similar to: Cisco 7960 phone can't leave a queue

I recently inherited a Vertical Xcelerator IP system with IP2007 phones. I would like to use the phones with an Asterisk system instead, but there doesn't seem to be much information on it on Google. Is it even possible? These phones claim that they are SIP phones. Thanks! Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL:

Good Day everyone, Yesterday I upgraded the firmware on my 7960 to Sip 8.12 as provided by Cisco, however now the phone does not and will not read the RINGLIST.dat file. I've tried rebooting the phone, tried resetting the phone back to factory, have deleted the RINGLIST.dat file and reloaded the phone then reinstalled the RINGLIST.dat, and still the bloody phone will not read the file.

Hi, When we use BudgeTone where the DTMF is set to "via RTP (RFC2833)" all the DTMF functionality of Asterisk is working OK. When use Cisco 7960 the transfer is working OK, but when I try to "remote pick-up the call" through '*8#' I can't do that because the Cisco Phone start busy signal. How can I start using all DTMF features using Cisco Phone? Best

Hi folks. I attempting to set up a Cisco 7960 (SIP) so that if the user is on a call, other incoming calls will ring through to her phone and can be answered. So far I have only been able to get this working by using the call-waiting function, which is cumbersome and does not properly allow the first call to be retreived. Is there a better way to do this? Thanks in advance. John Mensel

Hi all, i have a problem with some cisco 7960. Yesterday i did a firmware-upgrade from 3.1 (1.2) with P0S30203.bin as described in the most documents. Now i get the message Phone unprovisioned and in TFTP-Log i find the following line: 07.03.2005 19:58 :Timeout error sending P0S3-07-3-00.bin to 192.168.0.83, 0 bytes So i think it now wants to load the file P0S3-07-3-00.bin. But i have not

Hi All, I am running into a problem setting the ringtones via _ALERT_INFO on the Cisco 7960 phone. I am using * 1.2.1 and have tried setting the variable to several values. I have also tried setting the phone's software to both 7.5 and 8.2 thinking that it might be a version issue, but with no success. I have examined the packets and do see the ALERT_INFO header being sent, but the

Hi all (is a bit of a Cisco question, but perhaps it's already been asked here). I'm trying to setup a TFTP server for dynamic configuration of my 7960 Cisco phones. Cisco docs suggest I download the following files from their web site: * OS79XX.TXT * SIPDefaultGeneric.cnf * SIPConfigGeneric.cnf * RINGLIST.DAT * P0S3xxyy.bin * P0S3-xx-y-zz.bin * dialplan.xml * syncinfo.xml I only

Hello, I'm using two Cisco 7960 phones currently loaded and showing Firmware POS3-07-4-0 (Version 7.4?) and I'm having a strange problem. Whenever the phone is supposed to try to load anything over HTTP from my Apache 2.2.x web server, the connection just sits and times out. Nothing shows up in the Apache logs unless I hit cancel. What could the trouble be? -- Mark P. Hennessy

Before I got down the path of converting a Cisco 7960 I have over to SIP I wanted to try and set it up using Skinny. The phone registers ok with Asterisk. When I call a SIP softphone extension on my network the call is made and I can answering it. However no voice is heard over the call. When I debug Skinny on the console after the call has connected I see the following messag: Recieved Alarm

Hello, Anyone else having problems with the Cisco 7940/7960 (5.3 firmware) and the latest CVS build, placing conference calls from the phone? I've noticed the party on the Cisco phone's side will sound very garbled, and delayed by several seconds. I haven't begun troubleshooting yet, though I'm able to reproduce this easily... Thanks in advance, -a

I know this is a little off list but I can't think of a better place to ask this question. I upgrade the phone to 7.1 and it installed the Universal Application Loader. Now I'm getting Protocol Application Invalid after it reads tftp SIP(MAC).cnf Any ideas? Again sorry this is off topic

Hi, I have followed the thread on where to get a maintenance contract so that I can upgrade the firmware to a SIP version and have called two different places, CDW and Insight. Neither place will sell the contact, all I get is a B.S. excuse to why they can't. Has anyone had luck with a more friendly place or do they have the images? TIA, Trevor

Hi there! I am having problems registering my 7970 hardphone with Asterisk 1.4(with FreePBX interface). I had an earlier post about trying to get it to work first with a 7970 emulator (Cisco IP Communicator) on the Asterisk Forum : http://forums.digium.com/viewtopic.php?t=19160 Instead I decided to try the real phone instead, and was able to advance further. The firmware was able to install

So, I can't figure out how to get the parkandannounce application to work the way I want it to... I have cisco 7960 IP phones using SIP, and this is what I have in my extensions.conf: exten => 700,1,ParkAndAnnounce(pbx-transfer:PARKED|90|SIP/${EXTEN:1}|internal,${EXTEN:1},1) exten => 700,2,Hangup and in my parking.conf: [general] parkext => 700 ; What ext.

To be clear: I’m concerned about the amount of our infrastructure (as well as downstream infrastructure, this would be actually pretty painful for both of my downstreams) that the community would have break/need fixing as a part of that. So I want this to happen ONCE. I think it is well motivated now, but switching from ‘default’ to ‘main’ when that becomes the ‘standard’ one seems way less

I mean, we could change it twice? There are about a hundred scripts out there for doing it. -eric On Fri, Jun 19, 2020 at 11:40 AM Keane, Erich <erich.keane at intel.com> wrote: > Do we have any ability to reach out to github (at least?) to see what they > are going to do? I’d very much like to avoid being the odd-project-out > here. > > > > > > > >

I am guessing you are running without reinvite's, I'm running with reinvite's with latest CVS release and 79x0 phones without any issues with conferencing... > -----Original Message----- > From: Adam Rothschild [mailto:asr@latency.net] > Sent: 08 October 2003 15:49 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Cisco 7940/7960 phone and >

This is probably better suited to a cisco forum, but thought i'd drop it in here also. Using a 7960 with *, and have a very specific need. If the end user is currently on a call on the 7960, and a new call comes in, i need the phone to: show a visual indicator of the call (pref flash the call light) emit a ringing tone *NOT* play a call waiting beep inline on the current call. currently

Hey Guys, I apologize for my ignorance on this one. I've got several 7960s running on Asterisk1.4 with 15 or separate queues and am trying to figure out a way to identify to the 7960s, what queue the incoming call is on? Is this possible at all? Thanks! -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281

I have the cisco 7960 phone. If I call out on the phone I can hear the person called and they can hear me. If I have that same person and same phone call into my asterisk box and select my extension I answer as it is ringing then I cannot hear them speak. However, they can hear me. I dont understand? Anyone know what I am missing. I have 4 other sip phones on this machine (grandstream) and they