similar to: Comdial PBX -- can use Asterisk as VM box?

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

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

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

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:

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

Thanks everyone for the input. After pricing everything we need out, it's not worth trying to get our old system to work, so I've pitched ditching everything and starting over. I'm very excited and hoping they'll go for it. Regardless, I'm going to throw a box together for my house, we have no home phone (just cell phones) so this'll be a great way of testing. All

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 there! I wanted to drop a line on the markdown syntax used for links: When i reference to [some website][1] and want to do [the same reference][1] later ai can only do this in endnote style (or howeer it's called). What i would want to be able to do is to make [an inline reference to a website][2](http://anotherexample.com) and be able to reuse [that reference][2] just the same. What do

Dear list: I am looking for a good way to create an S4 class that would extend numeric, but would allow NULL instead of data as well. As far as I can see there is no way at the moment to do that, but please correct me if I am wrong. The best solution I came up with so far was the following (it also indicates a problem of using setClassUnion with numeric as one of the classes): I define a class

Greetings - A quick perusal of some of the posts to this maillist suggest the level of the questions is probably beyond someone working at my level, but at the risk of looking foolish publicly (something I find I get increasingly comfortable with as I get older), here goes: My research group recently purchased a multi-Opteron system (bunch of 880 chips), running 64-bit RHEL 4 (which we have

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

Ken - take a look at using IAX protocol to route calls between your Asterisk boxes. Cory Andrews -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ken Williams Sent: Wednesday, November 01, 2006 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [SPAM HEADER] - RE: [asterisk-users]

>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

Hey, all. I quoted a customer about $100 for some cheap SIP phones. I was planning on using the BT-102's, but he called said they look like "Princess phones," and I have to admit that he has a point. Some of the other inexpensive phones look decent, but (for example) the SPA-841's wiki entry says the remote end gets a lot of static. Since it'll be being used from a noisy

https://bugzilla.netfilter.org/show_bug.cgi?id=1173 Bug ID: 1173 Summary: Documentation/error messages around NAT statements in "inet" tables incorrect Product: nftables Version: unspecified Hardware: All OS: All Status: NEW Severity: critical Priority: P5

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

Hi, I'd really appreciate it if you can explain this to me. I have a D/41JCT-LS Dialogic board and I want to use it as an IP-PBX. I'm new to IP Telephony and telephony and general and I researched a lot but still confused about what I really need. I know that I can setup an IP-Telephony for my LAN using a SIP server and SIP compatible software phones. But the challenge is how can I

Hi, all. I've got an old Telrad PBX with an Emagen(?) voicemail box. The VM box, itself, is beginning to show its age. Big-time. We're thinking it might be time to look for a replacement. I'd love to install Asterisk with an FXO card or something, but I don't think it supports whatever protocol legacy PBX's used to speak to VM systems. If someone can tell me I'm