similar to: Looking for better "Follow Me"

Does anyone have a good resource to learn how to program the soft and hard buttons on a Cisco 7940 or 7960 phone? Using SIP Firmware...thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060303/e5e63834/attachment.htm

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

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.

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?

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

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

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.

All, I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones. All seems well other than the phones have to be reset up to 5 times per day. It is like they lose thier ip connection or maybe thier SIP connection. Has anyone else experienced this issue? I have the phones set for static IP addresses and that doesnt seem to help either. Any help would be greatly

Wow. My message made it to the list after more than 3 hours. Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Gesch?ftsf?hrer: Stefan Wintermeyer Handelsregister: Neuwied B 14998

I've added lines like this: speeddial = 123,test speeddial = 260,Bob in the [device] section for my 7921, but the speed dials do NOT appear on the menu (click right from the main screen). Am I missing something obvious here? Thanks MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

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"

I have sent a few emails over the past couple of days that simply have not arrived on the list (or so it seems). Is anyone else encountering this ? Julian