similar to: cisco 7960 image

Hi, I was woredering if you could help me to put into practice this solution. The idea: Create a IVR-Voicemail The scene: PSTN------/6------PBX--------/12--------- Internos | /4 ports | IVR-Voicemail The Operation: 1)Where a call enters from the PSTN, the PBX

G'Day All, Can someone help me out please. My new CISCO 7960's manual says I have to setup a TFTP server. Googled it and got a little understanding, but from * standpoint, well I am still a lost. Can I set this tftp server on the same * box? Can in be on a WinXP box? Which tftp software would you recommend? Thanks much. BTY: Does anyone have a How-To on getting the 7960 fully configured

Thanks But how about the config files (SIP...) that needs to be inside the tftp server, where can I get a sample of that? That's where the images for the firmwares of the ip phones come from, on boot right? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Alen Salamun Sent: Wednesday, January 26, 2005 5:47

Hello, list a have a problem i can start asterisk, i get the fowlling error: [chan_capi.so] => (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Feb 11 13:50:36 NOTICE[2535]: chan_capi.c:2636 load_module: CAPI not installed! Feb 11 13:50:36 WARNING[2535]: loader.c:345 ast_load_resource: chan_capi.so: load_module failed, returning -1 Feb 11 13:50:36

Hi all, I have just been reading an article on the asterisk-doc site about ISDN X-Over cables. The article mentioned the converting of an NT1 to make this possible, has anybody got the information required to modify a BT NT1? Or any information on the BT NT1. Thanks in advance. Regards Dave

You could try to use the native mp3 support for MOH if you really want mp3 support. It is a lot better than using mpg123 IMHO. mpg123 kept doing nasty things to my system :) See http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20musicon hold.conf there is a section about the native support. Guillaume > -----Original Message----- > From: Stefan Gofferje

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:

> -----Original Message----- > From: David Liu [mailto:david@deltapath.com] > Sent: 31 January 2005 14:34 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Announcement to caller when > called party haspicked up - without initial Answer()? > > > This is super easy to do. All you need to do is to put that > announcement

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, has anyone managed to establish an XMPP connection to the facebook Jabber servers? I'd like to send messages on missed calls vie FB. - -S - -- (o_ Stefan Gofferje | SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler & Koch - the original point and click interface -----BEGIN PGP

Hi, I got a problem of having to upgrade 35 Cisco 7960 phones from default firmware of 3.1 to 7.5. The problem I get is that when trying to upgrade I see on the tftplog that it can't seem to find the file (8 character issue). So I renamed the files to suit what is supposed to be in them. I am trying incremental upgrades from 3.1 -> 5.3 -> 7.5, with no luck. It goes to

I have an Asterisk PBX on the private lan, which is protected from the public Internet with a Linux iptables machine. The firwall is it's own seperate box running NAT with SPI. I want to drop a SIP phone at my brothers house, and have it be an extension off my Asterisk box. I've been looking around at some FAQ info on forwarding ports, and also looked at siproxd. Anyway, I'm

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! Are hints not yet implemented in res_jabber? I have this here: exten => 3000,hint,gtalk/gtalk_account/mari.xxxxxxx at gmail.com But the hint doesn't show any difference. It always shows online on the phone and core show hints always shows that: 6003 at internal : SCCP/6003 State:Unavailable Watchers 0 6002 at internal :

Hi, in case, anyone is interested... I have started compiling a blacklist of hosts and networks from which SIP fraud attempts occur. My criteria currently are: To block an IP: - Minimum 3 attacks within one week from the same IP To block a network: - Attacks from minimum 3 IPs from that network within 2 weeks Common criteria: - Provider does not react to complaints OR - Provider sends autoreply

Hi I'm trying to deploy asterisk, but I can't seem to find documentation for the TFTP server to run the cisco 7960 ip phones' I was told before that you need it and it could run on linux. Thank You ,jm

Hi, I would like to implement an auto-redial function in a context. The idea is about like this: Dial a number Hear busy Hangup Pick up again Dial a code like *123 => jumps into a context which redials until callresult is not busy Maybe like this: [autoredial] exten => s,1,Set(number=${CHANNEL(lastdialed)}) exten => s,2,Dial(SIP/${number}@account,60,g) exten => s,3,Wait(15) exten

Hi folks, when I start asterisk directly, I get a colored CLI. When connect to a already running asterisk with asterisk -R, it's never colored, despite I'm running both from the same console (tty). Is there a way to force asterisk -R into color mode? Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I have a general question about SIP access for nonregistered users. I would like to make it possible for basically anybody to make a SIP call to my asterisk without having to have a user account, but in a specific context. So that e.g. somebody could make a SIP call to SIP/stefan at my.asterix.pbx and it would go like this: [incoming_guest]

Before you all reply that its available via Cisco, I'm not qualified to be a tech member according to Cisco. I just bought 4 7960's with which to use with * and I want to load up the SIP image into them. Does anyone have it that they can make available to me please? Thanks -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/

Hi! The university where I work just bought four Cisco 7960G IP phones (they didn't ask, just came across the door and gave me a box and told me: "Can you make this work with the Asterisk PBX we have?"). According to what I read, there is no much hope, because I have not the SIP firmware (too bad). Has anybody succesfully got an answer from cisco?, or does anybody happend to

Chuck, The first thing I would do is to upgrade the load to version 6 or higher. I'm running the latest...version 7.2. (I'm very happy with it) Are you using TFTP to load the configuration or manually configuring the 7960? I know it's a pain to setup TFTP just for a quick test. However, it's well worth it. If you have a CCO account you can find the latest load and config files