similar to: 7960 show queue status

Hi, An Cisco 7960 ipphone has been set to SCCP firmware by one of our students. I want to set it to 7.5 SIP firmware and I've been unsuccessful yet. Firmware versions are SCCP 3.0 (Source: http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/mgcp/frmwrup.htm#wp1045789) ie: Application Load P003F300 Boot Load ID PC030300 When I browse, phone settings, I see

Morning All, Quick question that has me stumped. Have a queue with several members (Statically defined in queues.conf at this stage for testing) who use Cisco 7960's. The queue is configured to use rrmemory and generally this works correctly. However if a member is already on a call their phone will still ring (The 7960 can show multiple incoming calls for one line). I really don't want

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

Hey all, I'm running * (1.0.9) with some 7960's. When I pick up the handset it often will flap on and off hook eventually hanging up on me and not realizing that I am still holding the handset. These phones are running with the SIP 7.5.0 image on a LAN (no NAT). I am not sure if this is a phone issue or an Asterisk setting. Ideas?

Does anyone know what can cause queue members to go into a status of "Unknown"? pbxtel-01*CLI> queue show cs has 2 calls (max unlimited) in 'rrmemory' strategy (24s holdtime), W:0, C:447, A:20, SL:91.7% within 60s Members: SIP/1405 (dynamic) (Unknown) has taken no calls yet SIP/1420 (dynamic) (paused) (Not in use) has taken no calls yet SIP/1442

Hi, Despite much searching, I can't find anything quite like the problem we seem to be experiencing with our recently activated asterisk pbx. Of four people answering several calls per five minutes, I have reports of the occasional call starting with dead air. The user picks up the call from the queue, and they can hear that the line is too quiet. If they start talking, the caller either

Hi I've taken apart a 7960 to fit a backlight to the LCD. Would others on the list be interested in this as a project when I've finished (i.e. should I document and photo all the stages)? Also I can source some 7960's at a very good price but what are the rules about posting this information on the list if others were interested? P

Hi, We recently updated from an early Asterisk 1.2 SVN to 1.2.15 (on Debian Sarge) and the behaviour of our Call Centre queues has changed slightly. Before the upgrade, when a caller was waiting in the queue, the estimated hold time was announced as expected ("estimated hold time is less than 2 minutes ..."). Now the caller gets an announcement of their sequence in the queue

Hi, Has anybody been successful in running the 7914 expansion unit for the Cisco 7960G IP phone? For anybody unaware of what the expansion unit does, it provides 14 additional buttons, with an LCD display. The idea, is that with an expansion unit (a 7960 can take upto 2 of these units), a user can either assign more speed-dial's, or can monitor line status/account status. So, you can

Check my reply to your last post. Use SetGroup and Checkgroup before sending the call to your agents. Robert Jackson -----Original Message----- From: Henry Devito [mailto:hdevito@qwest.net] Sent: Thursday, September 30, 2004 10:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Queue Setup almost got it Ok I think I have the queue

I'm trying to setup a couple of Cisco 7960's in asterisk. I have asterisk working fine for sip clients, and can call the 7960's just fine, but I can't seem to dial out on them. As soon as I enter the first digit, the phone attempts to dial it without waiting for the rest. I've changed timeout settings, etc but can't seem to get it to work. Any ideas? Asterisk

Hey, I'm testing out some call queues. I have 7940's and 7960's with the SIP 7.4image. I have a queue that looks something like: [testqueue] strategy = rrmemory timeout = 15 retry = 5 weight = 0 announce-frequency = 0 joinemtpy = yes reportholdtime = yes I dynamically add a phone or two to the queue (AddQueueMember, not agents). When a caller calls in, connections are made and

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Before you all reply that its available via Cisco, I'm not qualified to be a tech member according to Cisco. I just bought 4 7960's with which to use with * and I want to load up the SIP image into them. Does anyone have it that they can make available to me please? Thanks -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/

Hi, I was wondering if it is possible to setup with Asterisk a Cisco 7960 to use extension mobility / roaming. Meaning that a user logs into a phone and his profile moves with him / her. I have a network of ~75 Cisco 7960 phones, running SIP 7.5 distributed across 2 asterisk servers in 2 cities. I would like to enable the users to have the capability to move from one phone to another one

hi! I'm looking for a robust hardware IP phone which supports SIP protocol inorder to implement a call centre. Have anyone used CISCO SIP phones (eg:- 7960G ) with asterisk. From what I know these CISCO IP phones are very robust and feature rich. Yet I'm nervous whether * don't like CISCO at all. Thoughts are most welcome. denzel. -------------- next part -------------- An HTML

Hi Everyone, I am doing a queue show and I can't see any column that shows a queue member priority. Is there any other command that can show the member priority on the Asterisk 1.4x CLI? We are using this format of dialplan to login agents: exten => 123,Answer() exten => 123,n,AddQueueMember(500|Local/${CALLERID(num)}@from-internal/n) exten => 123,Hangup() ^^^^ Where 500 is the

I've got 4 Cisco 7960's and they're behind a firewall (sonic wall). The asterisk box is on a WAN connection on the other end of a DS3, the phones connect fine to the Asterisk server as you can see from the output of show sip peers below. tp3/tp3 <firewall-ip> D N 255.255.255.255 60665 Unmonitored tp2/tp2 <firewall-ip> D N

Hello All, Figured I'd ask this. I have an asterisk running with some X100P clones. The system runs fine. My question is, the lines are tied into a panasonic dbs analog extension. When you call through, all goes well except that the call does not know it is connected. This causes problems if lets say, I want to do a 3 way conference, since my cisco 7960 does not see the call as connected,

I recently upgraded a Cisco 7960 to the SIP firmware, it worked fine without a "call-manager". I just put the SIP firmware and associated config files in the TFTP directory of my asterisk server so that the phone could pull the firmware off of my asterisk server via TFTP. It took me about 5 minutes maximum to get the phone working with SIP through asterisk. NO Call-manager was used in