similar to: Cisco CallManager Image for 7940/7960

Hi All! First, let me introduce myself, as this is my first post to the list (I've been lurking for quite some time now). My name is Matt Hardeman, and I work for a software development firm in Birmingham, AL. We are interested in the Asterisk PBX and it's various configurations first as an internal solution for our occasionally bizarre telephony needs, and eventually are interested

Hi! Can anybody help me with convert Cisco 7940 CallManager Phone to a SIP Phone? I have continious error in tftp log: connect from 192.168.1.111 Mar 18 12:12:30 AKrasavin utftpd[10081]: peer requests OS79XX.TXT, conversion octet Mar 18 12:12:30 AKrasavin utftpd[10081]: unterminated option value in init packet Mar 18 15:12:30 AKrasavin xinetd[10068]: START: tftp pid=10080 from=192.168.1.111 Mar

Cisco should really be ashamed of this product... While it is physically well constructed, and has excellent sound quality along with a very pleasant user interface, the device has SERIOUS stability issues, unless you run your network with an iron fist... Quite by accident, while configuring my Asterisk system to connect to a Cisco 7960 via SIP in a standard office PBX type arrangement, I

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:

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

Does G.729 provide better voice quality than GSM? (a question for people who have tried both) --J. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 188 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20030715/52f18205/attachment.pgp

Has anyone thought about an ISO file that could be used to make a CD for a bootable install for a "basic" Linux/Asterisk system? Just re-boot and config. -- James Taylor jltaylor@metrotel.net 903-793-1953 --

Hi, I'm looking for a good codec to use on a personal VoIP setup. It is strictly for my personal use, I'll never resell it, make money or it, or whatever. It seems a free personal-use G729 codec is available as a WIN32 library. I find it puzzling that at the same time one has to pay license fees to use it under Linux, even non-commercially. I was wondering -- am I missing something?

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

Arrrrgh I hate trying to sift through all these messages and keep track of the various threads going on ......... Who else on here prefers the newsgroup/threaded approach? If you haven't already, check out news.gmane.org for mailing lists turned into newsgroups readable by news readers....... only problem being that this list requires list membership before

Hi All! Can anyone direct me to any websites / manufacturers out there who are making small, put-it-in-the-closet-and-forget-it type systems for building routers, home gateway servers, that sort of thing? My fantasy machine for this purpose would be along the lines of a mini-itx system with external power supply, dual Ethernet interfaces on board, and one PCI slot available. If it had one

Hello all, I know this isn't strictly Asterisk, but I'm sure that there are more people here using the Cisco 7960 w/ SIP, so I thought I'd post here. I've just bought a Cisco 7960 phone to use with Asterisk. It came with the CallManager image on it. I've got the 4.4 SIP images (P0S3-04-4-00). If I put "P0S3-04-4-00" in the OS79XX.TXT file, the phone downloads this

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

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

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I have asterisk 1.09 running with chan_sccp.20050701 on Fedora core 4 with kernel 2.6.12. I have a Cisco 7940 working with X-lite working on XP too, connecting to each other, Asterisk and FWD. The 7940 upgraded to SIP7.4 with xmlDefault.cnf.XML getting delivered via tftp using the loadInformation8 parameter method taken from voip-info.org ("Configuring Cisco 79xx phones with Asterisk"

Hi everyone, I'd say this question has come up and been answered before but I haven't been able to find it. I have a Cisco 7940 that I've upgraded to SIP firmware (currently P0S-3-06-3-00 - for some reason there was a failure when trying to upgrade to V7 so I left it at V6). The problem I'm having is that when I connect to voicemail the DTMF key presses dont seem to work

Does anyone have any insignt on this? Any client programs that could be used? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030925/0e9e9920/attachment.htm

Hi, ? I am currently trying to get a Cisco Callmanager 4.1 and an Asterisk server (1.6.2.21) to talk via a SIP trunk so I can use the Voicemail component of the Asterisk (all the phones are associated with the Callmanager). The connection seem to be there. When I do a "sip show peers" on the Asterisk server?I see the Callmanager as Monitored and online however I can't get any calls

Off topic, but involves an Asterisk deployment in a roundabout way. Anyone here intimately familiar with Cisco Callmanager (Version 4-5), that can tell me where a directory of the standard system voice prompts for Callmanager might be obtained? I am looking for the text and filenames of the standard prompt set that ships with Callmanager, have been all over the Cisco site and I can't find it.