similar to: Cisco 7940/7960

hi! i had some good news regarding the cisco 7920 and the internetworking with asterisk (and possibly SIP ?). Status: chan_sccp.so not coredumping anymore :-) Phone contantly in reboot loop [see below] :-( Reboot Loop means: ------------------ Phone auth's with AP Phone gets IP from DHCP & TFTP Server Phone loads OS7920.TXT Phone loads SEP<macaddr>.CNF.XML Phone loads

Does anyone have the .bin file(s) to convert a Cisco 7940/7960 back to the CallManager image? I want to start playing around with the chan_skinny addition, but it seems the .exe's from cisco want to open a connection to a SQL server or CallManager (which I don't have).

Hello, I'm considering using Asterisk with some type of Cisco phone, and currently considering either the 7940 or 7960 because of its more-complete functionality (compared to the 7905). I'm currently wondering: Do all the expected functions (transfer, conference, voice mail, message waiting indicator, etc.) work normally with Asterisk over SIP? What caveats are known about using

Hi All, I'm thinking of creating a new asterisk server using the latest 1.4 stable release to replace my ageing Asterisk SVN-branch-1.2-r7231 (its been a while!). My only concern - my phones are Cisco 7960's (with sccp firmware 7.2 loaded) and to support them better, I remember compiling in a skinny(?) driver to replace the (from what I could tell) basic in built sccp support. After

Interesting notes on the 79xx series. The 7920 is the wireless phone; not mentioned here. For a more complete guide to Cisco's phones, see: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheets_list.html The 7902 is the "very inexpensive" Cisco phone, and it looks like it will be SCCP (Skinny) only. Twiddling my thumbs here waiting for the chan_sccp to

Hi All Will cisco 7910 ip phone compatible with Asterisk? I know that 7960 are fine. David Kwok -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 1878 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040112/e8023f35/smime.bin

Hi, I am new to Asterisk and have some general questions _before_ I start buying equipent to install and get everything up-and-running. (this means I have no running Asterisk (yet)). I have read already a lot of doc, but some things are not clear to me, since I'am inexperienced in Asterisk and PBX. The goal: Make PBX using Asterisk and Cisco 79xx equipment for 25 phones (1 x 7970, 4 x 7960,

Hello all, I've got myself into a bizzare situation that I can't seem to get myself out of... Was wondering if anyone had some advice that might get me 'over the hill' on this... Some background: PBX consists of an Asterisk box (running TrixBox), 4 Cisco 7960's, 2 Polycom IP500's, and now an additional Cisco 7960. The phones are all on a separate LAN. There is

Maybe not the place for this but thought I'd post the info for others. I purchased a cisco 7960 off ebay and needed to convert to SIP for *. I know * supports SCCP but I wont go into that here. I'd read on voip-info.org that a contract could be purchased for approx $8 to allow me to download the firmware. I though, being in the UK, i'd get one through a reseller in the UK.

Before I got down the path of converting a Cisco 7960 I have over to SIP I wanted to try and set it up using Skinny. The phone registers ok with Asterisk. When I call a SIP softphone extension on my network the call is made and I can answering it. However no voice is heard over the call. When I debug Skinny on the console after the call has connected I see the following messag: Recieved Alarm

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031201/c9e420c5/attachment.htm -------------- next part -------------- Hello all, I have 1 IP 7940 with the following Firmware versions App Load ID: P00303011201 Boot Load ID: PCO303010001 Version 3.1(12.1) Could you please confirm, if my IP phone has the correct SIP image. My asterisk

Here is what I am attempting to do (which works well on Cisco Call Manger). I have some 7960's that have multiple lines on them. The second line specifically is a "helpdesk" line that is shared among multiple phones. Here is how I am making that line ring on multiple phones, maybe you have other suggestions: exten => 135,1,Dial(SIP/135@100&SIP/135@101,20,rt) So this

Hi again, I found chan_skinny and that seems to work pretty good. the SCCP one filled out all the buttons really nice, but skinny seems to be working. How do I fill out the second line button on the phone with skinny.conf? Thanks much! ...Jeff

Was in the process of upgrading a 7960 to SIP and in advertently applied a skinny image (P003G302.bin), now no matter what i put in OS79XX.TXT and OS7960.TXT it simply wont upgrade. The phone is pulling down OS79XX.TXT from my TFTP server but then goes on to repetedly ask for SEP<MACaddr>.cnf. Help!! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all; I have been searching for an answer to a question that a customer asked me and I have only found a few older answers. So, wanting to find out if anyone has any experience with this issue and can help provide me with some advice. I have a customer which is strongly interested in using Asterisk as a PBX. One of the core requirements, however, is that the system MUST be able to

Hello all, I recently purchased a Cisco 7940 IP phone to do some testing with (to validate a migration to asterisk for our internal PBX needs). I understand that I need to update the phone for it to support SIP, so I configured the phone with an IP address and pointed it at my tftp server. When I reboot the phone I am currently getting "TFTP File Not Found SEPDefault.cnf" in the

I'm using happily this cheap phones, but I still have a little problem. Configuring the phone is extremely easy on * and I've a couple of them perfectly working, except when i try to call some toll-free number (in italy 800xxxxxxx ). If the number called is an IVR system, often with GrandStream (but also with Cisco 7905.h323) it's impossible to make the menu choices via the Dialpad.

I have Asterisk up and running and it is working great with my SIP phones. However, I have some "Skinny"-protocol Cisco 7960s. Does Asterisk support the Skinny protocol? I've seen some references to Skinny in the software. If so, should I stick with Skinny with the 7960 or convert to SIP? If anyone has some Skinny confs they would send me I'd be much obliged. If I should

ive had enough of cisco unity and microsoft exchange and im looking for alternatives to our voip system. right now, we have 3 cisco callmanagers, 1 cisco ip icd system, and 1 cisco unity voicemail system. all phones are cisco 7940/7960's and some ata186/188's. voice gateways are cisco vg200's with pri cards (5 total). im running h323 on the gateways and phones are of course

Hello: I would like to know if anybody has test various usb phones with an usb hub conected to an asterisk server. The idea is to build a public telephony site with one or 2 analog cards 4 or 8 FXO, least cost routing and billing and usb handsets with dialpad for users, so I really don't need a graphical interface for the soft phone ( the users just dial from dialpad). Any has done this