similar to: cisco 12sp+/30vip IP phone

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

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'm thinking about getting a couple of Cisco 12SP+ phones to use on my Asterisk system. I have just bought a Cisco 7960, and they are great, but too expensive to buy a lot of them, so I though I might try the 12SP+ ones. I have seen in the archives that the phones work on Asterisk, but I can't see much in there about the images in use. When I got my 7960, it had the call manager

Hi there, I know this was discussed many times already, but after trying to get my 30VIP to work for a week, I think I might be doing something wrong. I'm trying to set it up with chan_skinny. Haven't tried chan_sccp yet, but since there are many of you already doing it with skinny, it should be working also in my case. I read and used all the hints and samples I was able to find on the

Can anyone recommend a good voip provider? Thanks

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

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 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 list, first of all, I'm new to all this so please excuse me for asking stupid questions :) I've setup an asterisk server. I connected a Cisco sp12+ phone to it, that works. Now I have a windows SIP client, SJphone, and I want to call the cisco phone. In my skinny.conf: [fabje] device=SEP000196c0066c version=F2.02 context=did line => 101 in my sip.conf: [test] type=friend

I'm using sipgate.de as my sip provider. When I'm using xlite as client on sipgate.de, everything works fine: I call number, hear ringing (real progress tone form called party, not one generated in xlite) and then talking with called person. But, when I'm using Asterisk as sip client on sipgate.de, I don't hear progress tones: I hear only one (locally generated) ring tone, and

Do somebody using CISCO AS5350 with Asterisk? Which protocol do you using: H323, MGCP, SIP? This direction: [12sp->Asterisk->h323->as5350->isdnPSTN] is ok But reverse: [isdnPSTN->as5350->h323->Asterisk->12sp] cannot hear 12sp, but 12sp hear PSTN (codec g711u) -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I have just successfully configured a Cisco 30VIP to work with my Asterisk server. I have seven of these phones new and would like to deploy them. I am wondering if anyone has this phone deployed with Asterisk and can suggest configuration of the various buttons, etc. (Bare with me as I am new to Asterisk.) Thanks, Chris

Looking for firmware (anything) for the 12sp model phones. Anyone got it drop me a line! steve_mcmahon@hotmail.com or ajener@qwest.net Sincerely, Steve McMahon Digital DataBits Innovations Salem, OR 97301 Office: (503)371-6448 Ext. 2 Cellular: (503)881-6828 -------------- next part -------------- An HTML attachment was scrubbed... URL:

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).

Hi Lin, In addition to bug 6541114... Bug ID 6541114 Synopsis GRUB/ZFS fails to load files from a default compressed (lzjb) root ... I found yet another way to get the "Error 16: Inconsistent filesystem structure" from GRUB. This time when trying to boot a Xen Dom0 from a zfs bootfs Synopsis: grub/zfs-root: cannot boot xen from a zfs root

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

The list of status bits we're seeing has increased over the years. Currently for instance, the usbhid-ups driver has defined no less than 20 status bits (and I intend to add two more in the next couple of days, FRANGE and FANFAIL): #define STATUS_ONLINE 0x00001 /* on line */ #define STATUS_DISCHRG 0x00002 /* discharging */ #define STATUS_CHRG 0x00004 /*