similar to: Newbie Question: Building an Asterisk systemto replace an old PBX but using existing phone

I have a brief from a local hotel to build a PBX using Asterisk but they want to use their exisiting telephones and wiring from an old PBX that no longer works. Basically, I can build the system but an looking for a card that will allow for upto 20 extensions to be wired into the back of the PC. Doeas anyone know of a solution to this Sean-- ICQ: 679813 FidoNet: 2:263/950 Jabber:

Ok, I am going for A@H with the CentOS iso disk. Installed and just checking a few things out. My other question is this, which I forgot to ask before. We have no Broadband here and more than likely will never have, so I am just looking at building Asterisk to handle inbound and outbound calls, at home via a ISDN card and for my hotel job via a PSTN line setup. Is this very complicated to setup

Log report is reporting a lot of these lately.. following is just a short snippet from the beginning on one server. WARNING!!!! Possible Attack: Attempt from 104.29.broadband2.iol.cz [83.208.29.104] with: command=HELO/EHLO, count=3 : 1 Time(s) Attempt from 106.7.broadband7.iol.cz [88.102.7.106] with: command=HELO/EHLO, count=3 : 1 Time(s) Attempt from

You are right. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tarpo, Louie Sent: Thursday, August 11, 2005 5:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Newbie Question: Building anAsterisk systemtoreplace an old PBX but using existing phone You write out a

Recently we began seeing lots of these log entries on our off-site mx smtp host. I have googled this but I am not clear from what I have read if this is something we can stop altogether or should even worry about. Comments? Logwatch. . . --------------------- sendmail Begin ------------------------ SMTP SESSION, MESSAGE, OR RECIPIENT ERRORS ------------------------------------------

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi folks, I currently use courier imapd, exim and ldap on my mail box. There are no users on this box, all info is taken from the ldap server. I note that dovecot does not use ldap as a backend, but curious if this in on the cards, or even mysql backend. Sean - -- Sean Rima http://www.tcob1.net Linux User:

Hello! I have set up tunnel between a FreeBSD machine and Windows Vista. Tunnel is established, but when I try to ping either end ping fails. I have temporarily switched off firewalls on both machines, no luck. Here is client tinc.conf on Vista: Name = lenovo_client ConnectTo = lenovo_server Interface = tinctap Subnet = 10.20.40.0/24 Sevrer tinc.conf on FreeBSD: Device=/dev/tap0

Hi, I am about to buy ip pbx asterisk system but what ip phones do you recommend? Are polycom ip all functional with the ip pbx system??? Be waiting.thanks a lot Marlo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050811/16cc52cd/attachment.htm

Chris Mason (Lists) wrote: > Cory Andrews wrote: >> >> >> >> >> IP430, will sit between the IP301 and IP501, IP430 will have dual >> Ethernet, PoE, and full duplex speakerphone. List price (MSRP) $239 >> street price should fall likely between IP301 and IP501. >> > That looks great, the 301 is almost useless due to the lack of speaker

Hi! I get this strange behaviour... I don''t know how some packets get into wrong rules. My rules are those: 0: from all lookup local 50: from all lookup main 201: from 192.168.17.0/28 lookup 201 202: from 192.168.16.0/28 lookup 202 222: from all lookup 222 32766: from all lookup main 32767: from all lookup default Table main has: 192.168.17.0/28 dev eth2

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Jerry Glomph Black > Sent: Monday, January 30, 2006 11:59 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Polycom IP301: Pass-through ethernet port > unusable? > > Have just done a

Is there anyway to prevent an extension from dialing certain codes. ie I want to prevent extension 203 from dialing number which start with 00 087 086 etc Sean -- +----------------------------------------------------+ |VOIP: FreeWorldDial 689482 VOIPBuster thecivvie | |GPG Key: http://thecivvie.fastmail.fm/thecivvie.asc | +----------------------------------------------------+ --------------

Using the install instructions for A@H, I setup a FWD account, this I tested using X-Lite and it works okay, Nowever I cannot make calls to fwd using Asterisk, my log showes: Aug 14 21:06:09 NOTICE[1324]: Registration of '689482' rejected: Registration Refused Aug 14 21:06:59 NOTICE[1324]: Registration of '689482' rejected: Registration Refused According the to info screen of

Hello everybody. First of all thanks to all the people giving their opinion on the subject I proposed: "Trixbox vs. custom install". You've all been very helpful. I try to summarize what has emerged from the various messages. Forgive me if I miss or forget something or if I simplify too much some of your messages... - Elman Efendiyev says that you should install from sources if

Good day, I have a puzzling issue that people in the IRC channel recommended I post to the list so here goes :) I am trying to call a SIP softphone from an H.323 hardphone. The hardphone is connected to a Definity Prologix R12 PBX with a MedPro card and a CLAN. The Avaya is setup to send any call to extension 1609 down an H.323 trunk group that is destined for the Asterisk server. When I call

Hi; I'm looking for a Polycom distributor in the UK who can supply a small number (around 20) IP301 / IP501 handsets. Can anyone recommend someone ? jd -- John Daragon john@argv.co.ok argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127

Does anyone on the list have a script for notifying pagers that they would be willing to share? I have found a reference in the archive to such a script, but previous attempts to find the author of that posting have failed. Anyhow, I am looking to set up a system whereby a message is sent to a pager when a voicemail is left in a specified mailbox. (This is easy, it's built-in to

Hi All, I'm looking for a good voip hardphone that has a decent set of the "regular" features (conference, 2 lines, etc) thats reliable, has decent quality, and isn't too pricey. Does anyone have any suggestions? Thanks in advance. Derek -- Derek Fedel

Hi, I have configured a certified asterisk 13 server with chan_mobile and res_pjsip. I have a Cisco 7940 hardphone and I use ekiga as softphone client. Now the problem is, using the hardphone I'm able to call the softphone and hear everything properly. But when I call from the hardphone to some number that has to be dialed via chan_mobile, I'm not able to hear what the other side says (I

Hi list, I just installed Asterisk 1.4.38 (on an updated Centos 5.5 machine) and am getting this error : WARNING[6472]: res_musiconhold.c:856 moh_scan_files: getcwd() failed: No such file or directory with the default musiconhold.conf file. When I change musiconhold.conf to this: [default] mode=mp3 directory=/var/lib/asterisk/mohmp3 (and have converted all the wav files to mp3 and put them