similar to: asterisk GUI's that supports zap fxs extensions

by "GUI" do you mean a configuration utility or a User Interface? MATT--- -----Original Message----- From: Jon Gabrielson [mailto:jon@directfreight.com] Sent: Thursday, February 10, 2005 10:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] asterisk GUI's that supports zap fxs extensions Are there any gui's that support zap fxs

Hi all, With the ability of an easy install using Asterisk@Home, I have decided to give it a try. It is my understanding though, that one cannot add zap fxs ports as extensions using AMP. Is there anyone using Asterisk@Home and have added any extensions as zap fxs channels? Would be interested in how you accomplished this. Robert

We're having issues with the FXO/FXS ports on our Digium TDM cards sporadically. I'm wondering if anyone else has had these problems, or if anyone can provide guidance diagnosing or fixing the issue? The symptoms are that the FXO and FXS ports "stop working", usually after 2-4 weeks of server uptime. When this happens, sending a (SIP) call to an analog phone on an FXS port

It's a udev configuration. Read UDEV.txt, or, read the AMP wiki http://aussievoip.com for a step-by-step. --Rob -----Original Message----- From: asterisk-users-bounces@lists.digium.com on behalf of Jachin Rupe Sent: Wed 7/09/2005 9:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] /dev/zap* is not showing up (gentoo, portage,asterisk 1.0.8 hi

I've got a 400P, with a couple FXO's and and FXS. I've got an analog plugged into the FXS, and it gets dialtone fine. However, whenever I press any digits, I get the doo-dah, doo-dah "unhappy" sound. I've got a functioning FXS system at home, but I was trying to plug this into an AMP-created configuration, and having no luck. Can someone -- especially if they used AMP

Hi All; Can I do transfer for the call from zap/1 to zap/2 (both are fxs)? All what I need is to add the t argument for the Dail function? And how can I transfer to be in that senario: zap/1 dial a code to transfer for zap/2, once zap/2 answered, then he can talk with zap/1 (where the third party will not hear what they are talking), then if zap/1 hanged up the handset, the call will be between

Hi, I'm setting up * to extend Norstar locals to our off premises support personnel. It works quite well as a simple extension extender. When the Norstar connects to the extension that is tied to the zap fxs port, I automatically dial the voip phone number and the call is connected. I would like to use the Norstar hunt group feature, that will dial four phones and connect to the one that

I am trying to get remote extensions to work correctly with agents. I have ackcall=yes and have agents logged in to extension 101 using agentcallbacklogin with extension 101 defined as: exten => 101,1,Dial(Zap/3/18165551234,20,tTA(custom/presspoundtoanswer)) This setup works great on local and/or voip channels, but on zap channels, the zap channel answers immediately as soon as it goes off

The HP DL380 G5 (like many rack servers) has no AMP Mate-n-Lok connector available to attach to a card that needs more power than the PCI bus can provide, like the TDM400P when FXS modules are used. HP has confirmed that there is no part they sell to give you such a connector, and Digium says their business edition folks got it to work, but only by doing nasty warranty-voiding things to the

I have fixed this before, but I cannot for the life of me remember how I did it. I have a TDM400P with one fxo module and one fxs module. I setup Asterisk @Home and everything works fine, except when I try and call out. If I call out with a SIP phone it calls the zap extension and not the pstn line? If I take the zap extension offhook and call with the SIP phone it dials out the pstn line

Ok, got my last issue sorted, now another one. I can call out fine on this zap channel which is connected to a carrier access bank 1 channel bank, using asterisk 1.2 (aah2.0), I can call out, call other extensions & such.. but I cannot call into this zap extension, it always says the user is on the phone & asterisk -r shows "s-CHANUNAVAIL|1". In AMP does something other than the

I'm a bit confused on how to setup Zaptel.conf and Zapata.conf when there is a TDM400P and a TE410P installed after upgrade. The TDM400P has 2 FXS in position 1 & 2 and 1 FXO in the fourth position. I see boot, WCT4xxP loading first and WCFXS loading second. According to my understanding, given above, the TE410P should be configured first, then the TDM400P. However, I'm not sure

Folks, First off, this is messy, and I hope someone will be kind enough to help me clean this up (the part added to extensions_additional.conf). You've been warned! For those of your using AMP or A@H, there has been a lot of talk about how to route incoming calls to different places based on which trunk is ringing. The standard answer is that you can only do this by using DIDs,

Hi All, Channels 25-28 on a customers PBX are regular Zaptel FXO cards that are hooked into 4 incomming phone lines. They are all in a group to do automatic rollover for outgoing calls (if channel 25 is being used, dial on channel 26, etc.). Sometimes when a user is dialing a number, instead of bridging to one of the FXO cards it goes and rings to Zap/1-1. This doesnt occur all the time but some

I have a TMD400 card installed in a PC with one fxs (installed in slot 2) and two fxos (installed in slots 3 & 4).? fxos work fine but I am unable to get a dial tone for any devices connected to the fxs.? I have correctly connected the power supply to the card and I have even tried moving the card from slot 1 to 2 on the board. Below is from the console when I try to route a call from FXO on

hi there I'm trying to get asterisk going on gentoo 2005.1 I'm just getting my feet wet so I thought I would just stick with the stable portage packages. Right now that's asterisk 1.0.8 I emerge asterisk with the following make.conf file: CFLAGS="-O2 -mcpu=i686" CHOST="i386-pc-linux-gnu" CXXFLAGS="${CFLAGS}" USE="-gtk -gnome -qt -kde -dvd alsa

OK, Asterisk Newbie I've read TFOT and the Asterisk handbook and lurked, but my skills are a bit poor so perhaps someone could post a dialplan fragment to help me Brief details Asterisk@home 2.6 installed on a miniITX system Digium 400 card with 3 FXO modules 3FXS interfaces by Iaxy (1FXS) and Linksys PAP2 (Sipura 2002) (2FXS) I started with AMP to get going but have started

I'm having trouble getting outbound calls going with aah 1.3 and a tdm400p w/ 4 FXO. Incoming calls work fine, outbound I get this: -- Executing SetVar("SIP/231-af2b", "OUTNUM=6643955") in new stack -- Executing Cut("SIP/231-af2b", "custom=OUT_1|:|1") in new stack -- Executing GotoIf("SIP/231-af2b", "0?19") in new stack

I have the following dialplan (segment) that isn't working as I expected it to: exten => s,n,Dial(Zap/1&SIP/202&SIP/203,18) exten => s,n,Dial(Zap/1&SIP/201&SIP/202&SIP/203,42) The plan was to have SIP/201 added to the group of ringing phones after 3 or so rings. What ends up happening, though, is the Zap/1 phone STOPs ringing when the dialplan falls through to

I've just setup a new asterisk box (cvs HEAD) with a digium tdm411 and a couple computers with eyebeam. I have one small. I cannot call the eyebeam clients from the phone connected the fxs port. I can call the phone from the eyebeem clients. And, I get both the fxs phone and eyebeam clients to ring when a call comes in through the fxo port. I have been trying to get this straightened out