search for: adit600s

All: Any have sample config's for an ADIT600 w/ CMG card connected to an Asterisk box via MGCP? Thanks, Garrett Smith <mailto:gsmith@voipsupply.com> gsmith@voipsupply.com VoIPSupply.com -- a division of B2 Technologies, LLC (716) 250-3408 Direct (716) 630-1548 Fax (716) 903-9495 Cell AOL IM: B2sales VoIPSupply.com is North America's largest VAR

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

Just starting to do the research on this one....I've got a customer who is showing interest in replacing any older Panasonic unit providing service to 96 tip/ring lines from a single PRI. Does anyone have any recent experience with a decent (as in, plays nice with * and has a reasonable per-port cost) channel bank or similar? Mediatrix only goes up to 24 port, as far as I can tell, which

Has anyone ever heard of this so called Dynamic Impedance matching on the ADIT 600? I called their support and they've never heard of it. We are of course having echo problems are on the far end due to digital/analog conversion on the local end using a channel bank. We have purchased an ADIT 600 and yes the complaints are "far less" however we're still getting them. While I have

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

Hello All, Just a question, I have an adit600 and I am looking for a way to pull the incoming cid into asterisk. Does anyone know if this is just not possible via t1? Or is it only available on PRI? Thanks, Greg

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

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

Hi everyone, Does anyone know which channel banks work well with asterisk. Are there any that have T1/E1 connection? Eran Gal Xorcom -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040818/0d2dd98b/attachment.htm

As soon I do a reload I see contant ringing like this on the CLI: -- Zap/14-1 is ringing -- Zap/23-1 is ringing -- Zap/22-1 is ringing -- Zap/20-1 is ringing -- Zap/19-1 is ringing -- Zap/14-1 is ringing -- Zap/23-1 is ringing -- Zap/22-1 is ringing -- Zap/20-1 is ringing -- Zap/19-1 is ringing This goes on continuously and no phones are ringing. I am using a digium T1 card and ADIT 600. Does

hello, please 'scuse the slightly offtopic question, but i see a lot of posts about the adit600, used as a channel bank, but from what i understand it can be used as a PRI interface as well. If anyone who is using the adit600 to interface to 4 T1/E1's has feedback, i would appreciate it, specifically involving the asterisk interface, echo, and DTMF. thanks, yair

Hi, I have Asterisk 1.2 + Adit600 Channel bank(which gives analog output and also takes PSTN lines into the Asterisk system). Conversations recorded by the ASTERISK comes in two separate Files: xxxxxx.0-in (GSM Audio) for the Asterisk Extension Side of the conversation; xxxxxxx.0-out (GSM Audio) for the Caller's side of the conversation. I have Quick Time Player to playback the

Can someone recommend a good quality 24 or greater port channel bank? Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070506/e5896021/attachment.htm

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...

I'm trying to set up asterisk so any phone connected to channel 1-16 of my Adit600 channel bank can pick up a call coming in on channel 24. I do not wish to ring any of the 16 channels on an incoming call -- this is strictly so I can pick up the line if I see it ringing and wish to answer at work. I have channel 24 in call group 3, and channels 1-16 in pickup groups 1 and 3. However

Asterisk has been working fine for me for several weeks using MGCP to a Adit600 for intra office calling. I have recently loaded h323 and the following errors occurs when starting asterisk. [chan_h323.so]Aug 10 09:09:18 WARNING[7824]: loader.c:258 ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object file: No such file or directory Aug 10 09:09:18 WARNING[7824]:

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 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

The problem: T100P connected to an Adit600. Channel 1-16 are FXS, 17-24 FXO. I have Zap/24 in callgroup 3 and Zap/1-16 in pickupgroup 3. When a call comes in on Zap/24 I cannot pick it up with *8 from Zap/1-16. *CLI> show version Asterisk CVS-04/27/04-23:48:08 built by root@tuck on a i686 running Linux The zapata.conf and extensions.conf are located here:

I was fooling around with echo cans today. I found that I could reproduce this problem regularly t400p adit600 for fxs mec3 enabled Two speaker phones side by side within about 8" of each other. Put phone 1 offhook with speakerphone Dial phone 2 Phone 2 rings and is loud enough to be picked up by the mic in phone 1 By the second ring the echocan thinks it's figured things out, but the