search for: 30vip

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

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....

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'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 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 fi...

...sk as sip client on sipgate.de, I don't hear progress tones: I hear only one (locally generated) ring tone, and then quiet till somebody (called party), answer. After that I can normally talk with called person. That's happening both with Xlite as sip client on Asterisk, and with Cisco 12sp/30vip as skinny/sccp client on Asterisk (so, my conclusion is that problem is in Asterisk). The biggest problem there is that some numbers have free voice messages (for example: "number is changed, call xxx-xxxx") and I do NOT hear that messages. *CLI> -- Executing Dial("SIP/xlite1-f8...

Hey all, I have set up my cisco 30vip using chan_skinny because chan_sccp wont register. The problem I am having is, everytime a call is sent to the phone Skinny/200@jason it rings once, then asterisk segfaults. heres the output -- Executing Answer("SIP/4437821638-7588", "") in new stack -- Executing Dia...

Hi, Does Asterisk have a limit to how many NAT'ed SIP clients it supports behind a single IP? I have the weirdest problem ever. I have three SIP endpoints. SNOM phones, if it matters. Their extensions are 200, 201 and 202. Apart from the username/password, the sip entries in sip.conf all have identical configuration. They're all NAT'ed behind the same IP. 200 and 202 registers

...] 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 = [101] device=SEP00B06409E748 model=12SP ; Valid models: 12SP, 30VIP, 7910, 7920 (so far) version=P002L2J2 host=172.20.1.1 context=telsip line => 101 callwaiting=no callerid="Juanjo",<101>; extensions.conf exten => 3,1,DBget(temp=clid/SIP/Phone3) exten => 3,2,SetCIDName(${temp}) exten => 3,3,DBdel(clid/SIP/Phone3) exten => 101,1,Dia...

Hi all, we are happy to announce the new test-release of chan_sccp. The Cisco 7920 support is working now, however, some call handling stuff is hardcoded for testing. Also some more in-deep-knowledge of Skinny was archieved, how they handle calls. Please test it with your 7940/7960 and 30VIP and report any problems using our bug tracking tool. The latest tarball is available at: http://www.lambda-solutions.de/7920/ The bugtracker is here: http://www.lambda-solutions.de/bugs/ Many thanks to Theo for the initial good work ! Keep going! Also thanks to my friend Mathias & Enno for so...

.... Re: L options in Dial() dont seem to work.... (Steve Murphy) 12. Is it possible to install CCM on a Linux platform ? (Olivier) 13. Re: L options in Dial() dont seem to work.... (Mark Reardon) 14. Scratchy Audio with Asterisk 1.2.4 over IAX on FreeBSD? (Benoit Panizzon) 15. Re: Cisco 30VIP Phone (Jason Parker) 16. SIP & NAT (Mike Hammett) 17. Re: maximum simultaneous calls (Matthew J. Roth) 18. RE: SIP & NAT (Alexander Lopez) 19. Re: Multi-line phones - Asterisk uses wrong callerid (Drew Gibson) ----------------------------------------------------------------------...