similar to: TDM400P & Channel Group

I have an Asterisk@Home 1.3 server (Asterisk 1.0.9) and recently added a TDM400P with (1) FXO card on port 4. Inbound calls are always successful but outbound calls fail 75% of the time with intercept messages from my dial tone provider that include "we're sorry, your call did not go through", and "we're sorry, when placing a local call it is now necessary to dial an area

A customer of mine has 3 TDM400P cards in a box running asterisk. On each card he has four FXO modules. I have set up the dialplan to dial via group 1 for an outgoing call. Channels 1-12 are in group 1. If he plugs a telephone cable into socket 2 or 3 etc, but not 1, when he dials out, it still tries to make the call via socket 1. Straight away the console says that it has dialed the

>Asterisk and it works fine untill the following >situation: > >- one of the telco lines occasionally becomes mute after call is completed, would not provide dial tone, (not sure about ringing on that >line) - both via old and new PBX. >- zap show channel <n> would show that line as 'Offhook', though no telephone is off hook. > >If physical line would be

Hello, I have an issue with Digium TDM 400 card series. When I try to make outgoing call (PSTN call) for example, the Zap channel could not be created and busy channel message appeared. Below is the full log : [Feb 5 09:26:17] VERBOSE[3047] logger.c: -- Executing [s at macro- dialout-trunk:20] Dial("SIP/213-09648720", "ZAP/g1/08170709XXX|300|") in new stack [Feb

Hi, We have two of these 4 port FXO cards. However, we are having some problems with incoming/outgoing calls. The latest version of Asterisk/zaptel from CVS is being used. Voicemail, internal SIP <-> SIP calls between Pingtel xpressa hard phones work terrific, echotest is fine, and so on. The zaptel and wcfxs modules load fine, and show all 8 FXO interfaces in dmesg:

I am trying to place an analog outbound call from a Sipura SPA-841 through a * server with a TDM400P and 4 FXO's. When I call in from an analog line everything works fine, I can talk over the SIP phone. When I call out, * says: == Spawn extension (from-sip, [phonenumber], 1) exited non-zero on 'SIP/sipphone-d29d' -- Executing Dial("SIP/sipphone-9eb0",

....I THINK. When dialing 1+10 digits, I occasionally get a telco message "You must first dial a 1....". When I look at the console, the number is being sent to the ZAP channel properly. We're talking about a couple of POTS lines on a TDM400P. I'm thinking that it may be starting the dial too early after coming off-hook because I can just redial and have it work (or not)

Greetings. We are running * on RH9 using a Digium TDM400P four-port FXO card. We use only two ports on the card (ports 0 and 2 in this case). A consultant set this up for us, and it mostly works OK. However, outbound calls use our secondary number rather than our primary number first. This undesirable because of Caller ID; we'd like the primary number to appear instead. Yes, I can

Hi, We are having a problem with asterisk detecting that an analog ext has been put down. This seems only to happen after a number of calls have been made. We have an FXO port (TDM400P with FXO module) connected to our PBX and are using this to test asterisk prior to rolling our for our small office. What happens is that we make a number of calls to this ext which 1st rings a phone (FXS)

Run /usr/src/zaptel/ztmonitor 32 -v And adjust your gains in /etc/asterisk/zapata.conf accordingly. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of bam Sent: Friday, May 07, 2004 3:35 AM To: asterisk-users@lists.digium.com

I have made bri-stuff.0.0.2rc19 to work (I think) but I can not achieve any in-dialing nor I can dial out; this is what I have from "pri intense debug span 1" command ---------- *CLI> pri intense debug span 1 Enabled EXTENSIVE debugging on span 1 -- Executing Playback("SIP/201-a862", "tt-weasels") in new stack -- Playing 'tt-weasels' (language

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

Hello all, I just installed a TDM400P with 2 FXO modules on my asterisk server. The card works perfectly. To get users to ring out from my SIP phones i setup an extension with 0 that basically does something like this: extension => 0,1,Dial(ZAP/g1) where g1 is the group of the two FXO channels extension => 0,2,Hangup This works exactly as i want so users basically can dial 0, wait for

Hello all, I have a problem with overlap dialing and don't know how to get rid of it. My setup is: 1 HFC card with bristuff -> ZAP/g1 (2B + 1D channels), SIP phones (I just removed TDM400P with 4 FXS) I created test extension 222 which goes directly to g1. In Zapata.conf overlapdial is set to yes. First I created this extension: exten => 222,1,Dial(zap/g1,100,tc)

> -----Original Message----- > From: Jacqueline Lee [mailto:jlee@isdomaininc.com] > Sent: Friday, September 23, 2005 11:46 AM > To: asterisk-users@lists.digium.com > Subject: channel offhook state > > > We are using a digium card (TDM400) with asterisk for our access to the > PSTN. Initially when the server starts, all the zap channels on the card > are in the

Hi, I have two small companies that are going to be sharing a * box. I have 2 TDM400's with 4 fxo ports each. Each company has its own sales person and they would like the sales people to always show their own caller id and have their own lines ring directly to their phones. Company 1 sales person uses the 1 port on the tdm400 and company 2 sales person uses the 2nd port of the tdm400.

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

Hi All, I'm attempting to use two X100P cards and one TDM400P card. What driver do I load for the TDM400P? Is their any documentation for this card yet? Also, I know this has come up before but I can't seem to find it in the archieve... how do I configure the second X100P card? Anyone have a working sample? Thanks in advance, Randy

All, I've recently installed a TDM400P and am unable to get it working. It has four FXO modules on it. What I would like to do is have the first two ports as inbound from pstn and have them ring the last two ports which will be connected to regular analog phones. In Zaptel.conf I have: fxsks=1-4 In Zaptel.conf [channels] context=default switchtype=national rxwink=300 ;

Is there a way to busy out a Zap channel? I have a customer who is having problems with a line connected to a TDM800 card and we would like to busy out that line. Since that line is the head of the hunt group I cannot simply disable that channel, I need to busy the line so calls will come over the other lines. -- ?Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director