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Can I get the voicemail application turn on / off the MWI (message waiting indicator) on the Cisco 7960?

These taken tonight (9/22/2004) at the Expo and Reception Enjoy. http://photos.tropiano.org/gallery/astricon-2004 Lenny

...| debbiet at arlut.utexas.edu | > | Environmental Sciences Laboratory | +1 512 835 3367 w | > | Applied Research Laboratories of UT Austin | +1 512 835 3544 fax | > | P.O. Box 8029, Austin, TX 78713-8029 | home email: debbie at icus.com | > _______________________________________________ > CentOS mailing list > CentOS at centos.org > http://lists.centos.org/mailman/listinfo/centos ----- End forwarded message ----- -- | Debbie Tropiano | debbiet at arlut.utexas.edu | | Environmental...

I have a DG-104S (which I reset to factory settings, it's DHCP'ing off my network, plugged into the WAN port). The system comes up, and I through the web browser set under Call Agent IP Address to: Notify Entry: dlinkgw@[192.168.1.1]:2427 (192.168.1.1 is the * server) I have RGW Name: and DNS IP addres the DNS IP of the MGCP box and DNS State disabled (not sure what to set it to) --

The echo canceller algorithms aren't doing anything. We get extreme echo during the conversation, it appears even before the call connects, the echo is there... This only happens with SIP to/from WCFXO (analog POTS). Looking at the Zaptel configuration: /etc/asterisk/zapata.conf: echocancel=yes echocancelwhenbridged=yes rxgain=0.8 txgain=0.8 (although none of the above options

All, I would like to play an announcement to the user on what external line a call came in, right before this call get bridged to this user. How would I go about implementing this in * ? Regards, Hans -- The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may

...ebbie Tropiano | debbiet at arlut.utexas.edu | | Environmental Sciences Laboratory | +1 512 835 3367 w | | Applied Research Laboratories of UT Austin | +1 512 835 3544 fax | | P.O. Box 8029, Austin, TX 78713-8029 | home email: debbie at icus.com |

Folks, I love Asterisk, have been using it for a while now. I'd like to know if anyone has some good comparison points on Asterisk vs. Vocal (Vovida) vs. GNU Bayonne. I know only a little about the later two. Also, one drawback I've hard about Asterisk (not for me, but for general consumption/deployment) is easy of configuration -- people like GUIs. They want point-n-click. I'm a

Hey all, Did I see something on here about using an AGI script to do reverse lookups via anywho.com? I have a PRI that only gets caller-id number and no Alpha. TIA, -- Daniel Jimenez <djimenez[at]pobox[dot]com>

...ebbie Tropiano | debbiet at arlut.utexas.edu | | Environmental Sciences Laboratory | +1 512 835 3367 w | | Applied Research Laboratories of UT Austin | +1 512 835 3544 fax | | P.O. Box 8029, Austin, TX 78713-8029 | home email: debbie at icus.com |

Guys, CentOS is not doing very well at this ranking site: http://www.distrorankings.com/ We need some good foo there, so if you like CentOS, go there and rank us ... {If you don't like CentOS then you are getting sleepy .... sleepy ..... You will not remember anything you read in this message} Thanks, Johnny Hughes -------------- next part -------------- A non-text attachment was

Now that I've been enlightened to the terrible write performance of ext3 on my new 3Ware RAID 5 array, I'm stuck choosing an alternative filesystem. I benchmarked XFS, JFS, ReiserFS and ext3 and they came back in that order from best to worst performer. I'm leaning towards XFS because of performance and because centosplus makes kernel modules available for the stock kernel.

...dates that connected the idea of ?here and there? and ?you and I? to something like a connection between Heaven and helium. He spoke a little bit about the idea of people becoming Holodecks in a world where that might appear to those gazing through the forward looking safe-harbor telescope of Copernicus and connecting the idea of ?becoming one with God? to a loss of individuality and failure to see that there is without doubt a reason ?you? is the heart of the word ?future? and it has absolutely everything to do with recognizing that in the spirit of Matchbox 20's ?there's a little bit of...

Ok, Skinny gurus. (btw, I'm super pleased to see development happen on this). Thoughts on this?? I added this context to my skinny.conf: [ppc] device=SEP00022D494F2A context=employees line => 50 ; Dial(Skinny/1@ppc) I've downloaded the 30 day Window eval of VTGO! (PPC) from www.ipblue.com and it hangs on Registering. *CLI> skinny debug Skinny Debugging Enabled

I'd like to play a sound to a user I dial (via SIP) once they answer play the sound and then allow me to talk to them. The new Cisco 7960 SIP code allows to set lines to autoanswer via the speaker phone, I'd like to play a "tone" after it rings through and then talk... Any thoughts on how to do this?

Folks -- I know this isn't directly an * issue, but I need to buy 14 7940s (preferably) (or 7960s if the price is also reasonable) --- no power cubes, immediately. If anyone has a good price, contact me offline at 512-427-1324 or lenny @ rocksteady.com Thanks, Lenny

I'm considering using the Agent "login/logoff" function to add to a queue that will be our "main number during the day" to answer. Periodically our receptionist is not at her desk and would be useful for her to login elsewhere and get the main number calls to transfer as she sees fit. If the agent's don't pick up in a specific amount of time, it's

Here's a simple small expect script ... I call it "phreboot", usage: phreboot IP $ phreboot 10.99.1.1 -- cut here -- #!/usr/bin/expect -f set timeout -1 spawn $env(SHELL) match_max 10000 send -- "telnet [lrange $argv 0 0]\r" expect -exact "word :" send -- "cisco\r" expect -exact "Phone> " send -- "reset\r" send -- ""

I've noticed I have an issue with my Dialplan ... apparently instead of a busy signal when the caller is busy it falls through and gets a Congestion... What's the proper syntax for this, reorder tone when there is a reorder and busy when there is a busy... SBC is a T1/PRI. [macro-sbc-outdial] exten => s,1,Dial(${ARG1}/${ARG2}) exten => s,2,Congestion exten =>

We're doing some SIP development and have a question on "additional parameters" supplied to the register (in this case maddr= and the non-standard clport= in our example below). What we're experiencing is the INVITE doesn't included these parameters and they get dropped when the INVITE is sent to the 10.1.1.97 address. Ideas? Supported? SIP Bug? REGISTER