sntp_server: time.nrc.ca sntp_mode: unicast time_zone: ESTOn my 7960 with 7.4 firmware, the time automagically disappears for someunknown reason. The phone still functions, but the time goes awayuntil I reboot it.Not a big deal to me, so I have not investigated it further.-Greg I use anycast. Seems like I read something about directbroadcast not working in recent SIP versions. ___ --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] Cisco 7960 SIP - Displaying Time
This issue has been fixed in SIP firmware 7.5 Omar A. Sabek On 3/9/06, Nathan Bowyer [EMAIL PROTECTED] wrote: On 3/9/06, Greg Oliver [EMAIL PROTECTED] wrote: On Thu, 2006-03-09 at 10:36 -0500, Nabeel Jafferali wrote: Is there a way to display the time of the 7960 running firmware 7.4? Im unable to find any information. Add the following to SIPDefault.cnf or SIPMAC.cnf: sntp_server: time.nrc.ca sntp_mode: unicast time_zone: EST On my 7960 with 7.4 firmware, the time automagically disappears for some unknown reason. The phone still functions, but the time goes away until I reboot it. Not a big deal to me, so I have not investigated it further. -Greg I use anycast. Seems like I read something about directbroadcast not working in recent SIP versions. ___ --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 ___ --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] Cisco 7960 SIP - Displaying Time
Is there a way to display the time of the 7960 running firmware 7.4? Im unable to find any information. Thanks, Ben Blakely ___ --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] Cisco 7960 (SIP) with Asterisk: how to get # towork during a call
Mahilal Silva wrote: Mike, Were you able to get this working? Even after with a entry in the dialplan.xml does not work for me. Thanks, This is what I have in my dialplan.xml DIALTEMPLATE TEMPLATE MATCH=*Timeout=5/ !-- Anything else -- TEMPLATE MATCH=*78Timeout=5/ !-- Anything else -- TEMPLATE MATCH=*79Timeout=5/ !-- Anything else -- TEMPLATE MATCH=#Timeout=5/ !-- Anything else -- /DIALTEMPLATE Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Cisco 7960 (SIP) with Asterisk: how to get # towork during a call
Mike, Were you able to get this working? Even after with a entry in the dialplan.xml does not work for me. Thanks, Ken On 6/20/05, Michael J. Tubby B.Sc (Hons) G8TIC [EMAIL PROTECTED] wrote: Andrew,I presume you mean in the Cisco 7940/7960 SIP Phone Administrator's Guide?When you say mapped, dou mean that it needs an explicit entry in the dialplan.xml like: TEMPLATE MATCH=# Timeout=0 User=Phone/ !--Explicit # for Asterisk --Mike- Original Message - From: Andrew Latham [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comSent: Thursday, June 16, 2005 2:53 PMSubject: Re: [Asterisk-Users] Cisco 7960 (SIP) with Asterisk: how to get #towork during a call# and * are mapped later in the SIP(Default/MAC).cnf it has a section in the manual if you want to see why.On 6/16/05, Michael J. Tubby B.Sc (Hons) G8TIC [EMAIL PROTECTED]wrote: Gents, I've built an Asterisk system to replace our PBX at work and have Cisco 7960 phones (SIP 7.4) running with Asterisk 1.0.7. How to I get Asterisk to recognise the '#' being pressed during a call? In sip.conf I have entries likle this: [2001] type=friend context=local-phone auth=md5 username=2001 secret=xyzzy callerid=Jack Tubby 2001 host=dynamic nat=no canreinvite=no dtmfmode=rfc2833 incominglimit=2 [EMAIL PROTECTED] disallow=all allow=alaw allow=ulaw callgroup=2 pickupgroup=2 and in the SIPDefault.cnf for the phones I have: # 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 DTMF works for voicemail and for remote services over both analogue Zap channels and digital (ISDN) channels. Asterisk doesn't appear to be 'monitoring' the audio so I can't get to Asterisk features like Asterisk's transfer, parked calls and one-tuch-record... Am I missing something? Mike ___ 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 --sigAndrew Latham - AKA: LATHAMA (lay-th-ham-eh)WWW: http://lathama.comEmail: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED]If any of the above are down we have bigger problems than my email!/sig___ Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Cisco 7960 / SIP tftp configs
Matt Schulte wrote: 1) You have to do a factory reset, or wipe out the line config. 2) By default it dials ext 8500 I believe. 3) You *should* be able to change _name, I can't remember the effect that has since you already have authname in. Matt -Original Message- From: Asterisk User Group [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 24, 2005 11:45 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960 / SIP tftp configs I have three questions about my 7960 phone that I can't discern from the docs/wiki. 1st - If I change the SIPxx.cnf file to change registrations it sets up new lines as expected. If I delete a line it doesn't get removed when I reboot the phone. I have to go to the phone, unlock it, and reset the SIP parameters. How do I make it forget what it has programmed and listen only to the download? In the SIPphone mac.cnf file put the value UNPROVISIONED into each lineX variable which you want removed. 2nd - Has anyone figured out how to get the Message button to launch a dial to VoicemailMain? Just set the messages_uri: parameter to be the lead number for your voicemail server. 3rd - How do I display on the LCD an alias to the registered line? line1_name: 2000 line1_authname: 2000 line1_password: ** I think you want the lineX_shortname parameter. The doc seems to suggest that line1_name is what it registers with and line1_authname is what it uses if challenged during the authentication. This doesn't make any sense to me. I am looking for the line to be 2000 but the display to say Home or Business, etc. Thanks, dbc. ___ 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 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 -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[EMAIL PROTECTED] ___ 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] Cisco 7960 / SIP tftp configs
Thanks for the responses. All is happy. For the record the correct answers are: Q1 - Additions/changes SIPxx.cnf take effect on reboots, deletions do not. A1 - Don't just comment out the line setting, change it specifically to UNPROVISIONED. Q2 - How to get Message button working. A2 - Simply set messages_uri: where is the extension for VM. (Sorry but this should have been obvious, I did indeed find lots of stuff once I started searching on uri instead of url. Thanks for not burning me for not doing my research.) Note this line does not appear to be in the default SIPDefault.cnf file, you must add it manually. Q3 - How do I display an alias on the LCD for a registered line? A3 - In SIPx.cnf add line1_shortname: what I want displayed Note: this line does not appear in the default SIPxx.cnf file, you must add it manually. dbc. ___ --Bandwidth and Colocation sponsored by Easynews.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] Cisco 7960 / SIP tftp configs
Asterisk User Group wrote: Thanks for the responses. All is happy. For the record the correct answers are: Q1 - Additions/changes SIPxx.cnf take effect on reboots, deletions do not. A1 - Don't just comment out the line setting, change it specifically to UNPROVISIONED. Q2 - How to get Message button working. A2 - Simply set messages_uri: where is the extension for VM. (Sorry but this should have been obvious, I did indeed find lots of stuff once I started searching on uri instead of url. Thanks for not burning me for not doing my research.) Note this line does not appear to be in the default SIPDefault.cnf file, you must add it manually. Q3 - How do I display an alias on the LCD for a registered line? A3 - In SIPx.cnf add line1_shortname: what I want displayed Note: this line does not appear in the default SIPxx.cnf file, you must add it manually. Somewhere on Cisco's site there are lists of parameters which are included in the config files by default, those which are not and those which can only be changed via the config file. fyi, Steve dbc. ___ --Bandwidth and Colocation sponsored by Easynews.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 -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] Cisco 7960 / SIP tftp configs
I have three questions about my 7960 phone that I can't discern from the docs/wiki. 1st - If I change the SIPxx.cnf file to change registrations it sets up new lines as expected. If I delete a line it doesn't get removed when I reboot the phone. I have to go to the phone, unlock it, and reset the SIP parameters. How do I make it forget what it has programmed and listen only to the download? 2nd - Has anyone figured out how to get the Message button to launch a dial to VoicemailMain? 3rd - How do I display on the LCD an alias to the registered line? line1_name: 2000 line1_authname: 2000 line1_password: ** The doc seems to suggest that line1_name is what it registers with and line1_authname is what it uses if challenged during the authentication. This doesn't make any sense to me. I am looking for the line to be 2000 but the display to say Home or Business, etc. Thanks, dbc. ___ 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] Cisco 7960 / SIP tftp configs
line1_shortname: Home line1_displayname:Home On 8/24/05, Asterisk User Group [EMAIL PROTECTED] wrote: I have three questions about my 7960 phone that I can't discern from the docs/wiki. 1st - If I change the SIPxx.cnf file to change registrations it sets up new lines as expected. If I delete a line it doesn't get removed when I reboot the phone. I have to go to the phone, unlock it, and reset the SIP parameters. How do I make it forget what it has programmed and listen only to the download? 2nd - Has anyone figured out how to get the Message button to launch a dial to VoicemailMain? 3rd - How do I display on the LCD an alias to the registered line? line1_name: 2000 line1_authname: 2000 line1_password: ** The doc seems to suggest that line1_name is what it registers with and line1_authname is what it uses if challenged during the authentication. This doesn't make any sense to me. I am looking for the line to be 2000 but the display to say Home or Business, etc. Thanks, dbc. ___ 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 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] Cisco 7960 / SIP tftp configs
I'm not in the office at the moment to make sure, but if memory serves,
to set a value to 'nothing or null'
line1_name: UNPROVISIONED
messages_uri: 123
where 123 is in extensions.conf as
exten = 123,1,VoiceMailMain(${CALLERIDNUM})
or something similar
line1_shortname: Alias
Best Regards,
Ben
- Original Message -
From: Asterisk User Group
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Cisco 7960 / SIP tftp configs
Sent: 8/24/2005 1:05:59 PM
I have three questions about my 7960 phone that I can't discern from the
docs/wiki.
1st - If I change the SIPxx.cnf file to change registrations it sets
up new lines as expected. If I delete a line it doesn't get removed when
I reboot the phone. I have to go to the phone, unlock it, and reset the
SIP parameters. How do I make it forget what it has programmed and
listen only to the download?
2nd - Has anyone figured out how to get the Message button to launch a
dial to VoicemailMain?
3rd - How do I display on the LCD an alias to the registered line?
line1_name: 2000
line1_authname: 2000
line1_password: **
The doc seems to suggest that line1_name is what it registers with and
line1_authname is what it uses if challenged during the
authentication. This doesn't make any sense to me. I am looking for the
line to be 2000 but the display to say Home or Business, etc.
Thanks, dbc.
___
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RE: [Asterisk-Users] Cisco 7960 / SIP tftp configs
1) You have to do a factory reset, or wipe out the line config. 2) By default it dials ext 8500 I believe. 3) You *should* be able to change _name, I can't remember the effect that has since you already have authname in. Matt -Original Message- From: Asterisk User Group [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 24, 2005 11:45 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960 / SIP tftp configs I have three questions about my 7960 phone that I can't discern from the docs/wiki. 1st - If I change the SIPxx.cnf file to change registrations it sets up new lines as expected. If I delete a line it doesn't get removed when I reboot the phone. I have to go to the phone, unlock it, and reset the SIP parameters. How do I make it forget what it has programmed and listen only to the download? 2nd - Has anyone figured out how to get the Message button to launch a dial to VoicemailMain? 3rd - How do I display on the LCD an alias to the registered line? line1_name: 2000 line1_authname: 2000 line1_password: ** The doc seems to suggest that line1_name is what it registers with and line1_authname is what it uses if challenged during the authentication. This doesn't make any sense to me. I am looking for the line to be 2000 but the display to say Home or Business, etc. Thanks, dbc. ___ 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 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] Cisco 7960 / SIP tftp configs
I've not had the deleting lines problem. When you're deleting a line, are you changing the config to ... line6_name: line6_displayname: line6_shortname: line6_authname: line6_password: #Change lineX_shortname: to whatever you want them to see on the LCD. line4_name: line4_displayname: line4_shortname: Line4 line4_authname: line4_password: password #Change to your VoiceMailMain() extension messages_uri: Louie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Asterisk User Group Sent: Wednesday, August 24, 2005 10:45 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960 / SIP tftp configs I have three questions about my 7960 phone that I can't discern from the docs/wiki. 1st - If I change the SIPxx.cnf file to change registrations it sets up new lines as expected. If I delete a line it doesn't get removed when I reboot the phone. I have to go to the phone, unlock it, and reset the SIP parameters. How do I make it forget what it has programmed and listen only to the download? 2nd - Has anyone figured out how to get the Message button to launch a dial to VoicemailMain? 3rd - How do I display on the LCD an alias to the registered line? line1_name: 2000 line1_authname: 2000 line1_password: ** The doc seems to suggest that line1_name is what it registers with and line1_authname is what it uses if challenged during the authentication. This doesn't make any sense to me. I am looking for the line to be 2000 but the display to say Home or Business, etc. Thanks, dbc. ___ 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 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] Cisco 7960 (SIP) with Asterisk: how to get # towork during a call
Andrew, I presume you mean in the Cisco 7940/7960 SIP Phone Administrator's Guide? When you say mapped, dou mean that it needs an explicit entry in the dialplan.xml like: TEMPLATE MATCH=# Timeout=0 User=Phone/ !-- Explicit # for Asterisk -- Mike - Original Message - From: Andrew Latham [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, June 16, 2005 2:53 PM Subject: Re: [Asterisk-Users] Cisco 7960 (SIP) with Asterisk: how to get # towork during a call # and * are mapped later in the SIP(Default/MAC).cnf it has a section in the manual if you want to see why. On 6/16/05, Michael J. Tubby B.Sc (Hons) G8TIC [EMAIL PROTECTED] wrote: Gents, I've built an Asterisk system to replace our PBX at work and have Cisco 7960 phones (SIP 7.4) running with Asterisk 1.0.7. How to I get Asterisk to recognise the '#' being pressed during a call? In sip.conf I have entries likle this: [2001] type=friend context=local-phone auth=md5 username=2001 secret=xyzzy callerid=Jack Tubby 2001 host=dynamic nat=no canreinvite=no dtmfmode=rfc2833 incominglimit=2 [EMAIL PROTECTED] disallow=all allow=alaw allow=ulaw callgroup=2 pickupgroup=2 and in the SIPDefault.cnf for the phones I have: # 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 DTMF works for voicemail and for remote services over both analogue Zap channels and digital (ISDN) channels. Asterisk doesn't appear to be 'monitoring' the audio so I can't get to Asterisk features like Asterisk's transfer, parked calls and one-tuch-record... Am I missing something? Mike ___ 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 -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ 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 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] Cisco 7960 (SIP) with Asterisk: how to get # to work during a call
Gents, I've built an Asterisk system to replace our PBX at work and have Cisco 7960 phones (SIP 7.4) running with Asterisk 1.0.7. How to I get Asterisk to recognise the '#' being pressed during a call? In sip.conf I have entries likle this: [2001] type=friend context=local-phone auth=md5 username=2001 secret=xyzzy callerid=Jack Tubby 2001 host=dynamic nat=no canreinvite=no dtmfmode=rfc2833 incominglimit=2 [EMAIL PROTECTED] disallow=all allow=alaw allow=ulaw callgroup=2 pickupgroup=2 and in the SIPDefault.cnf for the phones I have: # 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 DTMF works for voicemail and for remote services over both analogue Zap channels and digital (ISDN) channels. Asterisk doesn't appear to be 'monitoring' the audio so I can't get to Asterisk features like Asterisk's transfer, parked calls and one-tuch-record... Am I missing something? Mike ___ 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] Cisco 7960 (SIP) with Asterisk: how to get # to work during a call
# and * are mapped later in the SIP(Default/MAC).cnf it has a section in the manual if you want to see why. On 6/16/05, Michael J. Tubby B.Sc (Hons) G8TIC [EMAIL PROTECTED] wrote: Gents, I've built an Asterisk system to replace our PBX at work and have Cisco 7960 phones (SIP 7.4) running with Asterisk 1.0.7. How to I get Asterisk to recognise the '#' being pressed during a call? In sip.conf I have entries likle this: [2001] type=friend context=local-phone auth=md5 username=2001 secret=xyzzy callerid=Jack Tubby 2001 host=dynamic nat=no canreinvite=no dtmfmode=rfc2833 incominglimit=2 [EMAIL PROTECTED] disallow=all allow=alaw allow=ulaw callgroup=2 pickupgroup=2 and in the SIPDefault.cnf for the phones I have: # 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 DTMF works for voicemail and for remote services over both analogue Zap channels and digital (ISDN) channels. Asterisk doesn't appear to be 'monitoring' the audio so I can't get to Asterisk features like Asterisk's transfer, parked calls and one-tuch-record... Am I missing something? Mike ___ 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 -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ 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] CIsco 7960 SIP Image
Does anyone have a document I can use as a guide on how to load a SIP image on a cisco 7960 phone? Ryan ___ 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] CIsco 7960 SIP Image
www.voip-info.org has it Preston Garrison direct: 877-748-4142 fax: 310-774-3901 cell: 623-748-4140 -Original Message- From: Ryan Finnesey [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tue, 31 May 2005 11:18:47 -0400 Subject: [Asterisk-Users] CIsco 7960 SIP Image Does anyone have a document I can use as a guide on how to load a SIP image on a cisco 7960 phone? Ryan ___ 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 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] CIsco 7960 SIP Image
Ryan, This should have everything you need. http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx Phil. Ryan Finnesey [EMAIL PROTECTED] poratechnologies. To com Asterisk Users Mailing List - Sent by: Non-Commercial Discussion asterisk-users-bo asterisk-users@lists.digium.com [EMAIL PROTECTED] cc m.com Subject [Asterisk-Users] CIsco 7960 SIP 31/05/2005 16:18 Image Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com Does anyone have a document I can use as a guide on how to load a SIP image on a cisco 7960 phone? Ryan ___ 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 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] CIsco 7960 SIP Image
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/mgcp/frmwrup .pdf Quoting Preston Garrison [EMAIL PROTECTED]: www.voip-info.org has it Preston Garrison direct: 877-748-4142 fax: 310-774-3901 cell: 623-748-4140 -Original Message- From: Ryan Finnesey [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tue, 31 May 2005 11:18:47 -0400 Subject: [Asterisk-Users] CIsco 7960 SIP Image Does anyone have a document I can use as a guide on how to load a SIP image on a cisco 7960 phone? Ryan ___ 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 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 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] CIsco 7960 SIP Image
I posted this on my blog the other day: http://www.tracyphillips.com/2005/05/26/how-to-setup-a-cisco-7960g-with-sip/ Its mostly from memory so if it doesn't work, let me know and I will do what I can to help you out. --Tracy On 5/31/05, Ryan Finnesey [EMAIL PROTECTED] wrote: Does anyone have a document I can use as a guide on how to load a SIP image on a cisco 7960 phone? Ryan ___ 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 -- Tracy Phillips Weberize Inc. 800-677-1047 ___ 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] Cisco 7960 SIP Reject Call Option
Is there anyway of having a Reject Call button appear when there is an incoming call. Sometimes I am wating for a call, but one from another person comes through - I would like to press a button and send them straight to voicemail. Sort of a Dynamic Do Not Disturb ... :) Julian ___ 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] Cisco 7960 SIP Reject Call Option
Asterisk wrote: Is there anyway of having a Reject Call button appear when there is an incoming call. Sometimes I am wating for a call, but one from another person comes through - I would like to press a button and send them straight to voicemail. You can press the EndCall button while an unanswered call is ringing to achieve the same effect. ___ 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] Cisco 7960 SIP Reject Call Option
At 11:27 AM 5/1/2005, you wrote: Asterisk wrote: Is there anyway of having a Reject Call button appear when there is an incoming call. Sometimes I am wating for a call, but one from another person comes through - I would like to press a button and send them straight to voicemail. You can press the EndCall button while an unanswered call is ringing to achieve the same effect. The only available menu button is Answer when an inbound call is ringing on my 7960g. The menu with EndCall does not come up until I answer the call. Tom Sorry for jumping in but I am after the same thing. ___ ___ 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] Cisco 7960 SIP registration???
So, here's my quandary: 1) Asterisk running CVS HEAD as of a couple days ago 2) Cisco 7960 SIP phones in a different subnet than the Asterisk server 3) NAT/Firewall device between 7960's and * I can initiate a call from the 7960's just fine. They can call out using our Broadvoice account and access any of the vmail stuff on *. When calling in from the outside world and dialing one of their extensions, however, I always get a this user is on the phone message. The console spits out this nugget: == CDR updated on SIP/4252780761-933d -- Executing Macro(SIP/4252780761-933d, stdsip|tycisco|101) in new stack -- Executing Dial(SIP/4252780761-933d, SIP/tycisco) in new stack Apr 20 08:14:59 NOTICE[32728]: app_dial.c:973 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) == Everyone is busy/congested at this time (1:0/1/0) A showing of the sip peers: sip show peers Name/username HostDyn Nat ACL Mask Port Status rickcisco/cisco2 (Unspecified)D N 255.255.255.255 0UNKNOWN tycisco/cisco1 (Unspecified)D N 255.255.255.255 0UNKNOWN sip.broadvoice.com/425278 147.135.4.128 255.255.255.255 5060 OK (127 ms) 3 sip peers [1 online , 2 offline] I'm sure the reason I can't call to an extension is that they are appearing offline. How can I remedy this, however? I'm an * newbie, so go easy on me. :^) Thanks, Ty Christensen MCP, MCSP, MCSB Master Mind Productions Inc. www.mastermindpro.com http://www.mastermindpro.com/ (425) 378-7724 ___ 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] Cisco 7960 SIP registration???
Looks like you have sip.conf set up to expect registrations for tycisco since it has a D for dynamic. You can either set up the 7960 to register with asterisk and use something like this in sip.conf: [tycisco] type=friend username= someusername secret= somesecret insecure=no mailbox=757 host=dynamic callerid= or just not have the 7960 register and specify its IP address using the host= line instead. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of List Receiver Sent: Wednesday, April 20, 2005 11:19 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960 SIP registration??? So, here's my quandary: 1) Asterisk running CVS HEAD as of a couple days ago 2) Cisco 7960 SIP phones in a different subnet than the Asterisk server 3) NAT/Firewall device between 7960's and * I can initiate a call from the 7960's just fine. They can call out using our Broadvoice account and access any of the vmail stuff on *. When calling in from the outside world and dialing one of their extensions, however, I always get a this user is on the phone message. The console spits out this nugget: == CDR updated on SIP/4252780761-933d -- Executing Macro(SIP/4252780761-933d, stdsip|tycisco|101) in new stack -- Executing Dial(SIP/4252780761-933d, SIP/tycisco) in new stack Apr 20 08:14:59 NOTICE[32728]: app_dial.c:973 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) == Everyone is busy/congested at this time (1:0/1/0) A showing of the sip peers: sip show peers Name/username HostDyn Nat ACL Mask Port Status rickcisco/cisco2 (Unspecified)D N 255.255.255.255 0UNKNOWN tycisco/cisco1 (Unspecified)D N 255.255.255.255 0UNKNOWN sip.broadvoice.com/425278 147.135.4.128 255.255.255.255 5060 OK (127 ms) 3 sip peers [1 online , 2 offline] I'm sure the reason I can't call to an extension is that they are appearing offline. How can I remedy this, however? I'm an * newbie, so go easy on me. :^) Thanks, Ty Christensen MCP, MCSP, MCSB Master Mind Productions Inc. www.mastermindpro.com http://www.mastermindpro.com/ (425) 378-7724 ___ 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 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] Cisco 7960 SIP registration???
I've done that...I think. :^) Here's the excerpt from sip.conf: [tycisco] type=friend username=cisco1 secret=*** qualify=200 ; Qualify peer is no more than 200ms away nat=yes ;insecure=no host=dynamic; This device registers with us ;defaultip=192.168.0.30 canreinvite=no context=fullaccess dtmfmode=inband mailbox=101 disallow=all allow=ulaw allow=alaw allow=g729 I still get no registration when I do a sip show peers. Am I missing something simple? Thanks, Ty -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of end1r Sent: Wednesday, April 20, 2005 8:58 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Cisco 7960 SIP registration??? Looks like you have sip.conf set up to expect registrations for tycisco since it has a D for dynamic. You can either set up the 7960 to register with asterisk and use something like this in sip.conf: [tycisco] type=friend username= someusername secret= somesecret insecure=no mailbox=757 host=dynamic callerid= or just not have the 7960 register and specify its IP address using the host= line instead. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of List Receiver Sent: Wednesday, April 20, 2005 11:19 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960 SIP registration??? So, here's my quandary: 1) Asterisk running CVS HEAD as of a couple days ago 2) Cisco 7960 SIP phones in a different subnet than the Asterisk server 3) NAT/Firewall device between 7960's and * I can initiate a call from the 7960's just fine. They can call out using our Broadvoice account and access any of the vmail stuff on *. When calling in from the outside world and dialing one of their extensions, however, I always get a this user is on the phone message. The console spits out this nugget: == CDR updated on SIP/4252780761-933d -- Executing Macro(SIP/4252780761-933d, stdsip|tycisco|101) in new stack -- Executing Dial(SIP/4252780761-933d, SIP/tycisco) in new stack Apr 20 08:14:59 NOTICE[32728]: app_dial.c:973 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) == Everyone is busy/congested at this time (1:0/1/0) A showing of the sip peers: sip show peers Name/username HostDyn Nat ACL Mask Port Status rickcisco/cisco2 (Unspecified)D N 255.255.255.255 0UNKNOWN tycisco/cisco1 (Unspecified)D N 255.255.255.255 0UNKNOWN sip.broadvoice.com/425278 147.135.4.128 255.255.255.255 5060 OK (127 ms) 3 sip peers [1 online , 2 offline] I'm sure the reason I can't call to an extension is that they are appearing offline. How can I remedy this, however? I'm an * newbie, so go easy on me. :^) Thanks, Ty Christensen MCP, MCSP, MCSB Master Mind Productions Inc. www.mastermindpro.com http://www.mastermindpro.com/ (425) 378-7724 ___ 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 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 smime.p7s Description: S/MIME cryptographic signature ___ 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] cisco 7960 SIP setup
There's a long chapter in my book about re-programming the 7960 from skinny to SIP that might help you out. Figuring it out was non-trivial. You can get the book at Amazon. TKS, Paul Mahler I can't get the 7960 to reconfigure and work. I am a newbie to voip. I went through the list and read some other comments about the 7960 and unlocking it. It is a used 7960 that came with CallManager. I need to have SIP. I first reset the phone to factory defaults then I changed the TFTP server address in the settings. I have unlocked the phone with **# and it shows the lock as unlocked in the upper right hand corner. I was told that the phone should be able to download the SIP... file once the TFTP address was changed. So far nothing though. Any ideas? Mike Paul Mahler www.signate.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] cisco 7960 SIP setup
mk111 wrote: I was told that the phone should be able to download the SIP... file once the TFTP address was changed. So far nothing though. Any ideas? have you rebooted the phone after changing the tftp address ? -- Simone Cittadini IT Manager == COMVERT S.R.L. via F.lli Bressan, 21 20126 Milano - ITALY Tel +39.02.27006796(aspetta un beep)105 [EMAIL PROTECTED] http://www.comvert.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] cisco 7960 SIP setup
Yes, a few times. All it does is show the following on the screen: Configuring IP, then Configuring CM List then Defaulting Cm to TFTP server, then Opening 66.xx.xx.xx. Then it goes back to the beginning and repeats itself over and over. Mike On Apr 15, 2005, at 1:12 AM, Simone Cittadini wrote: mk111 wrote: I was told that the phone should be able to download the SIP... file once the TFTP address was changed. So far nothing though. Any ideas? have you rebooted the phone after changing the tftp address ? -- Simone Cittadini IT Manager == COMVERT S.R.L. via F.lli Bressan, 21 20126 Milano - ITALY Tel +39.02.27006796(aspetta un beep)105 [EMAIL PROTECTED] http://www.comvert.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 ___ 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] cisco 7960 SIP setup
What version of SIP are you trying to load mk111 wrote: Yes, a few times. All it does is show the following on the screen: Configuring IP, then Configuring CM List then Defaulting Cm to TFTP server, then Opening 66.xx.xx.xx. Then it goes back to the beginning and repeats itself over and over. Mike On Apr 15, 2005, at 1:12 AM, Simone Cittadini wrote: mk111 wrote: I was told that the phone should be able to download the SIP... file once the TFTP address was changed. So far nothing though. Any ideas? have you rebooted the phone after changing the tftp address ? -- Simone Cittadini IT Manager == COMVERT S.R.L. via F.lli Bressan, 21 20126 Milano - ITALY Tel +39.02.27006796(aspetta un beep)105 [EMAIL PROTECTED] http://www.comvert.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 ___ 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 -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[EMAIL PROTECTED] ___ 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] cisco 7960 SIP setup
Mike, Under status, what is the firmware version? You're looking for Appl Load ID, Boot Load ID and Version. Most likely you'll have to get a version 6 SIP image and then you'll be able to install the current 7.4 SIP image after that. In order to get these image files, you have to be a contract paying Cisco client to download them from Cisco's site. I just went through this on my 3 7940s, but I have them all converted over to SIP. You also need to run a TFTP server. I used Cisco's old TFTP program on my Windows XP Pro box. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mk111 Sent: Thursday, April 14, 2005 9:31 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] cisco 7960 SIP setup I can't get the 7960 to reconfigure and work. I am a newbie to voip. I went through the list and read some other comments about the 7960 and unlocking it. It is a used 7960 that came with CallManager. I need to have SIP. I first reset the phone to factory defaults then I changed the TFTP server address in the settings. I have unlocked the phone with **# and it shows the lock as unlocked in the upper right hand corner. I was told that the phone should be able to download the SIP... file once the TFTP address was changed. So far nothing though. Any ideas? Mike ___ 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 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] cisco 7960 SIP setup
I can't get the 7960 to reconfigure and work. I am a newbie to voip. I went through the list and read some other comments about the 7960 and unlocking it. It is a used 7960 that came with CallManager. I need to have SIP. I first reset the phone to factory defaults then I changed the TFTP server address in the settings. I have unlocked the phone with **# and it shows the lock as unlocked in the upper right hand corner. I was told that the phone should be able to download the SIP... file once the TFTP address was changed. So far nothing though. Any ideas? Mike ___ 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] cisco 7960 SIP setup
Mike: I know this sounds patronizing, but do you have the SIP image files? If so, what version? Per the Asterisk wiki page on the 7960/7940s, you may need to upgrade incrementally. Additionally, make sure you have the correct files in the root directory of your tftp server (for linux, this is probably /tftpboot). Also make sure that the tftp server works (you can test it from a linux client). Check the wiki out at http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx -Andy On 4/14/05, mk111 [EMAIL PROTECTED] wrote: I can't get the 7960 to reconfigure and work. I am a newbie to voip. I went through the list and read some other comments about the 7960 and unlocking it. It is a used 7960 that came with CallManager. I need to have SIP. I first reset the phone to factory defaults then I changed the TFTP server address in the settings. I have unlocked the phone with **# and it shows the lock as unlocked in the upper right hand corner. I was told that the phone should be able to download the SIP... file once the TFTP address was changed. So far nothing though. Any ideas? Mike ___ 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 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] Cisco 7960 SIP images
Sorry for the late followup, but I want to share my lovely Cisco experience. First, after placing orders for the $8 contracts with both CDW and INSIGHT and having both orders cancelled a week later (for some supplier problem), I went with the $74 contract from INSIGHT (CDW wanted $84, IIRC). I actually got that contract. Then, I tried to register the phone, only to find that the factory-applied serial number wasn't even in Cisco's database. (Another phone's serial number from the same purchase worked fine.) I actually had a Cisco customer support person tell me once you give us a valid serial number for the phone, we can open a case for the invalid serial number on the phone. I was speechless. I never had an issue with who owned the phone. I told Cisco it belonged to a client (true) and I didn't know who purchased it. They seemed fine with that. Upgrading old (Circa 2000) Cisco 7960 phones is a joy in itself. They don't actually follow any documented self-update procedure AND the procedure they do follow changes significantly by current firmware version. Plus, you can't upgrade directly from an old (v6) firmware version to a new version. tcpdump is your friend. Watch closely for the first file the phone wants. Edit that file. Keep in mind the phone may choose to ignore that file and look in SIPDefault.cnf or MCGDeulft.cnf instead. Having said all that, I firmly believe that the Cisco 7960s are BY FAR the BEST IP phones available. It's a real credit to Cisco's engineers, product designers, etc. that a product's setup/upgrade can be so completely horrific and people will still demand their product. cheers, glenn On Mon, March 28, 2005 12:54 pm, Rich Adamson said: As a side note to the above (in the US), the contract reseller is suppose to obtain the phone's serial number. If that serial number is not registered to the individual requesting the contract, the contract supposedly will not be issued. That process is apparently used to identify when used phones are sold via eBay (etc), and essentially says one does not have a valid software license therefore it cannot be placed on maintenance. (A software license cannot be transferred with the sale of a used phone or any of cisco's equipment.) That same process is used for all Cisco equipment, however some used equipment resellers have been able to find ways around it (one way or another). Once a maintenance contract number has been issued (regardless of whether its on a piece of paper or email), that contract number has to be entered into a cisco system that tracks the number against a customer account. If you don't have a customer account, that process can't be completed either. Some resellers will create your account for you and others won't. Once the account has been created and the contract recorded, then the customer is granted access to the download sections of their site via their login/authentication process. So the bottom line is the process requires a fair amount of manual labor and for $8 (in the US), few resellers have any interest in the sales commission resulting from an $8 sale. (Guess that says if you're buying 500 contracts, one might receive a different level of reseller interest.) Regardless of whether we like it or not, cisco wrote the license terms and asterisk users are not going to change their machine. It's obviously written to discourage reselling used equipment without paying a re-certification fee, and that re-certification re-license process can get to be far more costly then simply purchasing their new equipment. Surprise surprise! I don't work for cisco or any of their resellers. ___ 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] Cisco 7960 SIP images
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Chris W wrote: In a sense this cound be off-topic but I hope it isn't considered so. Apologies already if it is! Can anyone point me in the right direction to get new SIP images for the Cisco 7960 phone? I found P0S30202 around (ie v2.02) and it works but lacks a lot of the features the phone boasts so I'm looking for updates. I googled and found that you can get a support contract via 1-800-INSIGHT but guess what! They're in the US and won't issue licences outside the country. I'm in the Netherlands so that ain't gonna make matters easy. I guess I need v.3, 4, 5, 6 and 7 to get the latest stuff. What a lot of upgrading! Any pointers/help most welcome. Thanks in advance Unfortunatley, all the Cisco resellers in Europe I have approached don't seem to be interested in carrying these low value contracts (CON-SNT-CP7960 or CON-SNT-ATA186) or don't want to deal in such low volumes and have no method of dealing with such sales. Cisco want you to talk to their resellers, which brings you back right where you started. So to summarise: 1/ Cisco will not sell direct. 2/ North American Resellers will not sell to Europe. 3/ European Resellers do/will not sell single contracts What route is left for guy with a few Cisco phones in Europe? Piracy? /RANT - -- Ron Wellsted http://www.wellsted.org.uk [EMAIL PROTECTED] FWD:519961 Gossiptel:9309811 N 52.567623, W 2.137621 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iQEVAwUBQkfGYEtP/KMNOfRbAQKAZwgAi1QXt7d9Igy4o2dHG+qqG6KApixH01Xu x2lns+WvPwuDcHF5uBzJjfxGG4jVrgLtIg1la7M6P8Bu6u2nZQyz0fJk8UhVN4bp drsXHmjq44UyDel9Kn2Q6zvhfuND84qZTBAQ9MbLXnogQlg9vB067975P8rQ7+vK WX598aP0i5tDDMvhUNVZX/epYuIby0E6YdLwGaARcpcERWiQfG2tkY9EcVots1qt rcruHJZO4yutOwIY6irzmMpCShj+SShfRwNiI4+ggJIchUnaq+Il4ly4nMbDl2Px 5EZBWzECnQPRxeatKKyngZXUbMcFm9FozgLP7eHMol73QwlbWDjqfQ== =toHG -END PGP SIGNATURE- ___ 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] Cisco 7960 SIP images
On Monday 28 March 2005 09:54, Ron Wellsted wrote: [...] So to summarise: 1/ Cisco will not sell direct. 2/ North American Resellers will not sell to Europe. 3/ European Resellers do/will not sell single contracts What route is left for guy with a few Cisco phones in Europe? Piracy? /RANT I don't think http://www.s2s.ltd.uk/ care how little you buy. B ___ 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] Cisco 7960 SIP images
Ron Wellsted wrote: What route is left for guy with a few Cisco phones in Europe? Piracy? I looked around for nearly a year for a contract after a kind soul got me the images (the closest I got was a site in the US who were prepared to sell me the CON-SNT-CP7960 for £8 ... with £150 Postage!!!)... eventually gave up and ordered a CON-SNT-PKG1 package from lanway which I managed to get for £42. Of course being a Cisco contract it still hasn't arrived 2.5 weeks later. Cisco are the first company I've ever come across who seem to actively resent having customers and would rather you went with someone else. Tony ___ 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] Cisco 7960 SIP images
On Monday 28 March 2005 14:58, Tony Hoyle wrote: Ron Wellsted wrote: What route is left for guy with a few Cisco phones in Europe? Piracy? I looked around for nearly a year for a contract after a kind soul got me the images (the closest I got was a site in the US who were prepared to sell me the CON-SNT-CP7960 for £8 ... with £150 Postage!!!)... eventually gave up and ordered a CON-SNT-PKG1 package from lanway which I managed to get for £42. Of course being a Cisco contract it still hasn't arrived 2.5 weeks later. Cisco are the first company I've ever come across who seem to actively resent having customers and would rather you went with someone else. It doesn't arrive. It's all done instantly via email. B ___ 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] Cisco 7960 SIP images
Bob Goddard wrote: It doesn't arrive. It's all done instantly via email. There's a whole package apparently (hence the £150 postage I was quoted, although I suspect they just weren't interested in selling). Even the entry on voip-info.org says it takes two weeks... Once you buy it the request goes to Cisco who have to get off their backsides and actually issue you with the thing. Nothing yet, although I'll be chasing it again tomorrow (unfortunately it's impossible to chase it directly with cisco as they refuse to deal with mere customers). I've come *so* close to putting the phone on ebay and forgetting about it. Certainly I'll never buy a cisco product again. Tony ___ 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] Cisco 7960 SIP images
If you call Cisco contract support. 1-800-447-9347 and give them the serial number used when you purchased the smartnet they will give you the contract number over the phone. If the contract was sold properly the reseller would have asked you for the serial number of the unit and turned that into Cisco. Cisco should have then emailed the contract number to you. My experience has been they only email you about half the time and you have to call them the other half. Henry - Original Message - From: Tony Hoyle [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 28, 2005 10:06 AM Subject: Re: [Asterisk-Users] Cisco 7960 SIP images Bob Goddard wrote: It doesn't arrive. It's all done instantly via email. There's a whole package apparently (hence the £150 postage I was quoted, although I suspect they just weren't interested in selling). Even the entry on voip-info.org says it takes two weeks... Once you buy it the request goes to Cisco who have to get off their backsides and actually issue you with the thing. Nothing yet, although I'll be chasing it again tomorrow (unfortunately it's impossible to chase it directly with cisco as they refuse to deal with mere customers). I've come *so* close to putting the phone on ebay and forgetting about it. Certainly I'll never buy a cisco product again. Tony ___ 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 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] Cisco 7960 SIP images
It doesn't arrive. It's all done instantly via email. There's a whole package apparently (hence the £150 postage I was quoted, although I suspect they just weren't interested in selling). Even the entry on voip-info.org says it takes two weeks... Once you buy it the request goes to Cisco who have to get off their backsides and actually issue you with the thing. Nothing yet, although I'll be chasing it again tomorrow (unfortunately it's impossible to chase it directly with cisco as they refuse to deal with mere customers). I've come *so* close to putting the phone on ebay and forgetting about it. Certainly I'll never buy a cisco product again. As a side note to the above (in the US), the contract reseller is suppose to obtain the phone's serial number. If that serial number is not registered to the individual requesting the contract, the contract supposedly will not be issued. That process is apparently used to identify when used phones are sold via eBay (etc), and essentially says one does not have a valid software license therefore it cannot be placed on maintenance. (A software license cannot be transferred with the sale of a used phone or any of cisco's equipment.) That same process is used for all Cisco equipment, however some used equipment resellers have been able to find ways around it (one way or another). Once a maintenance contract number has been issued (regardless of whether its on a piece of paper or email), that contract number has to be entered into a cisco system that tracks the number against a customer account. If you don't have a customer account, that process can't be completed either. Some resellers will create your account for you and others won't. Once the account has been created and the contract recorded, then the customer is granted access to the download sections of their site via their login/authentication process. So the bottom line is the process requires a fair amount of manual labor and for $8 (in the US), few resellers have any interest in the sales commission resulting from an $8 sale. (Guess that says if you're buying 500 contracts, one might receive a different level of reseller interest.) Regardless of whether we like it or not, cisco wrote the license terms and asterisk users are not going to change their machine. It's obviously written to discourage reselling used equipment without paying a re-certification fee, and that re-certification re-license process can get to be far more costly then simply purchasing their new equipment. Surprise surprise! I don't work for cisco or any of their resellers. ___ 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] Cisco 7960 SIP images
Henry Devito wrote: If you call Cisco contract support. 1-800-447-9347 and give them the serial number used when you purchased the smartnet they will give you the contract number over the phone. If the contract was sold properly No serial number was asked for.. I just explained that I just wanted the smartnet contract and they took my credit card details. Presumably not all dealers work the way cisco would like them to. TBH I'm not even sure I know the serial of that phone - threw the box away months ago. Tony ___ 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] Cisco 7960 SIP images
Serial number is on the bottom of phone. Email me off list I will help. - Original Message - From: Tony Hoyle [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 28, 2005 12:02 PM Subject: Re: [Asterisk-Users] Cisco 7960 SIP images Henry Devito wrote: If you call Cisco contract support. 1-800-447-9347 and give them the serial number used when you purchased the smartnet they will give you the contract number over the phone. If the contract was sold properly No serial number was asked for.. I just explained that I just wanted the smartnet contract and they took my credit card details. Presumably not all dealers work the way cisco would like them to. TBH I'm not even sure I know the serial of that phone - threw the box away months ago. Tony ___ 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 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] Cisco 7960 SIP 7.4
On Mon, 14 Mar 2005 08:06:20 -0800, Scott Laird [EMAIL PROTECTED] wrote: I don't see any major changes in the release notes--mostly small bug fixes. They fixed some DHCP and NTP problems, as well as a 802.1x problem with some of their switches. There were a couple SIP protocol fixes in there too, plus a spelling fix. Has anyone else upgraded to 7.4 and found that the date time no longer appears on the phone? Ie: The phone doesn't appear to be grabbing the date time off the NTP server on my network, it worked alright on 7.3 (except for the time drift) but now they seem to have fixed the drift by no longer displaying time nor date. ___ 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] Cisco 7960 SIP 7.4
I recently upgraded to 7.4 and the time setting continued to work. You say you upgraded and still have the exact same SIPDefault.cnf and SIPMAC.cnf that worked in 7.3? Chad Brown - IdentityMine -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Lee Sent: Sunday, March 27, 2005 2:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 SIP 7.4 On Mon, 14 Mar 2005 08:06:20 -0800, Scott Laird [EMAIL PROTECTED] wrote: I don't see any major changes in the release notes--mostly small bug fixes. They fixed some DHCP and NTP problems, as well as a 802.1x problem with some of their switches. There were a couple SIP protocol fixes in there too, plus a spelling fix. Has anyone else upgraded to 7.4 and found that the date time no longer appears on the phone? Ie: The phone doesn't appear to be grabbing the date time off the NTP server on my network, it worked alright on 7.3 (except for the time drift) but now they seem to have fixed the drift by no longer displaying time nor date. ___ 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 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] Cisco 7960 SIP 7.4
On Sun, 27 Mar 2005 20:06:39 +1000, Chris Lee [EMAIL PROTECTED] wrote: Ie: The phone doesn't appear to be grabbing the date time off the NTP server on my network, it worked alright on 7.3 (except for the time drift) but now they seem to have fixed the drift by no longer displaying time nor date. Problem sorted... something is wrong with my local NTP server, I've now changed my config to get the time off my ISP's NTP server and it's working fine (note to self: make sure you use the IP address for the server and not a DNS name). ___ 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] Cisco 7960 SIP 7.4
Chris Lee wrote: On Mon, 14 Mar 2005 08:06:20 -0800, Scott Laird [EMAIL PROTECTED] wrote: Has anyone else upgraded to 7.4 and found that the date time no longer appears on the phone? Chris, As someone pointed out earlier, change your sntp_mode to unicast in your SIPmacaddress.cnf as such: sntp_mode: unicast Doug ___ 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] Cisco 7960 SIP 7.4
Chris Lee wrote: Has anyone else upgraded to 7.4 and found that the date time no longer appears on the phone? This problem was pointed at the SIPPhoneReleaseNotes7_4.pdf file. What I noticed is that when the phone lost the internet connection the date/time will no longer be present on the phone. ___ 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] Cisco 7960 SIP images
In a sense this cound be off-topic but I hope it isn't considered so. Apologies already if it is! Can anyone point me in the right direction to get new SIP images for the Cisco 7960 phone? I found P0S30202 around (ie v2.02) and it works but lacks a lot of the features the phone boasts so I'm looking for updates. I googled and found that you can get a support contract via 1-800-INSIGHT but guess what! They're in the US and won't issue licences outside the country. I'm in the Netherlands so that ain't gonna make matters easy. I guess I need v.3, 4, 5, 6 and 7 to get the latest stuff. What a lot of upgrading! Any pointers/help most welcome. Thanks in advance Chris -- Chris's lists go to lists at mokum dot org ___ 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] Cisco 7960 SIP boot takes 2 minutes?
Two minutes seems like a long time to initialize a Cisco 7960 IP phone. What times are others seeing for the load when you reboot a phone? We are running the SIP 7.4 load. Our * 1.0 stable is also our http, dhcp and tftp server. During boot, the display shows: Configuring VLAN 100 seconds TFTP SIP loads a few seconds back to Configuring VLAN the rest of the time. I am wondering if I have a network problem or could do something to speed this up. Thanks, Tom ___ 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] Cisco 7960 SIP boot takes 2 minutes?
Tom wrote: Configuring VLAN 100 seconds TFTP SIP loads a few seconds back to Configuring VLAN the rest of the time. Roughly the same there here as well. 7940 boots faster, but not by much. Doug ___ 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] Cisco 7960 SIP boot takes 2 minutes?
Tom wrote: Configuring VLAN 100 seconds TFTP SIP loads a few seconds back to Configuring VLAN the rest of the time. That's about normal; I wish Cisco would let us turn off CDP in these phones, it would help tremendously. ___ 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] Cisco 7960 SIP boot takes 2 minutes?
Tom wrote: What times are others seeing for the load when you reboot a phone? About the same here, but I don't care as I never reboot my phone (about once every month or two). ___ 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] Cisco 7960 SIP boot takes 2 minutes?
--On Monday, March 21, 2005 10:29 AM +0900 Hermann Wecke [EMAIL PROTECTED] wrote: Tom wrote: What times are others seeing for the load when you reboot a phone? About the same here, but I don't care as I never reboot my phone (about once every month or two). Our 40's and 60s both take about two minutes to load...the spend/waste a lot of time waiting on the alternate VLAN config stuff. I'd imagine if oyu had a 'fully' voice setup 2940 or 2950 that would advertise those settings via CDP for the phone it'd fire up quicker w/o waiting on the timeouts. I can't do that in my network as we have several dumb switches. ___ 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] Cisco 7960 SIP boot takes 2 minutes?
We have the same problem - started when we upgraded to 7.1. It isn't too much of a bother for us though, because the phones (once configured) are left alone. ___ 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] Cisco 7960 SIP Firmware
I would suggest contacting a dealer until you find one who will sell you a maintenance contract for the phone. Last I checked, over a year ago, they were somewhere around $10-$20. Once you have a contract you may register online and download all the software you need. However to use legally you would have the pay the ~$50 license fee for SIP. On Mar 18, 2005, at 10:17 PM, Patrick M. Gray, Jr. wrote: That seems to be what the various documents I've stumbled across seem to indicate. Maybe it's too late at night and my brain is shot, but Cisco's documentation on the upgrade path seems a little confusing... Would you mind giving me a brief summary of the upgrade path to the latest firmware if you know it, starting from P003AM30? Thanks again! Pat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pedro Sent: Friday, 18 March, 2005 22:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 SIP Firmware Just a note that you will need to perform quite a few incremental upgrades to get to a current firmware version. So if you do get someone who will sell you the firmware, make sure you get the all of them. On Fri, 18 Mar 2005 22:04:29 -0500, Patrick M. Gray, Jr. [EMAIL PROTECTED] wrote: I got a new old stock Cisco 7960 from eBay and the warranty expired bay in 2001 according to Cisco (I didn't even know they had VoIP in 2001 ;-) ). I spoke with a wonderfully rude gentleman at Cisco who told me there was nothing that could be done to get SIP firmware for the device, and would not even entertain the possibility of purchasing said FW from Cisco. He suggested I call a local reseller, and the single one I called was not interested in helping me either with my unsupported hardware. I'm using the 7960 to experiment with *, and was wondering if there are alternative means to finding the firmware, or if the out of the box SCCP firmware (I have version P003AM30) will work with *. I'm willing to pay any official resellers a fair price for the F/W, but the attitude I received from Cisco and the one reseller I contacted have me thinking this is a waste of time. I'm using [EMAIL PROTECTED] and can seem to find any SCCP info, and don't want to delve too deeply into this experiment if the phone is not going to work reliably. Thanks for any help or pointers in the right direction. Pat ___ 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 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 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 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] Cisco 7960 SIP Firmware
Jerry wrote: I would suggest contacting a dealer until you find one who will sell you a maintenance contract for the phone. Last I checked, over a year ago, they were somewhere around $10-$20. Once you have a contract you may register online and download all the software you need. However to use legally you would have the pay the ~$50 license fee for SIP. That is only true if your phone is a 7940/7960 (not G). The 7940G/7960G are legally allowed to be used with SIP firmware, if you have rights to obtain it. ___ 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] Cisco 7960 SIP Firmware
I wonder if VoipSupply can sell the maintenance contract for the phone? Wouldn't hurt to ask. The fellow from VS is a regular poster over on the asterisk-biz list. --On Saturday, March 19, 2005 11:09 AM -0600 Jerry [EMAIL PROTECTED] wrote: I would suggest contacting a dealer until you find one who will sell you a maintenance contract for the phone. Last I checked, over a year ago, they were somewhere around $10-$20. Once you have a contract you may register online and download all the software you need. However to use legally you would have the pay the ~$50 license fee for SIP. On Mar 18, 2005, at 10:17 PM, Patrick M. Gray, Jr. wrote: That seems to be what the various documents I've stumbled across seem to indicate. Maybe it's too late at night and my brain is shot, but Cisco's documentation on the upgrade path seems a little confusing... Would you mind giving me a brief summary of the upgrade path to the latest firmware if you know it, starting from P003AM30? Thanks again! Pat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pedro Sent: Friday, 18 March, 2005 22:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 SIP Firmware Just a note that you will need to perform quite a few incremental upgrades to get to a current firmware version. So if you do get someone who will sell you the firmware, make sure you get the all of them. On Fri, 18 Mar 2005 22:04:29 -0500, Patrick M. Gray, Jr. [EMAIL PROTECTED] wrote: I got a new old stock Cisco 7960 from eBay and the warranty expired bay in 2001 according to Cisco (I didn't even know they had VoIP in 2001 ;-) ). I spoke with a wonderfully rude gentleman at Cisco who told me there was nothing that could be done to get SIP firmware for the device, and would not even entertain the possibility of purchasing said FW from Cisco. He suggested I call a local reseller, and the single one I called was not interested in helping me either with my unsupported hardware. I'm using the 7960 to experiment with *, and was wondering if there are alternative means to finding the firmware, or if the out of the box SCCP firmware (I have version P003AM30) will work with *. I'm willing to pay any official resellers a fair price for the F/W, but the attitude I received from Cisco and the one reseller I contacted have me thinking this is a waste of time. I'm using [EMAIL PROTECTED] and can seem to find any SCCP info, and don't want to delve too deeply into this experiment if the phone is not going to work reliably. Thanks for any help or pointers in the right direction. Pat ___ 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 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 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 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 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] Cisco 7960 SIP Firmware
I'm ex cisco (left in '96). Cisco did change their corp policy recently in that they no longer will sell firmware directly to end users. I think as far as the phones go, that's a mistake on cisco's part. Perhaps we'll see cisco move some of these phones over to the linksys side someday. Any cisco product/marketing managers out there? You also need to realize cisco is primarily a software company. It's family jewels is IOS. That's why they license for each device. It would be nice to locate a cisco dealer who would be willing to sell service contracts/TAC logins for single units - so far I too have been able to find one. If found it would be great to add it to the wiki I just flashed a 7960G from voipsupply. It came with skinny. The SIP path I took was: 3.0, 4.2, 4.4, 5.3, 6.0, 6.3, 7.1, 7.2, 7.3. Don't know if ALL these flashes are really nessessary, but I had 'um so I did 'um. Word of caution: the 7.x firmware sent to me by voipsuppy was incomplete as to the 7.x releases. They left out the .loads files. Without those files you'll end up with the dreaded application load failure. BTW: The service from voipsupply was great otherwise. -- John Breeden Hawaii Ed Greenberg wrote: I wonder if VoipSupply can sell the maintenance contract for the phone? Wouldn't hurt to ask. The fellow from VS is a regular poster over on the asterisk-biz list. --On Saturday, March 19, 2005 11:09 AM -0600 Jerry [EMAIL PROTECTED] wrote: I would suggest contacting a dealer until you find one who will sell you a maintenance contract for the phone. Last I checked, over a year ago, they were somewhere around $10-$20. Once you have a contract you may register online and download all the software you need. However to use legally you would have the pay the ~$50 license fee for SIP. On Mar 18, 2005, at 10:17 PM, Patrick M. Gray, Jr. wrote: That seems to be what the various documents I've stumbled across seem to indicate. Maybe it's too late at night and my brain is shot, but Cisco's documentation on the upgrade path seems a little confusing... Would you mind giving me a brief summary of the upgrade path to the latest firmware if you know it, starting from P003AM30? Thanks again! Pat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pedro Sent: Friday, 18 March, 2005 22:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 SIP Firmware Just a note that you will need to perform quite a few incremental upgrades to get to a current firmware version. So if you do get someone who will sell you the firmware, make sure you get the all of them. On Fri, 18 Mar 2005 22:04:29 -0500, Patrick M. Gray, Jr. [EMAIL PROTECTED] wrote: I got a new old stock Cisco 7960 from eBay and the warranty expired bay in 2001 according to Cisco (I didn't even know they had VoIP in 2001 ;-) ). I spoke with a wonderfully rude gentleman at Cisco who told me there was nothing that could be done to get SIP firmware for the device, and would not even entertain the possibility of purchasing said FW from Cisco. He suggested I call a local reseller, and the single one I called was not interested in helping me either with my unsupported hardware. I'm using the 7960 to experiment with *, and was wondering if there are alternative means to finding the firmware, or if the out of the box SCCP firmware (I have version P003AM30) will work with *. I'm willing to pay any official resellers a fair price for the F/W, but the attitude I received from Cisco and the one reseller I contacted have me thinking this is a waste of time. I'm using [EMAIL PROTECTED] and can seem to find any SCCP info, and don't want to delve too deeply into this experiment if the phone is not going to work reliably. Thanks for any help or pointers in the right direction. Pat ___ 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 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 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 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 mailing list
Re: [Asterisk-Users] Cisco 7960 SIP Firmware
We can probably help you out with purchasing a Smartnet contract for your CP-7960 phone, although we are not an authorized Cisco reseller. We are Polycom authorized, and represent myriad other vendors as well, but in terms of Cisco products we have never been Cisco authorized. Cisco has recently changed the licensing distribution model for all of their phones. They are no longer currently selling the Spare version of the Cisco phones. From what I have been told and what I have seen on Ingram and TechData, the phones are only available in the CH1 (CallManager) and CCME (CallManager Express) flavors. The new licensing program, as it was explained to me, will force distribution buyers who purchase any Cisco phones to also purchase a $150 SIP/MGCP license, this adds $150 to the list price of any model you purchase. They are supposed to be releasing a new SP service provider edition of each phone model, which also will require the $150 SIP/MGCP license. The odd part is I have been told this SIP/MGCP license is a requirement for any version you buy, CH1 (CallManager), CCME (CallManager Express) or the SP (Service Provider) edition. If someone expressly wants to purchase the CH1 or CCME versions of the phones, they must be using CallManager or CM Express right? Why else would they buy that version other than to have the correct licensing for their Cisco PBX. If you are using CallManager or CM Express, why would you need a $150 SIP/MGCP license, when your PBX runs Cisco Skinny protocol, not SIP or MGCP. Perhaps there is a Cisco telephony authorized firm on this list who can shed some light on that seemingly illogical requirement. Cory @ VOIPSupply.com +++ I wonder if VoipSupply can sell the maintenance contract for the phone? Wouldn't hurt to ask. The fellow from VS is a regular poster over on the asterisk-biz list. --On Saturday, March 19, 2005 11:09 AM -0600 Jerry [EMAIL PROTECTED] wrote: I would suggest contacting a dealer until you find one who will sell you a maintenance contract for the phone. Last I checked, over a year ago, they were somewhere around $10-$20. Once you have a contract you may register online and download all the software you need. However to use legally you would have the pay the ~$50 license fee for SIP. On Mar 18, 2005, at 10:17 PM, Patrick M. Gray, Jr. wrote: That seems to be what the various documents I've stumbled across seem to indicate. Maybe it's too late at night and my brain is shot, but Cisco's documentation on the upgrade path seems a little confusing... Would you mind giving me a brief summary of the upgrade path to the latest firmware if you know it, starting from P003AM30? Thanks again! Pat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pedro Sent: Friday, 18 March, 2005 22:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 SIP Firmware Just a note that you will need to perform quite a few incremental upgrades to get to a current firmware version. So if you do get someone who will sell you the firmware, make sure you get the all of them. On Fri, 18 Mar 2005 22:04:29 -0500, Patrick M. Gray, Jr. [EMAIL PROTECTED] wrote: I got a new old stock Cisco 7960 from eBay and the warranty expired bay in 2001 according to Cisco (I didn't even know they had VoIP in 2001 ;-) ). I spoke with a wonderfully rude gentleman at Cisco who told me there was nothing that could be done to get SIP firmware for the device, and would not even entertain the possibility of purchasing said FW from Cisco. He suggested I call a local reseller, and the single one I called was not interested in helping me either with my unsupported hardware. I'm using the 7960 to experiment with *, and was wondering if there are alternative means to finding the firmware, or if the out of the box SCCP firmware (I have version P003AM30) will work with *. I'm willing to pay any official resellers a fair price for the F/W, but the attitude I received from Cisco and the one reseller I contacted have me thinking this is a waste of time. I'm using [EMAIL PROTECTED] and can seem to find any SCCP info, and don't want to delve too deeply into this experiment if the phone is not going to work reliably. Thanks for any help or pointers in the right direction. Pat ___ 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 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
Re: [Asterisk-Users] Cisco 7960 SIP Firmware
[EMAIL PROTECTED] wrote: Cisco has recently changed the licensing distribution model for all of their phones. They are no longer currently selling the Spare version of the Cisco phones. I was told by Ingram Spain that they could only sell me the 'spare' version if I also purchased a CallManager license with it, which IMHO beats the purpose of it being called 'spare'. So, apparently, each phone is tied to it's license so-to-speak and the concept of 'spare' becomes rather vague. The new licensing program, as it was explained to me, will force distribution buyers who purchase any Cisco phones to also purchase a $150 SIP/MGCP license, this adds $150 to the list price of any model you purchase. If this is so, I expect to see Cisco phone sales decline. I was told by Cisco Spain that I had to supply the details of *my* end client to them, for quality assurance purposes, so that they can call the client and tell them how good a dealer I am (literally!). I imagine if I were to become a bad dealer, they could also phone all my client portfolio and direct them to an alternative good dealer. I ended up purchasing the phones from a distributor who didn't ask me any questions. In any case, it may well be the last Cisco phones I purchase. They are supposed to be releasing a new SP service provider edition of each phone model, which also will require the $150 SIP/MGCP license. I bet they wish we all pulled our trousers further up so they could tighten the belt and squeeze our necks a bit more. SNIP Perhaps there is a Cisco telephony authorized firm on this list who can shed some light on that seemingly illogical requirement. Er...Cisco's logic IMHO is inverted - I was also told by Cisco that they are now targeting small and medium-size bussiness, I presume because their growth potential in large companies is getting close to zero. I don't see how this policy, which seems clearly aimed at making you purchase their very expensive PBX solutions and their now more expensive phones in favour of cheaper PBX that can also work with their phones, ties up with the statements I got from them. Eventually, they are going to be fighting decent taiwanese imports with very cheap PBX systems, and I don't think many small or medium companies will have the slightest doubts on what is more cost effective. Regards, thanks for the information, Mike ___ 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] Cisco 7960 SIP Firmware
Just a note that you will need to perform quite a few incremental upgrades to get to a current firmware version. So if you do get someone who will sell you the firmware, make sure you get the all of them. On Fri, 18 Mar 2005 22:04:29 -0500, Patrick M. Gray, Jr. [EMAIL PROTECTED] wrote: I got a new old stock Cisco 7960 from eBay and the warranty expired bay in 2001 according to Cisco (I didn't even know they had VoIP in 2001 ;-) ). I spoke with a wonderfully rude gentleman at Cisco who told me there was nothing that could be done to get SIP firmware for the device, and would not even entertain the possibility of purchasing said FW from Cisco. He suggested I call a local reseller, and the single one I called was not interested in helping me either with my unsupported hardware. I'm using the 7960 to experiment with *, and was wondering if there are alternative means to finding the firmware, or if the out of the box SCCP firmware (I have version P003AM30) will work with *. I'm willing to pay any official resellers a fair price for the F/W, but the attitude I received from Cisco and the one reseller I contacted have me thinking this is a waste of time. I'm using [EMAIL PROTECTED] and can seem to find any SCCP info, and don't want to delve too deeply into this experiment if the phone is not going to work reliably. Thanks for any help or pointers in the right direction. Pat ___ 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 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] Cisco 7960 SIP Firmware
That seems to be what the various documents I've stumbled across seem to indicate. Maybe it's too late at night and my brain is shot, but Cisco's documentation on the upgrade path seems a little confusing... Would you mind giving me a brief summary of the upgrade path to the latest firmware if you know it, starting from P003AM30? Thanks again! Pat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pedro Sent: Friday, 18 March, 2005 22:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 SIP Firmware Just a note that you will need to perform quite a few incremental upgrades to get to a current firmware version. So if you do get someone who will sell you the firmware, make sure you get the all of them. On Fri, 18 Mar 2005 22:04:29 -0500, Patrick M. Gray, Jr. [EMAIL PROTECTED] wrote: I got a new old stock Cisco 7960 from eBay and the warranty expired bay in 2001 according to Cisco (I didn't even know they had VoIP in 2001 ;-) ). I spoke with a wonderfully rude gentleman at Cisco who told me there was nothing that could be done to get SIP firmware for the device, and would not even entertain the possibility of purchasing said FW from Cisco. He suggested I call a local reseller, and the single one I called was not interested in helping me either with my unsupported hardware. I'm using the 7960 to experiment with *, and was wondering if there are alternative means to finding the firmware, or if the out of the box SCCP firmware (I have version P003AM30) will work with *. I'm willing to pay any official resellers a fair price for the F/W, but the attitude I received from Cisco and the one reseller I contacted have me thinking this is a waste of time. I'm using [EMAIL PROTECTED] and can seem to find any SCCP info, and don't want to delve too deeply into this experiment if the phone is not going to work reliably. Thanks for any help or pointers in the right direction. Pat ___ 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 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 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] Cisco 7960 SIP 7.4
For those that are interested, I was just out on the Cisco site and noticed that they had released firmware 7.4 as of March 11th for the 7940/7960 phones. Doug ___ 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] Cisco 7960 SIP 7.4
On Mar 14, 2005, at 5:20 AM, Doug Lytle wrote: For those that are interested, I was just out on the Cisco site and noticed that they had released firmware 7.4 as of March 11th for the 7940/7960 phones. I don't see any major changes in the release notes--mostly small bug fixes. They fixed some DHCP and NTP problems, as well as a 802.1x problem with some of their switches. There were a couple SIP protocol fixes in there too, plus a spelling fix. In other words, if things are working for you right now, there's probably no reason to upgrade. Scott ___ 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] 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
RE: [Asterisk-Users] Cisco 7960 (SIP) hold problems
The first example wasn't even touching SER.. 7960sip -- asterisk -- IAX2 -- PRI :/ -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Thursday, December 16, 2004 9:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [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! I'm not aware of any issues. One remote internet based with g729 and nat, another with g711, and several local. If its happening here, no one knows about it. We're not using SER though. ___ 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
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
RE: [Asterisk-Users] Cisco 7960 (SIP) hold problems
Anyone??? -Original Message- From: Matt Schulte Sent: Thursday, December 16, 2004 10:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Cisco 7960 (SIP) hold problems The first example wasn't even touching SER.. 7960sip -- asterisk -- IAX2 -- PRI :/ -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Thursday, December 16, 2004 9:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [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! I'm not aware of any issues. One remote internet based with g729 and nat, another with g711, and several local. If its happening here, no one knows about it. We're not using SER though. ___ 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 ___ 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] 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! I'm not aware of any issues. One remote internet based with g729 and nat, another with g711, and several local. If its happening here, no one knows about it. We're not using SER though. ___ 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] Cisco 7960 SIP + 7914
I found a few mentions of the 7914 being used with Asterisk, these all covered SCCP/skinny though. Does anyone know if the 7914 can even be used with SIP? If so, any pointers? Is it a services thing? Anyone get the operator (line/extension status) to work with it. Thanks for the help, Cisco doesn't even mention ANYTHING about SIP + the 7914. Matt ___ 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] Cisco 7960 SIP + 7914
Thanks for the info -Original Message- From: Jeffrey C. Ollie [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 15, 2004 12:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 SIP + 7914 On Wed, 2004-12-15 at 11:54 -0600, Matt Schulte wrote: I found a few mentions of the 7914 being used with Asterisk, these all covered SCCP/skinny though. Does anyone know if the 7914 can even be used with SIP? If so, any pointers? Is it a services thing? Anyone get the operator (line/extension status) to work with it. Thanks for the help, Cisco doesn't even mention ANYTHING about SIP + the 7914. The 7914 is not supported by Cisco's SIP code. If you look at the data sheet under System Requirements is says that you need Cisco CallManager, which implies SCCP/skinny: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0 9186a008008883d.html Jeff ___ 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] Cisco 7960 SIP + 7914
On Wed, 2004-12-15 at 11:54 -0600, Matt Schulte wrote: I found a few mentions of the 7914 being used with Asterisk, these all covered SCCP/skinny though. Does anyone know if the 7914 can even be used with SIP? If so, any pointers? Is it a services thing? Anyone get the operator (line/extension status) to work with it. Thanks for the help, Cisco doesn't even mention ANYTHING about SIP + the 7914. The 7914 is not supported by Cisco's SIP code. If you look at the data sheet under System Requirements is says that you need Cisco CallManager, which implies SCCP/skinny: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008883d.html Jeff signature.asc Description: This is a digitally signed message part ___ 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] Cisco 7960 SIP V6 and distinctive ring.
Scott
I managed to get the line working.but I can't hear a difference in cadence.
I read in the wiki there is a bug logged with cisco to make distinctive ring more
distinctive so i'm gonna wait till then before pursuing it further.
I'm going to focus on xml services in the short termgod these phones are powerful.
Thanks for your help.
P
-Original Message-
From: Scott Laird [mailto:[EMAIL PROTECTED]
Sent: Monday, July 19, 2004, 11:53 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco 7960 SIP V6 and distinctive ring.
On Jul 19, 2004, at 9:29 AM, [EMAIL PROTECTED] wrote:
Hi
Can anyone with distinctive ring on their 7960's possibly post how
they've got it to work?
I understand that the ALERT_INFO variable is involved but using the
examples for the variable value from the WiKi I'm just getting an
error message from the Asterisk concole.
I'm setting it to 'Bellcore-dr1' through 'Bellcore-dr4'. I'm grabbing
the value out of Asterisk's database and sticking it into ALERT_INFO
like this:
[macro-setalertinfo]
exten = s,1,DBGet(ALERT_INFO=distinctivering/${CALLERIDNUM})
Works fine for me. You should also be able to do
'SetVar(ALERT_INFO=Bellcore-dr1)' without problems. Can you show us
the line that's generating errors?
Scott
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[Asterisk-Users] Cisco 7960 SIP V6 and distinctive ring.
Hi Can anyone with distinctive ring on their 7960's possibly post how they've got it to work? I understand that the ALERT_INFO variable is involved but using the examples for the variable value from the WiKi I'm just getting an error message from the Asterisk concole. Thanks in advance. P ___ 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] Cisco 7960 SIP V6 and distinctive ring.
On Jul 19, 2004, at 9:29 AM, [EMAIL PROTECTED] wrote:
Hi
Can anyone with distinctive ring on their 7960's possibly post how
they've got it to work?
I understand that the ALERT_INFO variable is involved but using the
examples for the variable value from the WiKi I'm just getting an
error message from the Asterisk concole.
I'm setting it to 'Bellcore-dr1' through 'Bellcore-dr4'. I'm grabbing
the value out of Asterisk's database and sticking it into ALERT_INFO
like this:
[macro-setalertinfo]
exten = s,1,DBGet(ALERT_INFO=distinctivering/${CALLERIDNUM})
Works fine for me. You should also be able to do
'SetVar(ALERT_INFO=Bellcore-dr1)' without problems. Can you show us
the line that's generating errors?
Scott
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Re: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
Hiya! Looks like you have the same problem as I had... found the answer by doing a 'debug sip-messages' by telnet'ing into one of my cisco phones... The short answer is 'its your callerid= line' you need to remove the quotes around the text part. The cisco's cant handle it. eg where you have for [phone1] in your Sip.conf callerid=Lounge1 1 what you should have is callerid=Lounge1 1 etc... Threw me for a while but the debug options on the cisco's helped out there... I think the docs read like you should have the text in quotes - but as I said - my cisco's didnt like it :) anyways - hope this helps :) Wayne! [EMAIL PROTECTED] wrote: Hi Sean Both phones are set for context=sip in the sip.conf file. As I say the phones will both call out OK (I can dial the 500 test number and successfully connect to the remote PBX through my firewall). It's just that when I'm trying to call from phone to phone I'm getting the 404 not found error in the asteris verbose dialog. If anyone has a documented example of their 7960 config sipdefault.cnf and sipxipadd.cnf files together with their sip.conf and extensions.conf files I could have to test directly on my system I'd be appreciative to test them on my system. While the WiKi's are very useful as example files it would be great (and I may do it myself!!) if there was an up to date example file with all the options for each filed and a verbose description for the rational behind it (although I recognise that this is an 'in development' product and therefore the docs have to be done at the end!!). Part of the problem is there are so many dependencies that can affect the system including how the dhpcd server serves IP address's and associated files (for example the files have to be structured in a particular order on the tftpd server for the cisco's to pick them up correctly). Given this level of dependency I'm not sure where the break could be. The one thing I have noticed from the show sip peers field is that it's showing the phones as having a netmask of 255.255.255.255 although they're actually configyred for 255.255.255.0. P -Original Message- From: Sean Cheesman [mailto:[EMAIL PROTECTED] Sent: Sunday, July 18, 2004, 11:37 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk It doesn't look like you have a context set for phone1. Try putting context=sip in the phone1 section like you have in phone2. That'll put both in the same context of your extensions.conf file and should allow interaction between the two. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, July 18, 2004 7:13 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk Hi All Total noob on the list so all help appreciated I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows). I've plugged in two Cisco 7960 phones The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly... but I cannot get the phones to dial each other :( Initially I was getting a extension not found in local message (when dialling from console...from phone just engaged (busy) tone. when I add extension from console I now get a not found 404 messageI see that there was an earlier thread on the list that discussed removing the proxy forwarding from the phone settings and I've tried that from SIPDefault.cnf but it doesn't fix the problem. I've obviously missed something but am too inexperienced to spot it. P my files are as follows:- sipxx.cnf # Lounge Phone Settings # Line 1 Settings line1_name: 11 ; Line 1 Extension\User ID line1_displayname: Lounge1 ; Line 1 Display Name line1_authname: lounge11; Line 1 Registration Authentication line1_password: lounge ; Line 1 Registration Password - sipdefault.cnf # Image Version image_version: P0S3-06-3-00 # Proxy Server proxy1_address: ipaddress of A30P ; Can be dotted IP or FQDN proxy1_port: 5060 # Proxy Registration (0-disable (default), 1-enable) proxy_register: 0 # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: 3600 # Codec for media stream (g711ulaw (default), g711alaw, g729a) preferred_codec: g711ulaw # 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
Re: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
Thanks Wayne. P -Original Message- From: Wayne [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004, 3:48 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk Hiya! Looks like you have the same problem as I had... found the answer by doing a 'debug sip-messages' by telnet'ing into one of my cisco phones... The short answer is 'its your callerid= line' you need to remove the quotes around the text part. The cisco's cant handle it. eg where you have for [phone1] in your Sip.conf callerid=Lounge1 1 what you should have is callerid=Lounge1 1 etc... Threw me for a while but the debug options on the cisco's helped out there... I think the docs read like you should have the text in quotes - but as I said - my cisco's didnt like it :) anyways - hope this helps :) Wayne! [EMAIL PROTECTED] wrote: Hi Sean Both phones are set for context=sip in the sip.conf file. As I say the phones will both call out OK (I can dial the 500 test number and successfully connect to the remote PBX through my firewall). It's just that when I'm trying to call from phone to phone I'm getting the 404 not found error in the asteris verbose dialog. If anyone has a documented example of their 7960 config sipdefault.cnf and sipxipadd.cnf files together with their sip.conf and extensions.conf files I could have to test directly on my system I'd be appreciative to test them on my system. While the WiKi's are very useful as example files it would be great (and I may do it myself!!) if there was an up to date example file with all the options for each filed and a verbose description for the rational behind it (although I recognise that this is an 'in development' product and therefore the docs have to be done at the end!!). Part of the problem is there are so many dependencies that can affect the system including how the dhpcd server serves IP address's and associated files (for example the files have to be structured in a particular order on the tftpd server for the cisco's to pick them up correctly). Given this level of dependency I'm not sure where the break could be. The one thing I have noticed from the show sip peers field is that it's showing the phones as having a netmask of 255.255.255.255 although they're actually configyred for 255.255.255.0. P -Original Message- From: Sean Cheesman [mailto:[EMAIL PROTECTED] Sent: Sunday, July 18, 2004, 11:37 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk It doesn't look like you have a context set for phone1. Try putting context=sip in the phone1 section like you have in phone2. That'll put both in the same context of your extensions.conf file and should allow interaction between the two. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, July 18, 2004 7:13 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk Hi All Total noob on the list so all help appreciated I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows). I've plugged in two Cisco 7960 phones The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly... but I cannot get the phones to dial each other :( Initially I was getting a extension not found in local message (when dialling from console...from phone just engaged (busy) tone. when I add extension from console I now get a not found 404 messageI see that there was an earlier thread on the list that discussed removing the proxy forwarding from the phone settings and I've tried that from SIPDefault.cnf but it doesn't fix the problem. I've obviously missed something but am too inexperienced to spot it. P my files are as follows:- sipxx.cnf # Lounge Phone Settings # Line 1 Settings line1_name: 11; Line 1 Extension\User ID line1_displayname: Lounge1; Line 1 Display Name line1_authname: lounge11 ; Line 1 Registration Authentication line1_password: lounge; Line 1 Registration Password - sipdefault.cnf # Image Version image_version: P0S3-06-3-00 # Proxy Server proxy1_address: ipaddress of A30P ; Can be dotted IP or FQDN proxy1_port: 5060 # Proxy Registration (0-disable (default), 1-enable) proxy_register: 0 # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: 3600 # Codec for media stream (g711ulaw (default), g711alaw, g729a) preferred_codec: g711ulaw # TOS bits in media stream [0-5] (Default - 5) tos_media: 5 # Inband DTMF Settings
