similar to: asterisk gui?

Hello, I am just building my first Asterisk server. Looking for a couple of good quality ip phones. I like the Cisco 7960, are they easy to configure to work with Asterisk? What are the alternatives, with a good speaker phone, and simple clean and stylish look like the Cisco? I appreciate your advice. Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214

Hello, Is it possible to connect the Panasonic KX-T7665 IP phone to Asterisk? Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214 M: 0419 568 217 E: Michael@ITMedic.com.au http://www.ITMedic.com.au Keeping your computer systems healthy.

Hello, My asterisk currently will dial an outside number after I dial the number and press send on the phone. How can I get it setup so I have to press '0' for an outside line. Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214 M: 0419 568 217 E: Michael@ITMedic.com.au http://www.ITMedic.com.au Keeping your computer systems healthy.

Hello, Does anybody have the latest Boot ROMs for the IP500 and IP 600 Polycom phones. I have one of each and can't find the Boot ROM v 3 anywhere to download. I would also love a good sample phone.cfg and sip.cfg files from an Aussie asterisk user to look at. Also the ip500 is having problems trying to load the bootrom 2.6.2 ? Any ideas? Kind regards Michael Felder IT Medic

Hello, I am looking at the Polycom phones. The ip600 has a very nice screen, is that the only real advantage over the ip500 and ip300.. Is it worth the extra dollars? Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214 M: 0419 568 217 E: Michael@ITMedic.com.au <blocked::mailto:Michael@ITMedic.com.au> http://www.ITMedic.com.au

Hello What is the difference between these 2 version of Asterisk in terms of functionality. For a small office am I going to run into problems if I use the easy version... Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050717/311c56ec/attachment.htm

Hello all, The "Secret Agent" final release of the Asterisk Management Portal is now available for download: http://amp.coalescentsystems.ca/ This exciting new release adds a great deal of functionality and flexibility. Thank you for all the contributions and feedback! 1.10.007 - Added AMP Users (multi-department, basic multi-tenant) - Added incremental upgrade script

Hello all, Asterisk Management Portal 1.10.009 has now been released. This exciting new version has several notable additions (listed below). The AMP homepage is http://amp.coalescentsystems.ca. Here you'll find links to the download, install guide, and documentation wiki. As usual, please use amportal-users mailing list for discussions about AMP:

Hello all, Can I execute linux command like(ls, mkdir) through the Manager interface? I can't seem to access the manual at digium.com. I keep getting 'Forbidden' error. Looks like they are upgrading or something. CF

Hi all, I know that this has been passed around before, and I know that it happens about every 3 months or so, but evertime the answers change, so I thought I would pass it around again. A company I work for has 3 incomming lines and 4 phones. They require voicemail and MOH. Their phone systems VM hard drive died today, they were quoted a $2000 to replace it. I started to talk to them about

All - I've a new system, that since it's been in production, has seen a few issues, that look like they should be fixed by upgrading asterisk @ home to the latest version. I was curious if anybody out there can tell me their experiences with this, and what to expect. Thanks, Rolf Brusletto

Hi, I am performing experiments in the field of visual perception where we often apply balanced designs. Within a group of normal subjects, we vary different stimulus conditions (like contrast, luminance, temporal frequency of stimulation) and derive some psychophysical or electrophysiological results from our subjects. Often, the main question is to test the effect of these parameters

Hi there I am new to Asterisk and am looking for a web based management tool, for managers to manage hunt groups, extensions etc and for user to have access to there own phone features. I have seen there are a number of commercial tools available for this, but I presume there are some freeware options too I noticed one that I like at http://www.thirdlane.com/screenshots.htm but I am assuming

Hi all, I'm trying to setup a TFTP server to serve as repository for the config of all my Cisco network devices. As per the the tftpd man, I've added the "-c" option into the /etc/ xinetd.d/tftp (as follows) but I still cannot get write access (unless the file is already present). [root at chl1 ~]# cd /etc/xinetd.d [root at chl1 xinetd.d]# cat tftp # default: off #

Hello, I've just spent several hours going over several Google searches trying to find a way to configure rsync to log into a file named "/var/log/rsync.log". So far, every instance where I've found someone asking about rsync logging remained unanswered (which is kind of weird in itself). As far as I can tell, the only way to do this is to setup rsync as a daemon process so

Hi there. Is it possible to have different sip users have the same CallerId number in sip.conf. I need this because we got multiple companies on this Asterisk box. Company A's internal numbers: CID: User: 1000 - User 1 2000 - User 2 3000 - User 3 4000 - User 4 Company B's internal numbers: CID: User: 1000 - User 5 2000 - User 6 3000 - User 7 4000 - User 8 Is this allowed? Regards

hi My stations are behinds a firewall, the system is windows 2000 & 98, i use sjphone asterisk is on the internet gateway where is the firewall Shorewall the system is linux debian (sid) kernel 2.4.20 j do whaton http://www.automated.it/guidetoasterisk.htm (grateful Andy) to write my sip.conf but i can't call an external sip user. (an external user can call me) i try without asterisk with

Hi all, I curious to know if other UIP200 users have this same issue: You flash (XFER button) to consult-transfer a caller to another extension. If the transfer target party is unavailable (ie: voicemail), there appears to be no way to get the original caller back. If it's a known limitation, has anyone come up with a functional work around? Thank -- ..................................

I would like to get in contact with users/consultants who are or have worked with the Cisco phones and Asterisk to trade information. Cisco has reluctantly made SIP available on their phones and most of the information on voip-info and other wiki's appears to be reverse engineered. There is a wealth of information out there which is terrific. I have a client with about 40 phones

Hi , I want to build a PV kernel of 2.6.18-xen on x86_64, but I don''t now how to configure the kernel for PV. Is anybody can give me some advices, or give me a .confg file if you have one. Thanks very much! _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users