similar to: New to Asterisk - 2 question

Hi, I'm a bit new to phone systems technology, so sorry if this question may sound uninformed. I want to put together a system of about 20 stations. What I'm invisioning is a system where about 16 users have a inexpensive handset hooked up to their computer via some sort of modem and the computer would run their usual Windows apps with a client that serves as a more complex

I hope this isn't considered cross posting, i sent the following email to Digium support but figured someone on the list may also have better insight into my questions. I have purchased 2 g729 licenses from Digium for testing and have the following questions; ** My configuration is a single asterisk box configured with 2 g729 licenses and 2 x Cisco 7960 Phones, I have confirmed the

When using a channel bank for analog handsets, you have a couple options in the way you handle transactions involving the analog handsets and origination. With immediate set to no, it appears to me that soon as a digit is pressed after going off-hook, the single digit is taken and processed against the context that the channel is associated with from the configuration in zapata.conf. With

I have a small request re. R FAQ Frequently Asked Questions on R Version 1.5-10, 2002-06-13 ISBN 3-901167-51-X Kurt Hornik In section 6.2 Should I run R from within Emacs? The faq says "Yes, definitely." This led me to install emacs, and to install ess. However, I use a windows environment (mswindows2000) and I had some difficulty installing, linking and launching these

I have a client that has 5 Snom 320s. 4 work great, one does not. I upgrade the firmware to the latest (6.5.2) and the problem goes away, but then comes back a couple days later. There is a slight packet loss on the phone (about 1%), though there is no packet loss on any of the other phones. I determine the packet loss by the Linux command "ping -f -c 10000 192.168.2.10". Outgoing

Basically, pretty much all the 2 line cordless systems I've seen come with a built in digital answering system that I'll never use, the main problem with this is that these units don't support VMWI (visual message waiting indicator) with telco supplied voicemail. This is a problem because I'll be setup an Asterisk system in the next month or so to handle my 2 analog and 1

I'm trying to use Asterisk for its voicemail capabilities while interfacing with a legacy Toshiba PBX. Is there a way to have Asterisk send a DTMF code to an extension to turn on the message waiting indicator light? When a user leaves a voicemail, I want Asterisk to pick up one of the lines attached to it, and then dial #63<ext>, which is what sets the message waiting indicator light

I want Asterisk to delay answering the POTS line via a Wildcard (a Zap channel) by some period of time, either a number of rings or just a number of seconds. I have tried this: [from-pots] exten => s,1,Wait(30) exten => s,n,Answer ... exten => s,n,Dial(SIP/brian&SIP/joe,10,H) exten => s,n,Voicemail(u2001) exten => s,n,Hangup exten => s,103,Voicemail(u2001) exten =>

Does anyone know if X-Ten or SJPhone support multiple cordless handsets for multiple lines? I have an office with multiple roaming users(nurses) that are in and out. I'd like to provide them telephones, and my idea is to have a PC sitting in a corner somewhere running a softphone client. When a nurse comes in she just picks up any available handset(anywhere from 2-5 per office) and starts

Hi All, I hope someone can help. I have a system that I have recently upgraded to latest CVS and my voicemail is not working from mysql database. I get an error on the console saying " No entry in voicemail config file for 'number'" whilst there is an entry in the database for the specified number. It seems like app_voicemail is no longer checking the database even though

I understand there has been some issues sending DTMF tone through Broadvoice. Can some provide me with symptoms? --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.700 / Virus Database: 457 - Release Date: 6/6/2004

Hey gang, I'm looking at escaping from a Nortel Meridian CISC system to Asterisk/Digium/SIP phones. I'm currently in the testing and proof of concept phase. I'm going to need a SIP phone and don't want to re-purchase and have "orphans" around. We currently run Nortel 7310 phones and they work great. I'm sort of overwhelmed by all of the different IP phones. I

While I'm asking about the Polycom ip500, the answers for all phones where mic/handset/headset levels are adjustable would be of interest to many I'm sure. For the ip500, the default value for the handset seems to be voice.gain.tx.analog.handset="3" I've noticed that echo all but goes away when one reduces the mic volume on almost any phone. My question is, for you users

I have SIP handsets which allow the user to forward a call to another number after a specified interval of ringing time. On the SwissVoice this is refered to as CFNR (Call Forward on No Response). What actually happens is that after a specified period of time (default 15 seconds), the handset sends back a "302 Moved Temporarily" response to Asterisk. The problem is that when Asterisk

I'm looking to install a basic asterisk system for my church with: 8 inbound sip channels 8 sip handsets basic voicemail room to grow (maybe doubling each of the above) What would be a recomended system as to needed processor and memory? Thanks, Eric

I am trying to set up the Message Waiting Indicator (stutter tone/light) so that my cisco ata-186 will let my phones know there is a message waiting. However this does not seem to be very well documented. I found this on wiki mailboxnumber@context ... where does that go? Do I put it in my SIP.conf definition for my cisco ata, or where. In my SIP cisco definition i already have a

I'd like to allow my users to forward their calls using the forwarding feature on their SIP handsets and continue to receive Queue() calls. Currently I set the 'i' option in Queue() so that if a user forwards to their cell phone, or any other extension that has voicemail, the voicemail doesn't eat all the calls to the queue. I'm aware that I can configure the queue to

Folks, We have several people using SIP softphones in the office. When they leave for the day, they power down their workstations, causing their registration with Asterix to quickly timeout. Here's the entry for one such extension in extensions.conf: exten => 8102,1,Dial(SIP/someone,20) exten => 8102,2,Voicemail(u8102) exten => 8102,3,Hangup exten => 8102,102,Voicemail(b8102)

Now that I actually try and google for it, I can't find any dual mode GSM/DECT handsets, only pages telling me that they exist without any actual information!!! Does anyone know of any such handsets? (and even better, ones that are available in Australia) I've searched a few of the major gsm manufacturers (nokia, Panasonic, sonyericsson) but their web sites are absolutely pathetic to the

Version 1.0.5.23 is now available from http://gs-firmware.gratissip.dk/ Or directly from Grandstream at http://www.grandstream.com/BETATEST/Release-b21p1.0.5.23.zip Release notes doc here http://www.grandstream.com/BETATEST/Release_Note_1.0.5.23.doc while on the matter I just want to extend a note of thanks to Grandstream, I had 2 early handsets of theirs fail recently (about 9 months old)