similar to: ZAP CHANNEL QUESTION.

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

A little off-topic maybe, but it's still for the Adit used with Asterisk. ;) I wonder where I can buy 50 pin Amphenol cables, with connector on one side, and open cables on the other for mounting in our own patch panels. In Europe, or Sweden preferably. It's said to be very common on telcos, but I can't find it for sale anywhere! Thanks!

Hi All, I'm trying to hook up asterisk (CVS-HEAD-02/09/05-13:44:11 ) to a ADIT 600 via MGCP. Got it working really nice but now have a pretty bad problem: 1. When I perform a flash on the telephone, I usually get a second dialtone, but when I dial, dialtone doesn't break. If I flash back and forth a few times, it will eventually give me no dialtone.. here if I dial, it successfully

After months of setting up Asterisk. I completed the final testing last night We would go live on Monday. Or so I thought. As I moved the Adit 600 back out of the way, sliding it six inches. I noticed the major light was red as were both T1 and T2 lights ( eventhough only one t1 port is being used ) -- I have no idea what these lights mean or if they were lit before when things were working.

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 have set the following tbf tc qdisc add dev eth0 root tbf rate 0.5mbit \ burst 5kb latency 70ms peakrate 1mbit \ minburst 1540 I want to add a filter so the ip traffic pass from it.. plz help me __________________________________ Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html

Hi Friends, I have a problem with my Wine Installation in FC2-Athlon. I was following the PDF File - Configuring Wine Chapter 5 and it says that the "config" file is in ~/.wine/config. I can find the "config" file in the /etc/wine/config. Can you enlighten me on this? Actually, I can run already the "wine winmine.exe" and "wine notepad.exe" Also, I was

I have the following issue: 1) Call comes in from PSTN to Asterisk (IP A) and Asterisk forwards call to a SIP Proxy (IP B) 2) SIP Proxy (SER) forwards the call to a registered user. User does not answer and Call Forwarding is turned on for the user and the number to forward the call is a PSTN number. 3) After a specific timeout, SER has to forward the call to the "forwarded" PSTN

Probably doesn't help diagnose the problem.... but there were also audio problems experienced with this cvs version even on LAN / sip2sip / no transcoding > > ERROR[1171] UTILS.C:509 TVFIX: WARNING NEGATIVE TIMESTAMP -194931. ... > > I will be looking into this issue later today. __________________________________ Yahoo! Mail Stay connected, organized, and protected. Take

Hi: I need help setting-up multiple calling cards with different prices for the same routes using astcc. All my calling cards' routes now have the same price, but I need to be able to set multiple calling cards with different prices for the same route. I appreciate your feedback of How I can do that. Thanks; Chawki __________________________________ Yahoo! Mail Stay

Hi all, i am running a application which uses ODBC DSN. this is file and connecting with MS sql server. but when i run my program as $ wine c:\\crm\\PMonitor.exe i get fixme:vxd:VXD_Open Unknown/unsupported VxD L"secprov.vxd". Try setting Windows version to 'nt40' or 'win31'. But setting to this versions doesnot help as it provides dsn error like nddeapi.dll could not

--- Dan Pritts <danno@umich.edu> wrote: > Hi all - > > I've just finished building flac in the "darwinports" environment > on MacOS 10.3.9. > > The port maintainer (i've cc'd him) had disabled the xmms plugin > build. > > I wanted that, so I changed the portfile and built locally, yada > yada. > > I've run into three problems,

> All TCP connections have a port number, the default > for http is 80. Thanks. Good point. I was wondering how to handle this in icecast.xml. > As for the source of the relay, it can be shoutcast > or icecast, because > the relay is classed as a listener by the source. What if the source is neither icecast or shoutcast - but another mp3 stream source? If it matters, why?

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

Hiji wrote: >--- "David F. Colwell" <dfcolwell@dfcolwell.com> >wrote: > > >>Hiji et al, >> >>Couldn't find Half-Life or Counter-Strike in the DB >>yet they returned... >> >>Submitted version rejected >> >> >> >------------------------------------------------------- > > >>The version you

Hello all, I have a new Carrier Access Adit 600 and am having a problem dialing out and receiving incoming calls. After the initial ring, the calling party will here nothing. There is a silence, but the SIP phones that is supposed to ring will ring. The calling party will not hear the sound of the ring that * sends out on the line. Additionally, if someone picks up the receiver there is no

Another question, Is anybody using the Tellabs 2572 EC with a PRI. When we moved from our analog lines to a PRI, I thought it would be simple stuff moving the EC to the PRI. Changed signaling, made sure that channel 24 wasn't being ECd and everything came up. But, I was getting complaints of random echo on the PRI. Local echo. Also, we weren?t able to do any kind of modem dial-outs (Adit

Here's our config: cisco 7960's running 6.3 sip code latest cvs of * t100p zaptel card adit 600 channel bank 7 pots lines and 2 fax machines on the adit 600 dialing out from the cisco phones gets sent out via the zap channels, but I'm having some serious echo problems. I currently have the adit set to +3 rxgain and -6 txgain, with my zapata.conf containing: echocancel=128

Is there a way to reduce the volume of the dial tone on a zap channel? I don't want to reduce the audio volume on calls so txgain in zapata.conf will not work. I am having problems with asterisk not recognizing the first dialed digit from an analog phone about 8-15% of the time. Once the dialtone goes away, the digits are always recognized. Any other thoughts on how to solve this are also

Wondered if anybody might have some ideas about what could be causing this.... I have a T100p hooked up to an Adit 600 with 12 channels of voice off of a T-1 coming in. I have a t100p connected to a zhone z-plex with 24 fxs going to my stations. Some of the station are 2 line phones. These have 2 zap channels that are dialed when the extension is matched. Also, some extension require ringing 2