similar to: Special Delivery from China

Dear Harry, Thx for the explanation. My team manage building's PBX that use Asterisk 13.x. We use Asterisk PBX for this buildings that have apartment and office customer. >From my Asterisk PBX, we connect to IPPhone (yealink) or ATA Converter (cisco SPA112). Others are using PBX like panasonic analog, audiocodes SBC, etc, and we use ATA Converter to convert from SIP to Analog (CO Line)

Thanks Harry. I will check and revert. I hope it work perfectly with asterisk. Regards On Wed, Feb 17, 2016 at 8:32 AM, Harry McGregor <hmcgregor at biggeeks.org> wrote: > Hi, > > For analog, I really like telco grade channel banks. > > I would recommend the adit 600, there is a good market on Ebay, and you > can do 48 channels per adit 600, with 2 T1 interfaces. Having

Hi, I am thinking of starting a project at Work to pilot to use of Astrisk and VOIP, and would like to tie in several projects together. Currently we are looking at purchasing an ISDN and H.323 based video conferencing system from Polycom. It suggests the use of 3-4 BRI lines (ie 6-8 B channels, but in the form of individual BRIs). We need ISDN for the video conferencing to communicate with

Hi there, what phones are available that have two ethernet ports? I want to do some cabling at a new installation and i heard there are such phones (SIP i guess) out there. That way i dont have to run two cat5 to the user desktop. I think 3COM had one but can't find the web site reference for the two port phone thanks, erick

On Wed, Feb 17, 2016 at 8:14 AM, Mitul Limbani <mitul at enterux.in> wrote: > Sangoma 50 port FXS Thanks. Will I now stack 20 boxes in order to achieve the 1000 FXS lines? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160217/0d4c2800/attachment.html>

Hi, Is there anyone here who has experience connecting Asterisk (ver 13.8) with PBX Panasonic KX-TDA600 ? The architecture more less like this : Telco Sip Trunk ---> Asterisk 13 ---> Panasonic KX-TDA600 ---> Phone / Fax Thanks in advance, Regards, Ikka - Jakarta, Indonesia -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I can get Grandstream 100 SIP phones for EUR 75. I'm not sure about the pricing for these in Europe, so I'd like to hear from people here whether that is a reasonable price for them? TIA & BRgds -- Francesco Peeters ---- GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit

iaxComm is an Open Source softphone for the Asterisk PBX. iaxComm compiles and runs on Win32, Linux and Mac OS X (Panther) systems. Recent Changes: * Improved jitterbuffer code * Steve Underwood's Packet Loss Concealment Code Features Include: * iLBC support * GSM support * speex support * ulaw and alaw support * Blind Transfer. * Custom Ringtones per

I'm just wondering if anyone knows the story with these... http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&rd=1&item=5721202362&ssPageName=STRK:MEWA:IT He claims they support IAX2 and SIP... but almost no history on the account selling them. I didn't see anything in the wiki about this company either.. Does anyone have any history with these phones? Thanks, Jared

I am trying to add a Polycom IP 3000 to our Asterisk system and am not getting anywhere. h323.conf [8908] type=friend host=192.168.104.25 secret=polycom context=crv-default callerid="Conference Room Polycom" extensions.conf exten => 8908,1,Dial(h323/polycom,20,Ttr) ; Polycom exten => 8908,2,Hangup I have tried setting the Asterisk system as both gatekeeper

Hi all, I have a VOIP phone with the PA1688 chip; my firmware is V1.42.028. This IP phone is registered in an Asterisk PBX and I've a problem when I dialing internal number. If I dial an internal number, like for example 102, the IP phone takes 35 seconds to send the number to Asterisk; here below the debug output 192.168.0.75: first 192.168.0.75: 1 192.168.0.75: 10 192.168.0.75: 102

Hello, I am getting the following error: | rsync: open connection using rsh conn /usr/bin/rsync --server --sender -v . "/home/andrewm/test-rsync/tree/*" | large_file2 | write failed on large_file2 : Success:06 | rsync error: error in file IO (code 11) at receiver.c(271) | rsync: connection unexpectedly closed (201 bytes read so far) | rsync error: error in rsync protocol data stream

Pardon, this might be off-topic. I'm reading: http://en.wikipedia.org/wiki/Network_switch For a setup of ~5 agents, would I be wrong in thinking that a generic 16 port unmanaged switch would fit the bill? The first model to come up for me in an Amazon search is: http://support.netgear.com/product/fs116 Is this a reasonable choice? Would I be wrong in thinking that most any Fast

Hi, I've got two pa1688 phones that I want to set up to communicate between branch offices without a gatekeeper. Both phones will be behind a firewall and I want to use port forwarding so the phones can communicate. I tested the phones behind a firewall on the same network segment and there were no problems at all using sip. However, I then moved the phones into situ and port forwarded

On Fri, 20 Feb 2015 13:05:56 -0700, Harry McGregor wrote: > For a very basic setup it would work, but I would suggest POE at a > minimum, and vlan support if possible. > > Gigabit uplinks, 10/100 for the poe ports > > http://www.amazon.com/NETGEAR-ProSAFE-M4100-D10-POE-Ethernet-Managed/dp/ B00AUEYX0Y/ref=sr_1_3?ie=UTF8&qid=1424462577&sr=8-3&keywords=netgear+poe

I am about to buy AT-320EE phones for use together with asterisk. Has anyone used these phones ? Are there any known problems using them using Asterisk ?

Does anyone know if there are additional SIP devices out there which aren't phones? I'm basically looking for a fully-automatic SIP speakerphone. I'd like to be able to dial a sip-extension and make an announcement (PA) and/or simply listen in to a room (baby-monitor). Yes, I know, some of the more advanced phones can be configured to behave like that, but it seems to a waste of

I hit a small snag using Dovecot-imapd smoothly in my environment with maildir and most of my accounts in LDAP. Since the accounts are created through a web interface on another server home directories on the mail server don't get created automatically. There's the handy pam module pam_mkhomedir.so to automagically create home directories, but unfortunatly Dovecot wasn't calling

Dear all, I'm trying to make an R code for the drought severity index (DSI) developed by Philips and McGregor (1998). You can refer to the description of the algorithm on page 19 from http://dissertations.port.ac.uk/229/01/MaloneS.pdf The code is given in Excel as the following and can be found on page 60 from the same link. C7 = IF(C6<0,IF(@SUM(A6:A1)<0,C6+A6,"0"),

I'm trying to get it for my new palm but I cant seem to find it anywhere. If I have reached the right person could you please link me or send a copy as an attachment? Thanks, Dennis Daigneault Sales McGregor Hardware Distribution 604-253-7785 1-800-663-5625 Fax 604-253-7785