search for: comdial

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% sur...

...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 buy all new there'd be huge savings there. I can't find anywhere that anyone has written any type of interface for proprietary (no reverse hacks or anything anywhere from what I can find), so I figure this is a no. Now...

...ikely 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 box? To: asterisk-users@lists.digium.com Message-ID: <760F32DC-1C7E-426D-B84F-9532F582294D@eyeontech.com> Content-Type: text/plain; charset=US-ASCII; format=flowed On Dec 7, 2004, at 8:48 AM, asterisk-users-request@lists.digium.com wrote:...

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....

...idged 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 calls (when it comes to faxing) must be having timing issues since they intermittently fail. I found what seems to be an issue in zaptel.conf (timing source for the Comdial side was 2 - changed to 0), but I don't know if that's it. I've also turned o...

...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...

...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

...y - support: friendly and helpful ------------------------------------------------------- ----- Originalnachricht ----- Betreff: [Asterisk-Users] DTMF after answer Von: john@simlab.net An: asterisk-users@lists.digium.com Datum: 05-08-2004 19:57 > Hi, > I am trying to link up a comdial PBX to Asterisk using T1 tieline E&M. I > have it working for comdial to asterisk but not the other way. Comdial does > not listen for any DTMF before answering the ZAP channel and requires codes > before allowing asterisk to call an outside line or inside extension. > > Does bn...

...xt 103 A standard T1 or PRI service are both digital and require to be converted to Analog to work onto the Proprietary System, as the need to receive analog with the DNIS and ANI information is important a product must be used which will allow for this. <YUCK> The Lucent G3, Intertel and Comdial switches are capable.</YUCK> I want to put Asterisk in here All PBX switches must have the capability of passing the DNIS (DID) and ANI information onto an FSX analog port, please check your current switch or talk to your provider. My question is : "Am I correct in thinking that I...

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 figure out most everything I need on the computer hardware and software side, but I am a complete telecom newbie and get...

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,

...whine of sorts. The same noise can be heard by hooking up a buttset to one of the ports and putting it in monitor mode (where it monitors the line, but doesn't go offhook.) Is it normal for Digium cards to generate this noise, or does this indicate something is wrong with my card? (Our Comdial PBX at church does a similar type of thing when flashing on an analog extension, but that still doesn't tell me if it's normal or not.) Anyone else see this same type of thing? Fourth: I get this message in the log (related to our analog FXS ports) Aug 11 16:46:17 WARNING[23855]: zt hoo...

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