similar to: Cisco 12SP+

R-newbie question I have 2 matrices (a) P1 has only one column of 32K rows (b) PC has 2 column {P, C} of 3200 rows Every values in P1 matches with a value in PC[,p] (column p). I would like to use Which to search for all value in P1 that matchex PC[,p] and get the PC[,c]. However because P1 and PC does not have the same number of rows, I got lots of 'NA'. Thanks for your

Hi folks, I have got a Cisco 12SP+ working (thanks Derek!) but I'm having a minor issue with it. When I use it to call another desk I get no CallerID. The receiving phone diplsays "asterisk" as the CID. Below is the skinny.conf stanza. [2207] device=SEP00308062B006 version=P002L2J2 context=intern callerid="Luke's Room" <2207> mailbox=2207 transfer=1

I have been looking for the firmware for the 12sp+ and 30VIP and have been unable to find it. Any help in locating would be much appreciated. Thanks ======================================================= This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error

I've searched high and lo and googled to I can't google no more... I knew that Cisco bought Selsius to get their VoIP solution but what I didn't know was that the 12sp+ is based upon an ITE-12 product that is apparently used at universities. I've taken two of the phones apart and started swapping parts and have looked up the IC's on the Internet... Anyway... After using

Hi I have problem with Asterisk an 12sp+ phone. Asterisk's skinny implementation doesn't correctly processes 'onhook' event from phone, so voice channel stays opened and no new calls can be received by phone. What i'm doing wrong? :) -- Denis Chapligin

does anyone have a running cisco 12sp+ or 30 vip phone on their network? and if so could you also tell me what tftp files you actually use and if there are any special settings in skinny.conf that i need? (I ran several searches for setup, nothing has come up so far so i'll ask for advice now.) Thanks Rohde -------------- next part -------------- An HTML attachment was scrubbed... URL:

I was able to get Asterisk working with the demo on FreeBSD 5.3 without crashing, but not the music on hold, so I just have that disabled for now, but I'm ready to get some IP hardware working. So I picked up a Cisco 12sp+ IP phone (mistake?) and am having difficulty finding any truly helpful instructions / troubleshooting to get this configured to work with asterisk. If I could just get

I would appreciate any help in locating the latest firmware for the cisco 12sp+ phone. I currently have D2.04. I've searched the digium lists, google and yahoo with out any luck so far. Thank you for your time!

Hi, After reading this valuable forum and the voip-info wiki and follow all the steps , but my Cisco 12SP+ remains unregistered. These are my config files: skinny.conf [general] port = 2000 ; Port to bind to, default tcp/2000 bindaddr = 172.20.1.1 ; Address to bind to dateFormat = D-M-Y ; M,D,Y in any order (5 chars max) keepAlive = 120 languaje=es allow = all ; disallow

If you have been paying attention, you already know this, but this weekend I have spent time ironing out the various details with my chan_skinny code that has been out there, if you knew where to look. I believe I now have all basic features operational and am going to be working on getting the class 5 (hold, transfers, call waiting and caller*id, etc) operational in the comming week(s).

I have gotten some cisco VIP 12 and VIP 30 IP phones that I would like to use with asterisk, I have set them up using chan_skinny. The phones work well, except the only problem is that it is like the cisco phones are muted. When I talk on the cisco phones I can hear my self through the ear peice, but the person who I am calling can not hear me at all. I have tried various cisco phones from various

Hi there! Has anyone succesfully used a Cisco 12 SP+ with *? If so, how did you do? I'v not even tried, but before trying I thought I could bug you somewhat. =) //Filip

Can anyone recommend a good voip provider? Thanks

Regarding the brilliant wiki site: http://wiki.centos.org/HowTos/Amavisd?highlight=%28Amavis%29 I faced the following issue on CentOS 6.2: "Spamassind" saves each message and its attached part in a folder in clamd accesses the folder, creates itself a temporary folder and deletes it afterwards. This was stopped by SELinux and caused the virus scan to fail. This action causes SE-Linux

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm just learning about VoIP and Asterisk. I've got a developers kit on its way and I've managed to get hold of a couple of cheap Cisco VIP 30 phones. I've trawled the web and found a few snippets of information on these phones but I still can't get them to work. Does anyone have any config files or any idea on how (if I can)

Dear sccp users :-) We are announcing the support of the Cisco 7935 Conference Station within the chap_sccp channel driver (experimental version) AND the "alpha-beta" stage of the 12SP+ Support. I will later on today add the information on the Wiki on how to get this phones correctly registered. However the support is still not fully functional (some issues), but please test this

My configuration is: [ISDNPRI] -- [CISCO AccessServer AS5350] --<H.323>-- [ASTERISK] -- [CISCO ip phone 12SP+/Skinny] When call is initiated from IP phone -> Asterisk -> AS5350 -> ISDN everything working ok (RTP is ok). But, when call coming from ISDN -> AS5350 -> Asterisk -> IP phone IP phone party can hear ISDN party, but ISDN (incoming) party canNOT hear IP phone party

Try sending 5350 config and oh323.conf, versions, etc... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Goran Dj. Sent: Wednesday, December 22, 2004 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk->AS5350 misplaced RTP to 127.0.0.1(AS5350 party

Hi Steve, I am having this problem in which RxFax is still holding the file after receiving a complete fax. Somehow the zap channel is still active but on the fax client it was sent successfully. If you call the line it is still busy. Changed from phase 3 to 4 >>> MCF: 8c HDLC underflow in state 8 Changed from phase 4 to 3 Slow carrier up <<< DCN: fb DCN with final frame tag

I'm attempting to do a first-time Asterisk install at home, firstly for use by my self and my family, and secondly as a learning experience. I've got a new house, and the previous owners removed all but one (1) phone jack. So I figured I might as well build a PBX. Functional goals include station-to-station calling, rudimentary auto attendant/voice mail, and perhaps tieing into the