similar to: HOWTO volume per (7960) phone

All, Just came across the prompt #3 from inside the top menu of VM in latest stable. Allison does not announce the prompt, but if you know it is there, you can press 3 & successfully follow the prompts from there to send your message to other users on the system. But, of course, obviously, I am asking: how do I resolve the situation whereby the users are not hearing this prompt? (since most

Comrades, This is somewhat off-topic, but I hope also somewhat on-topic, since I'll bet a lot of others could save some time if this was able to be pulled-off. (in fact, I think it is, but I don't know where to begin) Is there a way to initialize my freshly downloaded & installed updates to the kernel & kernel source, without rebooting? I pretty much use either SuSe or CentOS

The Sjobeck Company provides Asterisk Integration, Configuration, Support, and Training. We are a crack team of Unix, Windows and Apple system integrators with 10+ years of experience working with clients both large and small. The Sjobeck Company can provide turn-key solutions, or design, build, deliver, install, configure and deploy solutions any where in the world. We also do performance

Hi All, I am running into a problem setting the ringtones via _ALERT_INFO on the Cisco 7960 phone. I am using * 1.2.1 and have tried setting the variable to several values. I have also tried setting the phone's software to both 7.5 and 8.2 thinking that it might be a version issue, but with no success. I have examined the packets and do see the ALERT_INFO header being sent, but the

Any one out there done the JiveMesenger jabber server? www.jivesoftware.com/messenger/ I want to get this running to then do the next step of tie-ing it in to the * server for presence & callerID screen pops. Pursued their site a bit but never found a HOWTO or anything that looked relevant. Appreciated. Peace. Love. Linux. Jason SJOBECK ICQ 5579183

Dear All, 1. any one know how to reset the 941 to factory defaults? 2. any one know how to config' the 941 via flat configuration file via TFTP? sample file? URL? (LinkSys tech support has been just a hair above worthless to me thus far) Thanks very much. www.sjobeck.com

I have PolyCom phones in one office working perfectly, but in another office with a new subnet, new server, new everything, the time does not work. Everything else about the phones seems fine, but the time. If you look at the internal webpage in the phone, it shows "clock". Our server, which is configured to allow others on the net to get their time from it, and it in turn gets its

HI: Is there any way to raise up sound volume on fxo on TDM04B without changing tx-gain and rx-gain ? __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

Hi, An Cisco 7960 ipphone has been set to SCCP firmware by one of our students. I want to set it to 7.5 SIP firmware and I've been unsuccessful yet. Firmware versions are SCCP 3.0 (Source: http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/mgcp/frmwrup.htm#wp1045789) ie: Application Load P003F300 Boot Load ID PC030300 When I browse, phone settings, I see

I'm installing my first Asterisk server. I have a TDM04B card installed in my asterisk server (4x FXO ports). I have 5 Cisco IP phone 7960 working fine on asterisk using SIP. My configuration to receive call is working as expected meaning anyone calling on one of the 4 FXO ports is answer by asterisk and asked to enter the extension of the person to reach and then it is transfer on the

Hello, I'm using two Cisco 7960 phones currently loaded and showing Firmware POS3-07-4-0 (Version 7.4?) and I'm having a strange problem. Whenever the phone is supposed to try to load anything over HTTP from my Apache 2.2.x web server, the connection just sits and times out. Nothing shows up in the Apache logs unless I hit cancel. What could the trouble be? -- Mark P. Hennessy

For those that have experienced low VM recording volumes when using a Digium TDM04b (or similar analog pstn card), a work around has been committed to cvs-head. Need some folks to test it; it doesn't seem to work for me, but need some feedback from others to ensure the work around is actually functioning. (The work around relates back to bug #2023, and was committed around Thursday or Friday

Hey, Hi, All, We have a few dozen of the Kirk (ie: Polycom bought this European brand) out there & most all work very well & work very well with most all versions of Asterisk. But we have been tripped-up by one combination of firmware & version & configuration variables. We are running Asterisk 1.4.23.1 (TrixBox CE). We are running latest stable firmware on the handsets. Most

(forgive the brief interruption to -users with a mostly -dev issue, just wanted to publicize this on behalf of the larger community) If there are any ambitious coders out there (not too many shekels yet, but I expect some folks may pony-up) please see: www.voip-info.org/?page=Asterisk+Bounty+Email+TTS We are at $150 & counting. Maybe lobby your exec's for $50 to contribute to this,

Can anyone point me to info on how to change the functionality of the SIP (7.4) 7960's. We previously had an SCCP firmware on the phone and the users want the phone to work like it used too. Here are some examples: The users do not want to push the new call softkey or the speaker button in order to dial a call. They want to be able to just begin dialing the number. The users do not want

Hi all, We have a small call centre here running with Asterisk 1.0.9. All the agents use Cisco 7960's with SIP 7.5 firmware. Is there any way we can show queue status on the those nice big LCD's. Especially we would like to display whether the agent is currently logged in or not. Is this possible? Thanks.... -------------- next part -------------- An HTML attachment was scrubbed...

We've been experiencing an odd issue lately. I'm not sure when it started because it's not happening on most calls--it seems confined to a couple of our queues. It's consistent though. Here's the CLI output: -- Got SIP response 400 "Bad Request" back from 192.168.249.94 -- SIP/502-9a58 is circuit-busy I've tried a few different Asterisk versions

Hello, I have installed asterisk version 1.2.12 and latest zaptel modules. but i can see some IRQ conflicts on the server. iam uisng two TDM04B cards. according to my previous knowledge on asterisk verison 1.07 asterisk has given lot of erros when starting if you have assigned the same IRQ number to any other device. My question is new releae version 1.2.12 has resolved the IRQ issue ?? TDM04B

Hello all, I've got a Cisco 7960 running version 7.4 firmware (heard there were problems with 7.5) and I can't get it to register with Asterisk. I've stripped down my configs on the phone to a bare minimum, and posted them below. Basically, the Cisco phone sends absolutely no packets to the proxy when it gets booted. If I make an outgoing call I see traffic getting to Asterisk, but

What purpose does "echocancelwhenbridged=no/yes" have on a tdm04b (or x100p) fxo interface when all phones are sip 7960's on the same wire? Rich