similar to: Suggestions for 96 tip/ring lines?

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Gary Franczyk > Sent: Thursday, January 15, 2004 10:37 AM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] Basic Asterisk capabilities question > > > Whaaa?? So, to allow 24+ lines of dial in access, how

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of David Burr > Sent: Sunday, January 11, 2004 4:31 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive > > > We have a new contest starting today! > > The first three

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Ray Burkholder > Sent: Monday, January 19, 2004 7:38 PM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] ADSI phone vs. IP phone (and > proper implementation thereof) > > [...] > I'm wondering if

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Asterisk > Sent: Saturday, January 31, 2004 10:05 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] PCI expansion slots. > > > > Hello, > > Did anyone use PCI expansion slots such as: > >

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Jeremy McNamara > Sent: Thursday, December 11, 2003 2:19 AM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Re: * with RADIUS > [...] > > Explain why you think you really need RADIUS Accounting? >

I'm a bit displeased at the way this happened. I received an email from VoicePulse. Here's some excerpts: ------------------ >We're sending you this important update so you can take advantage of improvements we've >been making to your VoicePulse Connect! service. >We've been working hard on improving the audio quality and reliability of your Connect! >service,

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Kevin > Sent: Wednesday, December 10, 2003 5:06 PM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] External Email Notification -2 > > > I was looking for an assistance to help with the > configuration of the

I'm looking for a way to give VoIP capabilities to an existing PBX: it's made by Mitel and it's used in a small/medium environment (a few dozen phones, but the PBX has capabilities for up to 200, if i remember correctly). Any high-level guidelines on how to integrate Asterisk with a PBX that's already in use? Probably that particular PBX is not supported directly, but are there

I am hoping to replace my Nortel 8x24 switch with Asterisk. Right now my cabling comes from my outside phone box into my office and into a punchdown block and leaves the punchdown block as an amphenol connector which currently plugs into the Nortel swicth. A second amphenol connector them plugs into the switch and extends to another punchdown block that I believe carries the lines throughout the

Say I get an ADIT 600 with two FXS8A and one FXO8A from ebay. a. Is it good for Asterisk? b. How do I connect the extensions and lines to it? Do I need a special jack? Can I get that jack in every corner? c. where can I find help for configuring it? d. what kind of backup does it have? Does it need to be reconfigured after a power outage? Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd.

Hi All I've done some reading on the wiki and read some of the mailing list archives, but can't see anything on this. I guess this means I'm either searching on the wrong thing, or have totally the wrong idea... Can anyone suggest if the following is possible? Currently, our office has a 24 analogue extension PBX, and 2 ISDN lines providing it with external connectivity. We have

We have a customer who is connected to our PSTN gateway using IAX and noticing that even when the traffic from their site is modest their outbound audio has short dropouts. Inbound audio is fine. (They have ADSL so it is expected that outbound audio would be the first to experience problems.) We have several questions to pose to the collective wisdom of this list. Q1: Are there any statistics

I know a few ppl have those CAC Adit 600's with t400 I can't seem to get my second span up on the T400 connected to the second spand on the adit (A:2) A:1 seems ok Can someone post they zaptel.conf span defintions And maybe a "print config" from the adit 600 cli I think my issue is timing srcs the coding, framing. bld out are all matched thx -------------- next part

I am having trouble getting my T100P and ADIT 600 talking. I have 2 FXO 8 port cards in the ADIT and a T1 controller card. I have a T1 x-over cable connecting two together. Here are copies of the relevant bits of my zaptel.conf and Zapata.conf Zaptel.conf span=1,0,0,esf,b8zs loadzone=us defaultzone=us Zapata.cfg context=default signalling=fxs_ks group=1 channel => 1-24 My

Assuming the price of an ADSI screen phone (say, Aastra 390) was the same as an IP screen phone (say, Cisco 7960) and someone was setting up an * server for their 20 employees (each of whom would have either an ADSI or IP phone on their desk), would there be advantages to using the ADSI phones over the IP phones, or vice-versa? For discussion, let's assume that the hardware needed to

Hello, I've recently purchased Adit 600 with 3FXS and 1FXO to be connected to my * server via T100P card. This is the output of "status equipment" command in the Adit600: For some reason the FXO card is seen as FXS, why? Is it ok? On the card it is written "FXO". Regards, Shlomi Bachar -------------- next part -------------- An HTML attachment was

....I THINK. When dialing 1+10 digits, I occasionally get a telco message "You must first dial a 1....". When I look at the console, the number is being sent to the ZAP channel properly. We're talking about a couple of POTS lines on a TDM400P. I'm thinking that it may be starting the dial too early after coming off-hook because I can just redial and have it work (or not)

Are they any issues still with hyperthreading processors, I've read and been told by a few people to make sure its disabled in bios if I want to use * on a hyperthreading machine. Kind Regards, Chris Bond

Hello, We use all SCSI PCI card hardware RAIDs on all 4 of our production Asterisk servers. They all have Digium quad T1 cards and they all have from 2 to 4 T1s hooked up to them. We have had no noticable problems with dropped calls/poor quality. What are you looking to do with this system? what kind of traffic will be going through these 4 T1s? MATT--- -----Original Message----- From: Deon

> -----Original Message----- > From: Travis Johnson [mailto:tlj@ida.net] > Sent: Monday, September 08, 2003 1:05 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Cisco 7940/7960 ethernet ports > [...] > We are having a problem with Cisco 7940 and 7960 phones when > the PC is plugged into the 2nd ethernet port on the phone. It > will drop the