search for: comdials

Hi! I've got a Comdial PBX that I would dearly love to replace with an Asterisk box. However, for various reasons, it appears not to be in the cards. Regardless of what management does, or does not, want, our current VM solution -- some Dialogic card with a "KeyVoice" application -- is dying. I'm 90% sure it's hardware. I'd rather shoot myself than replace the

I've been doing a lot of reading over the last few weeks on Asterisk, and will be implementing a test system this week to play with. I've got two questions in regards to the ideal implementation for our company. First, has anyone written any drivers to interface with proprietary phones? Specifically we have a comdial system and if we could use our existing 35 phones instead of having to

If your vmail is connected via serial to your PBX it is most likely using SMDI for MWI which isn't supported by Asterisk. I seem to remember that this was submitted as a feature request with a bounty tied to it. Keith Date: Tue, 7 Dec 2004 12:58:11 -0500 From: George Herndon <gherndon@eyeontech.com> Subject: Re: [Asterisk-Users] Comdial PBX -- can use Asterisk as VM

Anyone know if asterisk will work with a 1995 Comdial Unisyn system? Or know where I could located Execumail? It's an old voicemail software package for Comdial Unisyn phone systems. It runs under DOS. I can't seem to located the software Execumail anywhere for our old system and was hoping I might be able to use Asterisk with it. Regards, Chris -------------- next

I'm having (I think) timing issues in relation to bridged T1-T1 calls via dynamic spans. Fax calls are intermittently working, but voice is fine. My box has a Sangoma A400 inside it as the primary Zaptel timing source. My T1 PRIs that are hooked to the box come in via a foneBRIDGE2 (dynamic TDMoE spans). PRI #1 is the telco and PRI #2 is an existing Comdial FX-II. For some reason, bridged TDM

...rom: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Andrew Latham Sent: Wednesday, November 01, 2006 5:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: Newbie Questions Ken If these are older comdials then they are just analog phones with "extra signaling". The extra signaling could be on the main twisted pair (likely) or on the next twisted pair as data (9600 baud modem) like some of the nortels do. Always remember that it would cost the companies a ton to make every system totally...

One of the connections my asterisk PBX has is an analog extension from a Comdial hybrid. On the Comdial system, message waiting is turned on by dialing *3 and then the station number. It is turned off by dialing #3 and the station number. I was wanting to have Asterisk (or Comedian mail) set the message lamp in the Comdial system when a new message arrives for a user, and extinguish the lamp

...rom: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Andrew Latham Sent: Wednesday, November 01, 2006 5:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: Newbie Questions Ken If these are older comdials then they are just analog phones with "extra signaling". The extra signaling could be on the main twisted pair (likely) or on the next twisted pair as data (9600 baud modem) like some of the nortels do. Always remember that it would cost the companies a ton to make every system totally...

Given: Queue(foo|tHnr||bar) where queue foo includes something like IAX2/gw/18005551212 should # transfer work on the remote phone? A read of app_queue.c looks like it ought to work, but all I get is dtmf sent to the caller. (Incidently, I'd really prefer to be able to hit eg * during the announcement to have app_queue continue on as if there were a timeout. Has anyone looked into doing

>But currently, I only have one ethernet jack per office. Routing >another 60 or so ports would add a very substantial expense in both >cabling and backbone expansion (what category ethernet is required, >BTW?). Most decent phones have an ethernet passthrough (2 port) so you can plug in your PC. As long as your LAN is decent (Cat5 100baseT switched) the overhead using VoIP is

Hello, I'm looking for a similar feature... Dial a number via ZAP/g1 after the line gets answered wait 10 seconds send DTMF Regards, Marc -- Network Manager Marc Storck LuxAdmin.Org mstorck@luxadmin.org Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352

Good Day list, Need assistance determining the best place to read up on whether Asterisk can help me out. I have a situation where I need to do the following <PRI from Telco> ------- <Analog Channel Bank>------------<Proprietary Box> | | | | | | <PRI Port 1 of Digium Quad T1> <PRI Port 2 of Digium Quad T1> | | | | | |

Hi, folks. I am planning on implementing Asterisk in 2006, and need to budget for it now, so I need to know what I'll need to get. My company has about 50 users, and is currently languishing on a very old Comdial PBX. All of our client computers are Macs; our servers are mostly OS X, with a couple Debians and a Red Hat. I am thoroughly experienced at systems administration, and can

Merry Christmas from the Colorado Organization for Victims' Assistance. Our (Comdial) PBX fried after a power failure. Thanks to Mark Spencer, Digium, VCCH, and the friends who support this group, we are now back "on the air". We wish everyone good health for the coming year.

I recently inherited a Vertical Xcelerator IP system with IP2007 phones. I would like to use the phones with an Asterisk system instead, but there doesn't seem to be much information on it on Google. Is it even possible? These phones claim that they are SIP phones. Thanks! Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a need to interface Asterisk with a VoiceWorks voicemail system. I was wondering what kind of card would be needed either a FXO or FXS interface? Any help would be appreciated. Thanks, -gcc

Hello, I have a big old Fax machine that will only pick up a ring at 24volts and 20mA(minimum) ring[according to the technical specs manual]. None of my SIP -> Analog phone adapters supply this: Cisco - 50 volts SIPURA - 70 volts Handytone - who knows, but it doesn't work can anyone tell me if there are any analog adapters or channel banks that can send a 24V 20mA ring signal? Also,

We have a TDM400P installed here with four FXS modules. It works well except for a couple of issues: First, I have a Panasonic KX-TG2431 telephone (so others can reach me when I am in o ther parts of the building) hooked up to one of the FXS ports. When the other end hangs up, I get the usual CPC disconnect signal. After the CPC, sometimes it will go to a dialtone, and other times a

Hi All, We recently had an old office building burn down. The office housed maybe 20-30 people. Only about 10 or so of those had their own extensions. We had a standard pbx from an area communications company, and I'm not quite sure about what kind of phone lines were there, I only know that their were actually 3 phone numbers, but everyone could get an outside line if they needed

Yo, Inspired by the example in the tips & tricks-section of "http://www.junghanns.net/asterisk/", I built a more elaborate call divert-feature. This one validates if the extension a call-forward is to be set to is actually valid for the current context and additionally saves this context into the DB and always uses it to originate the divert from, as you can't expect the