[asterisk-users] On-the-phone
Hi there, I have a Polycom phone that has two extensions registered to it, let's say 200 201. Is there anyway to code in the Asterisk dialplan to show BOTH lines are busy when either of 200 or 201 are in use? Reason is that my Polycom phones will show the presence info via BLF red light, but I would have to have 2 separate entries in the monitoring phone (one for each extension), even though it's only one device. Thanks!___ --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 7961 new firmware stops reading configuration files
Chad, I had the same problem when upgrading to some of the newer firmware. The newer firmware gets even pickier (if that's even possible) about the config files. Go the phone's webpage and look at the debug log. It will show you where it's not parsing correctly. I'm not in front of my phone now so I can't look, but I remember it getting upset about networkLocale or userLocale or something of that nature, so I just removed that section of the XML code and it loaded fine. Good luck, Preston Edwards - Original Message From: Chad Osmond [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, December 20, 2007 9:32:43 AM Subject: [asterisk-users] Cisco 7961 new firmware stops reading configuration files Hello, I have been running SIP41.8-2-2SR4S firmware on the Cisco 7961 and have recently upgraded to SIP41.8-3-2SR1S to add the DND button on the front of the phone and also to hopefully resolve some issues with the phones not registering after a long period. Once we upgraded the phones now display Error Verifying Config Info in the Status messages and will not process the configuration file. To make a change on the phone I have to downgrade to 8.2.2R4 and change the configuration, and then upgrade to 8.3.2R1, which is a bit of a pain. The tftp logs indicate that the phones is getting the correct SEPMAC.xml.cnf file but will not parse it, so that doesn't seem to be the issue. The Wiki pages for 79x1 indicate that it's a known issue, has anyone managed to get past the issue? I tried logging a call with Cisco TAC, but they're giving the We don't support SIP on anything other then CME... Thanks, Chad __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ --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 7970 BLF/Presence
I have been trying to get the 7970 (running SIP firmware) to display presence information about other extensions. Thus far, I have been unsuccessful. Does anyone have BLF working on the SIP-loaded 7941/7961/7970/7971? I have been using the following as a guide for my work: http://www.voip-info.org/wiki/view/Asterisk+Presence+for+Cisco+79x1+Phones I have successfully implemented every step (although there are a few minor errors in the code on that page that I have corrected) except for the part where the 7970 has to connect to Asterisk via SIP/TCP. I have chan_sip.c patched with the SIP/TCP patch to allow for connectivity, but I cannot seem to force the 7970 to connect that way. I have read several posts that indicated that the SIP firmware does not support presence information as of yet, but supposedly, whoever wrote that article above has it working somehow. Does anyone know how to make the phone connect over TCP? Or, better yet, does anyone have a working method that they would be willing to share? These are great phones but in the environment that we're in they are almost useless if we don't know who's on a call when. I'd rather not go the SCCP route unless I absolutely have to. Thanks!___ --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
