similar to: asterisk tries to dial out on lines already in use.

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

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

Just starting my search for a SIP/MGCP/H323 channel bank. I need analog ports in a building that only have network connections back to my * server. I could install another * server and use a normal CB with a PRI but I would like to investigate any good CB's with network trunk abilities Thanks Dave Packham

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

I'm going to install asterisk with four digium cards. Can anyone mention a brand that carries boards with 4 compatible pci slots? Thanks Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200

Hello, I don't have any experience with channel banks and would appreciate any feedback on my theory outlined below: We have a single T1 entering the building with channels 1-12 being voice lines and 13-24 being a 768k internet connection. This T1 terminates to an Adit 600 (T1-1). Here's what I know. Channels 11-12 go out the Adit 600's 25-pair connector to a wiring block (and

I've finally got my Adit 600 and are configuring it right now. But I have to say, there aren't much documentation for it. I've setup MGCP and Asterisk seems to find it. But all channels (40 FXS channels) are "Down"! But the MGCP itself is "Up" according to the statistics. I can't find any documents how to set each channel to "Up" in the CLI. Any

According to CarrierAccess, the Adit 600 uses CAS for voice signalling. What is this? This should not be a problem for Asterisk? Does the Asterisk server need to reencode CAS into aLaw when going to Euro ISDN? BR Daniel Nystr?m

I'm emailing this as the customer in this case, since my carrier appears to be completely unable to solve this. A brief rundown of the problem: - We have several voice lines going through the ADIT, of course into a VoIP type of arrangement. - Voice traffic will become choppy, even drop calls completely, at random but quite often. Modem calls (which we needed unfortunately) are a joke. -

Hello, I'm about to install asterisk as the PBX at a location that my company has just moved into and I would like to get some comments and advice on the installation. I am new to * and don't want to make any big mistakes so I would love to hear whatever anyone has to say. Here is what I have so far Server: * 2.8Ghz P4 - 1G ram * T400P Tormenta II (is this as good as the

To all Asterisk guru's... Here is my question. 1. Asterisk PBX - 5 Trunks / Incoming lines 2. 1 Building - 3 Companies (sharing phone system) Ok that's the basic layout. Here's the low down - Each company will have one dedicated channel for there company. The other 2 channels they want to be set aside as rollover's (rotary). That is not an issue. Where my concern is, how

Charles S. Antrim wrote: > I am using a card that has an fxo and fxs module. I am no where near an expert but I have my sip phone working through my pstn line and this is my config. /etc/asterisk/sip.conf [general] port = 5060 bindaddr = 192.168.69.1 context = sip disallow = gsm allow = alaw disallow = ulaw nat=disable srvlookup=no localnet=192.168.69.0/255.255.255.0 subscribecontext =

Got at suggestion from CarrierAccess to use the Adit 600 as an VoIP router using MGCP IP protocol, instead of controlling it through an E1. Have anyone tried this configuration? How does MGCP works? I've tried to search for it on Google, but I only find the protocol specification for it. Is Asterisk fully capable of this? I can't find any documentatin covering the use of MGCP in Asterisk.

We have 1000's of Remote Call Forward #'s across the USA / Canada, which forward into 1000's of 800 #'s in our call center. Is it possible to automate a solution where Asterisk could dial a given number, record the first 3 seconds of the call, save it to disk, and then go on to the next number, and just do this all day long ? We need to regularly check that the numbers work, for

Hi can someone help me, I want to do 'Dial(IAX2/bla/1234567|50|tT)'

It seems to me silly to have a T1/E1 card to connect to a channel bank when you could just have a 24/30 way FXS card in the slot in the first place. Does such a thing exist? Wouldn't Digium have a lot of customers if they could produce one for say < $1000 retail?

Hi, I am using a channel bank and zaptel hardware. I have a credit card machine on one of the channels that appears to be dialing "too soon" for asterisk, every complete number recognized by asterisk is missing the first 1-4 numbers. This is a serious problem for me, anyone have any ideas on whats going on? The pstn picks up on the dtmf tones just fine. I was able to get it to

Hi, i am recording a message, but it is always cut off at 40 secs. There are no time out configured. Gabriel -- The educated person is not the person who can answer the questions but the person who can question the answer.

I have set up a couple of test Asterisk servers and have never had a problem with sound. I've just done a fresh install on a dual 1GHZ PIII Asus box running Fedora Core3 with the Digium PCI Dev kit and following all the various Core 3 How-To's. I can make calls ok but when any sound is sent from the Asterisk box such as voice prompts and music on hold the sound is completely chopped up in

Hello All, I could use a recommendation if anyone has a moment. I have the T100P but I have not gotten my service yet. I want to have at least 12 lines of digital voice with DID. Should I just seek out a PRI ISDN provider or is there something else I should look for? I want to keep cost as low as possible. Also, I want to own my own router for the phones since it is always a hassle to get