I'm not sure if ALL use the same TFTP address, but I believe so. My guess is that it is actually the TFTP server that the previous phone vendor used for the phone's initial configuration before shipping it to us. So in that sense it would be an old config.
Is there a way to extract the current configuration somehow to regenerate the XML? To make matters worse, I don't have the phone's admin password (it's not the default "cisco"). Thanks! -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Sunday, July 25, 2010 1:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 phone can't leave a queue It may be pulling a tftp server from dhcp, or it may just have an old config. Do all the phones (even the ones that work properly) use the same tftp address? Thanks, --Warren Selby On Jul 25, 2010, at 4:47 PM, Kevin Keane <subscript...@kkeane.com> wrote: > Stupid question (sorry, I'm pretty much an Asterisk beginner) - where > do I find the dialplan.xml? As far as I can tell, there is no TFTP > server in this network. I found the IP address that the phone tries to > use for TFTP (192.168.1.7 in this case) but there is nothing at that > device. > > Thanks! > > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Warren Selby > Sent: Sunday, July 25, 2010 10:04 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Cisco 7960 phone can't leave a queue > > Check your dialplan.xml file that the affected phones are loading. > > > > Thanks, > --Warren Selby > > On Jul 25, 2010, at 10:52 AM, Kevin Keane <subscript...@kkeane.com> > wrote: > >> I recently inherited an Asterisk system (PBX in a Flash, based on >> Asterisk 1.4 and FreePBX). The phones are mostly Cisco 7960 phones >> with the SIP firmware. >> >> >> >> The Asterisk setup relies heavily on queues with dynamic agents. The >> problem I am having is that on SOME (but not all) the Cisco phones, >> the phone will not allow dialing a second *. As a result, the agent >> can log in to queue 600 by dialing 600* but cannot log out again with >> 600**. >> >> >> >> Is this due to a setting on the phone, or within Asterisk? I suspect >> it is on the phone, since not all devices are affected. >> >> >> >> I’d appreciate help with tracking down which setting might cause thi >> s! >> >> >> >> Thanks! >> >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> 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 -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users