similar to: Dial/Zap doesn't work

Hi, I've purchased two x100p clones, and when I try accessing a line from asterisk with something like this: exten => _1NXXNXXXXXX,1,Dial(Zap/2/{$EXTEN}) (is that only supposed to put you on channel 2 or actually dial the # for you?) but I first hear noise, then a dial tone, but as soon as I start dialing numbers I get feedback and noise, and the call doesn't go through. Any

Hi, I'm trying to dial out on a vonage line connected to a zap channel using stuff like: exten => 200,1,Dial(Zap/2/${EXTEN}) but it doesn't work - when i dial in the extension, i can see on a phone connected to the same line that it's gone active - but there's no dialtone. also tried adding a wait period before accessing the line and exten =>

Hi, in response to a previous posting regarding getting the x100p to work, I was told to run "zap show channels," but when i do i get "no such command 'zap'" There was a previous posting on this, but the guy never posted the solution. thanks, Imran

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

Dear List; Thanks alot for the help. But how can I let the second dial tone (after pressing the extension to select that FXO port) to be difference than normal dial tone? Regards Bilal Ghayad -------------------------- Correction, on FXO port not FXS, second, read his email first: "Also, how it will be possible to assign an dedicated line (connected to FXO) to an button on the Polycom IP

I am having a problem trying to do inband DTMF passthru via asterisk. My setup: PSTN gateway MAXTNT v11.0 SIP (T1 PRI/NT2) Asterisk HEAD or v1.0 makes no difference (I am using HEAD mostly) 12/10/04 and 01/17/05 (no difference) CAC ABII-T100P to/from analog lines to/from asterisk BTW, I have used a ABI and it works just like the ABII with asterisk. What I am seeing is: I make a call from a

Dear all, I connected a normal phone line to the FXO port but the call is not being processed. The following is the output to asterisk console when I dial 9150 "9 is the prefix I configured and 150 is a local service in to know the current time" *CLI> -- Executing Dial("SIP/4444-d365", "Zap/1/150") in new stack -- Called 1/150 -- Zap/1-1 answered

I have a problem that is probably so "doh" I will be embarrassed. However, I have spent all evening on this with no success: I have the following setup (asterisk cvshead as of today) 10 Channel EuroISDN<=>Asterisk<=>Meridian What I can do: Call from outside into the asterisk, dial an extension, and pass through to the meridian. WooHoo. What I can't do: Call from

Hello, I just setup the Asterisk to integrate with Panasonic legacy PBX. Config as followings, PSTN <-- PanasonicPBX--TDM400P(FXO)--AsteriskPC --> Internet * is for AA / Voicemail and VOIP in/out Currently the AA / Voicemail function for incoming PSTN calls are working well. My problem is for the incoming VOIP call. It can ring my internal extensions and talk without problem. But

I have more FXO ports on TDM400's than I have PSTN lines available for testing. When all the lines were used up (the FXO ports are all in zap group 2), and I did a Dial(Zap/G2/${phone},5) it thinks the Dial succeeded even though there is neither line voltage nor dial tone. Can at least the lack of voltage be detected? It would be good in case one of the phone wires fell out that it would

Hi, I'm trying to answer a call on one line and dial out a number on a zaptel x100p fxo, but all I get from the phone I'm dialing is silence after it is picked up, and on the line that's supposed to be dialed out itself, noise. Thanks, Imran

I am testing analog phone and fax machine plugged into Rhino Channel Bank which is connected to TE412P card. This site is in China. I am running RHEL 5, Asterisk 1.4.21.2, Zaptel 1.4.11 and libpri 1.4.4 I ran into a problem which is analog phone can hear dial tone and can make outgoing calls. Another phone (ether internal or external) can call the analog phone ***but the phone does not

I have a TDM400 running telco lines on ZAP2-4 My after hours config is supposed to receive the incoming call then divert it to my home phone by calling out one of the other zap channels available. console output as such... Starting simple switch on 'Zap/3-1' -- Executing Dial("Zap/3-1", "ZAP/G1/0823274210") in new stack Aug 20 19:06:44 NOTICE[646]:

>> I have more FXO ports on TDM400's than I have PSTN lines available for >> testing. When all the lines were used up (the FXO ports are all in >> zap group 2), and I did a Dial(Zap/G2/${phone},5) it thinks the Dial >> succeeded even though there is neither line voltage nor dial tone. >> Can at least the lack of voltage be detected? It would be good in

I have been struggling with my Asterisk setup for 3 days now and I think I have done well...apart from the small detail that I cannot dial out on my phone (PSTN) line. My setup is: Suse Linux 9.0 1 fxo card connected to a BT(UK) line 1 Cisco ATA186 sip v3.0 with two analogue phones attached to it Asterix CVS-HEAD-05/30/04-06:56:31 with the UK Userid patch applied. Asterisk loads without any

I have a working system with two fxo and two fxs channels. I recenlty got an IAX2 account I would like to use also. While I have gotten the IAX2 channel to "register", it remains non functional, as the incoming calls, go nowhere and the outgoing calls attempt to go out over the ZAP channel. I can see this, via the CLI, with debugs on. I strongly suspect this is a dial plan/config

Hello, It seems that i spent days and days trying to make Asterisk do automatic dial-outs and i am clueless as i have tried everything i could. I would like Asterisk to automatically dial-out a specific number and leave a goodbye message. However, the outbound call isn't triggered. Even after i reload asterisk and restart it many times. I am using a TDM400 card from Digium with 4 Fxo ports

First time installing Asterisk and I have a strange problem with my TDM400P(TDM22B) card. When I attempt to dial out the first digit seems to intermittently get dropped. For example, I will make a call to 18882467555 and the first time it will connect and then the next time it will tell me to dial a 1 before dialing a number outside my calling area. This happens for any of the calls that I try

Hi, I'm trying to set a rule to dial out through multiple Zap groups so that, say, g0 is the cheaper POTS lines group and must be used first. However, if g0 is busy or disconnected then try dialing out g1. My g0 group is made up of 4 analog lines connected to a 4-FXO card. I disconnected the RJ-11 wires from the FXO card to simulate a line disconnection. So theoretically all calls should