similar to: Cisco remote reboot

Anyone? :-) > > ----- Original Message ----- > From: "Matt Klein" <mklein@nmedia.net> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Sent: Wednesday, December 29, 2004 7:37 PM > Subject: Re: [Asterisk-Users] Cisco 7690 Voicemail Problem > > >> >> a faint scratching sound

is there a way to support login and logout functionality in a phone? We are using Cisco 7940 and 7960 phones and have 2 shift. We want to be able to use the same phone using like 2 different extensions. The phone will then "remember" your settings if possible, if anyone has left you a voice mail etc. Is this possible? Regards, Paul -------------- next part -------------- An HTML

With my thanks to Brian West and his offering in the thread, "Subject: Re: [Asterisk-Users] Remote reload Cisco 7960" I offer PolyReboot.pl, a perl script for rebooting Polycom IP Phones PolyReboot.pl takes an IP address as a single argument and reboots the phone. You must have a cfg file in the Polycom style, i.e., 00ab00cd00ef.cfg - all lower case. Further, you need to use ftp for

Shoval, Interesting Mention. I agree, most people don't have CO exp. And I wish daily I had enough. Understand that what I mean by my e-mail is consumer side FXS ports, in broader terms, I mean, customer picks up a phone line, it signals a channel bank which signals *. 24 of those channels. Not channels equipped to Send Signal to the CO that a loop has been made.. meaning FXO. 24

Kevin wrote: > There is a reboot script posted on the wiki to reboot Polycom > telephones. When I execute this script, I get the following messages. > I am concerned as this is causing issues with asterisk and the PRI. > Does anyone have any ideas why this would be happening? > > > > asterisk console: > > -- Remote UNIX connection > -- Remote UNIX

Hello, I'm trying to setup the revised Polycom remote reboot script as found on: http://www.voip-info.org/tiki-index.php?page=Polycom+reboot+hardphone+script I'm not sure how to use this script, it's just a perl script, so I tried creating an executable perl script and running it, but I get the following: [root@asterisk1 agi-bin]# ./polycom_reboot.pl 192.168.3.205 Checking ARP

Hello, I hope this isn't too off topic, but I'm attempting to set up a Spectralink 8002 Wifi phone with our Asterisk installation, and seem to be running into a brick well (more of a wall than others that have posted their experiences). My problem is that the phone boots, associates with the wireless, grabs an IP (tried static too - same thing), contacts the TFTP server for firmware, then

any way the you can get * to send a NOTIFY SIP message to all SIP phones? to have the SIP sets recheck thier configs etc?? Like this? NOTIFY sip:sip@192.168.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP ipaddress:5060;branch=1 Via: SIP/2.0/UDP ipaddress From: <sip:webadim@192.168.0.1> To: <sip:sip@192.168.0.3> Event: check-sync Date: Mon, 10 Jul 2000 16:28:53 -0700 Call-ID: test@192.168.0.1

Hello List, This is what I want to do: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I'm trying to investigate and solve some postgres latency spikes that I'm seeing as a result of some behaviour in the syncer. This is with FreeBSD 8.2 (with some local modifications and backports, r231160 in particular). The system has an LSI 9261-8i RAID controller (backed by mfi(4)) and the database and WALs are on separate volumes, a RAID 6 and a RAID 1 respectively. It has

Hi, I am trying to install xen under Fedora Core 4 and when I try to boot it, right after selecting xen from the grub list, I get an error. It says kernel /boot/xen-2.0.gz dom0_mem=131072 Error 15: File not found I installed all the prereq''s., did ''sh ./install-sh'', added the three line to /boot/grub/menu.lst, and disabled the tls. Is there anything else I have to

I have looked at matt's asterisk gui client at sourceforge. I am not a programmer by trade. The documentation there seems to be a bit lacking. Has anyone have the experience in installing the gui client and may perhaps have a how-to document available for sharing. -- David Kwok Tel: 612 99292086 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 -------------- next part -------------- A non-text

Has anyone used either the 8 port or 4 port FXO device from Grandstream? (GXW-4108 or 4104). They seem to be the lowest cost multi port FXO devices that I can find, so I'm getting ready to buy the 8 port version. I just want to see if there are any opinions on the device before I commit to the purchase. If people have not used the Grandstream, are there any issues with using

https://bugs.freedesktop.org/show_bug.cgi?id=98030 Bug ID: 98030 Summary: Stuttering video playback in totem after update to 1.19-rc1 Product: xorg Version: unspecified Hardware: Other OS: All Status: NEW Severity: normal Priority: medium Component: Driver/nouveau

Hey Guys! I have perl script for allpage which is working fine with asterisk 1.8.2.3 version but same script same dialplan wouldn't working on asterisk-1.8.3.2 is there anything changes ? If i run this script from command like it works but not from asterisk dialplan. This script nothing but just connecting AMI interface and using Variable: SIPADDHEADER=Alert-Info: Ring Answer variable to

Hi, I'm trying to write a simple perl script that will run the following: Action: Originate Channel: local/xxx@callback/r/n Exten: 1234 Context: callback Priority: 1 Extensions.conf exten => 500,1,agi,callback.pl callback perl script: use Net::Telnet (); $mgrUSERNAME='fred'; $mgrSECRET='bloggs'; $server_ip='127.0.0.1'; $tn->print("Action:

Hello. I have been beating my head over this problem for about 6 hours now. I have a SIP peer, who I register to (successfully), who should be directing all incoming calls at my [default] stanza in my extensions.conf: [ Context 'default' created by 'pbx_config' ] 's' => 1. Wait(1) [pbx_config] 2.

I am having some issues when trying to connect with perl to the asterisk manager and doing an "IAX2 show channels". If i do that on a server that is heavily loaded, i sometimes get some events instead of the channels i asked for. Any suggestions how i could fix that ? zoa.

Hello guys, I have been looking for a way to call a cell phone after someone has left a voice mail and allow the user to enter the mailbox password to listen to it and found a very old entry of vmoutcall. I tried unsuccessfully to get it to work with 1.4 and is beyond me. Has anyone gotten this to work or has a way to accomplishing this? Most analog pbx's have this feature and I am amazed

## Not run: from <- sprintf("<sendmailR@ to <- "<olafm at datensplitter.net>" subject <- "Hello from R" msg <- "It works!" sendmail(from, to, subject, msg, control=list(smtpServer="ASPMX.L.GOOGLE.COM")) ## End(Not run) the above commands are provided in this document ie http://cran.r-project.org/web/packages/sendmailR/sendmailR.pdf