similar to: Program Buttons on Cisco 79xx Phones

Heya list, I need some advice/experience. Some of our customers are asking us about DECT solutions for their asterisk install. Some others will not go to asterisk if there won't be a DECT solution. They now have a Siemens or a Samsung PBX. Those PBX-es come with a DECT basestation and optionally repeaters etc. All those basestations speak some own protocol to the PBX, so we cannot use them

Hi everybody :) I am a new member here and hope that someone gives me a hint for my problem: Let's say I am at work and my SIP phone (KPhone in my case) is connected to my private Asterisk. I want to call my wife at home so her SIP phone rings. She does not pick up the phone (maybe she is somewhere in the house and has to run to the phone) so after 15 seconds her cell phone should ring.

Does anyone know if the 7912G (which the wiki says can do either sccp or sip) uses the 7940/60 sip firmware? I ask this because the only firmware I can seem to find on TAC for the 7912G is sccp, no sip...if it takes it's own firmware and doesn't use 7940/60 firmware, can someone point me to the right location for it? Thanks, Marty Mastera M3 Resources marty@m3resources.com Phone:

I?ve read various stories about the Asterisk SCCP implementation. Some say it works fine, others say it doesn?t ? stick with a SIP phone! My dilemma is that I?d like a WI-FI phone with multi-line capability, and so far Ciscos 7920 is the only one I?ve found. The disadvantage is that it cannot be upgraded to support the SIP-protocol, so if I purchase such a phone I am stuck with the SCCP.

I'm setting up a 7921 and now want to add a second line to the phone. In my SCCP.conf file I have: autologin = 235,299 However, on reloading SCCP the phone fails to login to the second line with this error: [Sep 12 12:46:49] WARNING[12224]: sccp_actions.c:185 sccp_handle_register: SEP001BD457F8B1: Failed to autolog into 299: Max available lines phone limit reached 299 Is there a

Hello @List, i'm using a Cisco 7970 / 7914 phone with Asterisk 1.4 & sccp part of my sccp.conf type = 7914 (Cisco 7970 with 7914 phone extension) autologin = 117 description = Test speeddial = 10,Test (10),10@mycompany my speeddial line is for internal call howto call a external number eg. +xx467584933 ?? i want to press a button and call directly out thank you greetings, Daniel

Ok, so multiple people have said that hinting is possible with chan_sccp on the 7940/7960's and such, has anyone got this working? How do you go about getting this to work? I'd use the wiki, but it's link to the mailing list topic on that doesn't work anymore :( -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

What is involved in getting SIP firmware into a Cisco 7960 with sccp installed? Expensive image from Cisco? Plated in unobtanium? joe a.

I don't think that will fix my problem. The hints on the individual user extensions (101, 102, 103 and 104 below) are working just fine. sccp.conf example of 1 user: [devices] type = 7970 description = User1 tzoffset = -6 autologin = 101,401 speeddial = 102,User2,102@wct-internal speeddial = 103,User3,103@wct-internal speeddial = 104,User4,104@wct-internal device => SEP000F90CEF9D3

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

Any one knows where to install chan_sccp for asterisk 1.4 ???????. Please guide me from where can I download the asterisk 1.4 sccp channel driver and how to install it because I tried to get chan_sccp-mayday05.tar.gz When I trying to install it ,error happened like this. Please help me how to solve this issue. [root@XPL t]# cd chan_sccp [root@XPL chan_sccp]# make clean rm -rf

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?

Say there! This is a new Asterisk install on Fedora; I downloaded and installed Asterisk 1.2.1 and it runs fine. I downloaded the 20051217 version of chan_sccp, compiled and it installs ok. When I start Asterisk, it dies on startup at the activation/recognition of the chan_sccp module... A snippit from the startup: Asterisk Dynamic Loader Starting: [res_musiconhold.so] => (Music On Hold

hi All, Sorry to repost but................ I have a server setup that runs sip no problem. I want to try a cisco phone. how do I a) Tell if I have skinny support loaded b) Load it onto a debian system Many thanks

I am running Asterisk 1.4.5 on a debian Linux server. Saturday night/Sunday Morning Daylight Savings time occurred. The server shows Mon Mar 10 10:59:42 PDT 2008 when I do a date command, but the Cisco 7940 and 7960 show 09:59 10/03/08. How do I update the time display on the telephones please? No virus found in this outgoing message. Checked by AVG. Version: 7.5.518 / Virus Database:

Hi, as noone out there seems to be able to maintain chan_sccp, i'm trying to switch to chan_skinny. With the newest 1.4 svn the Softkeys are mostly wrong/non functional. I see Redial NewCall CFwdAll more (more) CFwdBu... GPickUp Confrn more NewCall works, CFwdAll seems to toggle DnD, the rest of the buttons do notting. Any ideas how to fix this? Regards, Andreas

Hi all, Does someone know if it's possible to use hint function with skinny ? Can anyone send me an example ? Thanks in advance, Alex.

> I encountered a semi-lockup situation with chan_sccp and a 7960 when > switching between calls. It's kinda reproduceable - When holding, > resuming, holding, resuming between two calls on a line, one of the > calls become kinda zombie. The phone enters a state where the call is > signalled active but no softkey is there. Already fixed in the CVS Sergio

Hello everybody, I've got Asterisk 1.2.4 running with two Junghanns QuadBRI cards using the qozap driver from bristuff 0.3.0-PRE-1l. One of the cards is running in TE mode, the other one in NT mode. Whenever I call into an empty MeetMe conference room on one of the NT ports or via SIP and hang up the call during the "you are currently the only participant in this conference"

Has anybody successfully (or I guess unsuccessfully for that matter) implemented Cisco Call Manager and used an * box for voicemail? I checked the wiki and google and I see some references to Call Manager Express and *, but CME is completely different than CM. If anybody has done this or has any insight, it would be appeciated. We are trying to migrate ~ 300 users off of Cisco Unity and