similar to: Disconnect Supervision, SBC, and Adit 600

Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress enabled in * we are having a few disconnects while calls are in session (about 2 reported in first 5 days of use). I have talked both to

you can always do a "restart when convenient" within asterisk, and it will do it's thing when all lines are clear.... -----Original Message----- From: Jonathan Moore [mailto:moorejon@usd465.com] Sent: Thursday, January 08, 2004 12:31 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: 911 and lawsuits and redundancy Is there a way to reload a module from the

I have also noticed that sip.conf doesnt get updated without a restart..... was thinking I am doing something wrong, but maybe not now...... Chris -----Original Message----- From: Jonathan Moore [mailto:moorejon@usd465.com] Sent: Thursday, 8 January 2004 8:42 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: 911 and lawsuits and redundancy Another concern I have on this

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.

I just moved my system over to a new server with * 0.7.1. The old machine was using a cvs from August/Sep timeframe. On the new machine I did an make samples but then ovewrote with tar files of the production configs in the /etc/asterisk /var/spool/asterisk /var/lib/asterisk folders. Now the system seems to be working fine but only records blank audio in the voicemail files. Same thing with

I am trying to find a manual for the Carrier Access Adit 600. Does anyone know where I might be able to find one? Thanks -Jon -- Jon J. Brandon jon@monsoonretail.com http://www.monsoonretail.com

Hardly finished building our phone system for our school district and I have an opportunity to sell and install a system for a local small business. We are competing against a bid for an integrated voicemail/switch that runs about $1300 (without phones and cabling) and will work with analog phones. Is there hardware configuration (either using analog or IP phones) that would meet these needs and

Hello All, I have a customer site that is using * for ACD. In comming calls are eventually routed to a support rep via a queue. For new accounts the agent needs to be able to send a flash to the PSTN trunk (a POTS line with 3-way calling enabled), dial the number of an authentication center and then connect all three parties together. The trick is that both the agent and the customer need to be

I saw on the wiki that asterisk supports a *0 dial code to flash the external trunk. When I try to use this on my system using a t100p card connected to a channel bank that is agregating 6 POTS lines the code doesn't seem to do anything. Do I need to set a config value somewhere to enable this code? Is anyone using this feature successfully? -- Jonathan Moore Director of Technology Winfield

I just purchased 30 of these after testing one for a few months and would like to quickly purchase another 40. We really like these phones: good sound quality, good echo control (no echo in speaker phone), power over ethernet support, 10/100 switch, 8 programmable keys. Unfortunately we missed on big problem with Call waiting in our testing. When using asterisk 0.9.1 or rc2 the phone will reboot

Well, to do an upgrade on a traditional system you have the same issues, perhaps even worse as everything is physically wired to one system. To develop for production you must have a dev environment, a beta test and a scheduled release right? Todd Jonathan Moore <moorejon@usd465.com> wrote: __________ >These are good issues, but I am even thinking of something simpler and more

I am wondering what is the best way to send the smallest files with the vm to emai l integration? I am not sure what order the three lines of the format command take, so I have just tried trial and error swapping. I think when set to "gsm" I get the smallest sizes. I can get my Windows Media player to play at least part of the file (get missing codec message from Realplayer), but get a

I just purchased a T100p from digium and a Carrier Access Access Bank 1 channel bank (12fxs/12fxo). I have the setup partially working thanks to some help from IRC. However I still have the following issues I can't seem to resolve 1. When calling into the system from the PSTN call hangup is not detected. * leaves line in use until it is shutdown. 2. When calling an analog phone connected to

Hello, I have an Adit 600 (3 FXS cards) hooked up to a digium T1 card in my asterisk box. I 'connected' the slots to the a:1 T1 interfaces via the command line. The slots (3 fxs) are configured with 'ls' signaling. I configured the T1 card with the same line settings as the T1 interfaces on the adit and I get green lights on both the T1 card and the T1 interface on the adit (so

I have an adit 600 with an fxo card connected to a digium T1 card. If I try to make an outgoing call and the T1 cable is disconnected, asterisks returns congested like it should. But, if the adit 600 is connected to the T1 card, the adit 600 immediately "answers" the call even if there are no physical lines attached. I even removed the fxo card and the adit 600 still

Hi, Anybody out there running Adit600s? I have in my care an Adit600 channel bank connected to an old (version 1.0.6) Asterisk instance with MGCP. When trying a more recent Asterisk (1.4.21.2~dfsg-3+lenny1, Stock current Debian) calls fail. I have attempted to add the "slowsequence = yes" line to mgcp.conf. (It seemed to be the only likely candidate in the example files I found

Ok, I've pretty much decided to try the Adit route. Somebody who has experience with these tell me if I'm missing something. I have 15 incoming PSTN lines. T1 is not an option at current location. I want to put in an Adit 600 with 2 8-port FXO boards. The adit will then connect to * via a digium t1 board. I configure zaptel.conf for the T1. What other parts would be needed? How do

Hi, I'm using an Adit 600 Channel Bank with *. I love it and it works really great for my FXS lines. One problem that I have with it (It's really not a problem yet, but it's a potential one) is that I've scoured the manaual for the Adit to see if there's a way to dump out a config file from the bank so in the event of a power and battery failure I don't have to type in the

I've been reading the Google searches trying to understand how to tie together Adit 600 to Asterisk to provide 2 way service. I'm about blind from reading. I assume, the answer is using MGCP between the boxes. However, the examples I found don't really explain fully enough to know how to modify examples to work for me. I'll have in the ADIT with T1's. There is a CMG and

Has anyone had any success using the Adit 600 with the CMG card talking MGCP to asterisk? I want to have a central asterisk server with 10 Adit 600's at various locations providing 24 FXS ports.... Thanks, Isaac