similar to: Remote reloading Cisco phones...

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

I took your suggestion and turned my (ill advised) sudoers bash script into an expect script! It works a lot better this way and is more secure. Because I'm not trying to store a password in a script (which I recognize as a bad idea anyway, I I think I've learned my lesson here). It really works well. But the only thing I'm still trying to figure out is how to put a if statement in

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

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) --

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

I can't login anymore... used to be able to. Timing doesn't seem to be working well any ideas? Also what is this "NOTICE" I'm getting? *CLI> == Accepting call on 'SIP/lenny-b19c' ("Lenny Tropiano" <5555>) -- Executing VoiceMailMain("SIP/lenny-b19c", "") in new stack == Parsing

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

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

Hello - I never heard back on this and continue to experience this bug. Can someone help me to locate the patch for this so that I can be sure it's included in my source? It's not happening on every boot, but about every other or third. And just as described in the bug report (the boot works OK, but at the end rather than GDM starting nicely, I get a "fuzzy" screen and the

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>

Does anybody have any insight into how to make this faster? I suspect, that the rounding going on may be an issue, as is the stepping through data frame rows using integers ... If you have the patience to teach a noob, he will highly appreciate it ;0) Joh digit <- 4 for (minute in seq(from=25,to=lrange[2])){ # Extract all data associtaed with the current time (minute) frame <-

Does anyone have a working way of having a Cisco 7960 reload its config remotely. I have tried some of the scripts that I have found on the web, but to no avail. Thanks for the help. B. J. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040116/aa4eda3c/attachment.htm

Hello - One of my users would like a kernel parameter changed. We're running CentOS 4.2 so I'm not sure if we need to upgrade the kernel for this change or not. He's seen that in the 2.6.11.12 kernel "PIPE_SIZE is defined in include/linux/pipe_fs_i.h and takes on the value of PAGE_SIZE" He wants PIPE_SIZE to be increased to 8*PAGE_SIZE. Can this be changed in the 2.6.9

Hi, Running 9.2-PRERELEASE #19 r253313 I got the following panic Fatal trap 12: page fault while in kernel mode cpuid = 22; apic id = 46 fault virtual address = 0xffffff827ebca30c fault code = supervisor read data, page not present instruction pointer = 0x20:0xffffffff81983055 stack pointer = 0x28:0xffffffcf75bd60a0 frame pointer = 0x28:0xffffffcf75bd68f0

Hello - I've been seeing system crashes similar to this (http://bugs.centos.org/view.php?id=1045) on my Rocks head node. Instead of the standard kernel, I've got a SNARE patched 2.6.9-34 kernel running. Initially I'd used the RHEL kernel source to build it (due to laziness ... it was easier to track down :-), but have just rebuilt with the CentOS kernel source and am now seeing these

Hi All; In cisco IP Phone, I can assign for a features for the buttons at the Phone (one button to be for call pickup and one button to be for call forward and button to be for bridge and one button to be for another extension). How can I do the same thin in the Cisco IP Phones (or other IP Phones) while they are registering in Asterisk? Regards Bilal

Hi All; How to configure the buttons in the Cisco IP Phones to be used for different functionalities like "Call Forward, Call Pickup, ... etc"? For example, if I need to assign one of the buttons existed at Cisco IP Phone to be used for CallFrw, how to do this in Asterisk? Regards Bilal

I have the following in extensions.conf: [global] MP3ROOT=/var/lib/asterisk/mohmp3 [default] exten => 1111,1,Answer ; Answer the line exten => 1111,2,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten => 1111,3,MP3Player(${MP3ROOT}/sample-hold.mp3) The command that runs is: 14030 pts/0 S 0:00 /usr/bin/mpg123 -q -s -b 1024 --mono -r 8000

Hi, I heard some time ago that, when running a SIP firmware, Cisco hardphones needed a Cisco call manager to get localized (ie non-english) menus ? Is it still true ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081212/fcc22a7b/attachment.htm