similar to: Best VM codec for Linux/OS X/Windows environment

I'm looking at the Sayson 480i or the Cisco CP-7960. Which one would you suggest and why please ? Best Regards, Alex Brecher Visit us at http://www.Successfulhosting.com <http://www.successfulhosting.com/> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041128/8e282c51/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

Hi, all. I've got my Uniden UIP200 configured via TFTP (had to get DHCP 3.0.1 -- Debian's latest is 2.0.x!), and all seems well... except for the minor detail that it doesn't work. It registers fine with Asterisk, but when I copied my Grandstream's sip.conf info and plugged in the Uniden stuff, no dice. Any ideas? Thanks... -Ken unidencom.txt: OverwriteLocalSettings

Is there a protocol I'm supposed to use here? It seems that people are asking 100 questions a day and SOMEONE is helping them, and I've posted this three times and not even an "I Don't Know". My third repost: Ok, I've been trying to figure out why my A@H won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has

Hallo, we have started playing with asterisk about one month ago, and we do like very much what we are experiencing. Now we would like to take some step further towards "standardizing" installed modules, functionalities, tools etc. The "wall" we are facing now is: choosing the right tool for * management. We tried AMP, very powerful but incomplete (CAPI is very important to

I installed oh323 and everything seemed to go smoothly (compile & everything upto calling through using oh323). I must admit, there is some behavior that's doesn't seem right but generally, I'm able to dial-out of any oh323 device whether to an extension or to a trunk. Audio is sometimes muted when dialing out until the extension or dialed number answers. Sound quality is good

I'm looking at deploying some Polycom 501's here, but one thing that still needs confirmation before I can move forward is global paging. I figure that I can couple polycom auto-answer (http://www.voip-info.org/tiki-index.php?page=Polycom+auto-answer+config) with this script: http://lists.digium.com/pipermail/asterisk-users/2004-March/040186.html However, that script was posted over a

Does anyone know of any obscur reference for detecting an incoming fax. I currently have AMP running and everything else is working great. Installed the spandsp patches and software... using the default AMP extensions.conf, I start sending a fax, I hear it pick up and transfer to voicemail after 20s. Fax is set for system... Here is the detail from the extensions.conf [global] FAX_RX = system

I know this isn't an asterisk thing, but since the recommendation to get one came from here I figure lots of people out there have one. I read the docs, and it says that in order to do a blind transfer I should hit "flash", then dial "*__" then the number. Now, how on a normal phone do I dial "asterisk underscore underscore"? Can someone tell me how doing a

Hello All, I'm new to the list and the whole voip server side. I'm trying to setup Asterisk to just do internal dialing, no access out to the pstn is required/wanted at the moment. I'm running Fedora Core 3 with Cisco 7960's phones (running SIP 6.3). I've set it up following these guides: http://www.voip-info.org/wiki-Asterisk+Fedora+Core+3

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:

The company I work for is looking at vendors for a PBX, one of the requirements is VoIP. I have been sitting there listening to people pitch very proprietary implementations of VoIP where you are locked in to their hardware, their interface... I know a little bit about asterisk (set up a couple offices with it... run it at home...) and would like to pitch it to this company. Does someone have a

Hello All, I am going to try and install Asterisk on a server machine that I have and wanted to get the opinion of experienced people as to the best method. Initially I was going to purchase the Asterisk Starter Kit at: http://www.thevoipconnection.com/store/catalog/product_16215_Asterisk_Starter_Kit.html But from what I can find out in the reviews, the book that comes with the kit is not

Hi, I have the newbie guide from AMP's website and (fair enough) it is all about whitebox linux. Has anyone found any gotchas with the newbie guide relating to SUSE 9.2 ? Any help appreciated. Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050125/6b7a2f61/attachment.htm

Can anyone recommend any reading to help me get a better understand of VOIP technology and all the terms used in this discussion thread? I've read many documents on the wiki but I find it a bit lacking. Sincerely, Tyler

Hi all, We are successfuly running an Asterisk server with standard SIP hard phones and it is working well. We are looking to deploy some soft phones on our Linux desktops. There seems to be several floating about. Anyone out there with some good/bad experiences with particular Linux softphones. We only need g711 and prefer IAX but a SIP one will do.... Thanks....

What experience can be shared about installing and running the OrderlyQ application? I have a bunch of queues set up and want to start adding some additional apps and this one looked promising but I have very little java experience and it doesn't seem to be running properly. Jason Kawakami -------------- next part -------------- An HTML attachment was scrubbed... URL:

Phone won't register on LAN port registers but doesn't work on PC port. SIP to SIP works. Anyone have a Configuration that works out there? Phone has 4.63 Firmware Asterisk@Home 2.0 Beta -------------------------------------- Jeff Herring / jeffh@sldsi.com Seacoast Laboratory Data Systems, Inc. Voice: 603 431 4114 x14 FAX: 603 431 2112 --------------------------------------

does anyone know of a 2.4 or 5 ghz cordless phone system that has an ip base station? thanks

I have read every thread, I have Redhat 9, Asterisk 1.0.6, 2 T1 lines connected via TE405P. Everything works great, except MOH. I added an exten with MusicOnHold(30), and it plays just fine. Conferences have music when no one is in. I have SIP phones. When I place a call on hold, the CLI give no indication the call is on hold. I have set musiconhold(default) everywhere, removed it from everywhere,