similar to: outgoing call (Sip phones to PSTN)

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 installed Trixbox and a TDM400P with 2 FXO and 2 FXS ports and am having an annoying issue with the FXO ports. As soon as I plug either one into the phone line it's as though the line is disconnected i.e. get disconnected tone when trying to dial out, line is busy when dialling in. The CLI shows the following: trixbox1*CLI> zap show channel 4 Channel: 4 File Descriptor: 18 Span: 11*

the caller is out his money anyway when you call any phone and voicemail kicks in, although i think on a payphone they give you a 2 or 3 second window to hang up. Suggest you implement i'm here / i'm away dialplan logic or set the do not disturb button that way when someone calls and the guy is away it hits voicemail right away and the caller can hear this and still have the 2 or 3

Hi, folks: Okay, so here's an idea. I have a TDM-400 card with an FXO card in it connected to the PSTN and a Polycom IP 501 phone. Observe the following simple dialplan for illustration: > [incoming] > ; incoming calls from the FXO port are directed to this context from zapata.conf > > exten => s,1,Answer() > exten => s,2,Dial(SIP/polycom) And zapata.conf: >

I got this nice TDM22B with two green modules left and two modules right /var/log/messages shows: Nov 26 23:47:48 dns kernel: Zapata Telephony Interface Registered on major 196 Nov 27 00:37:53 dns kernel: Freshmaker version: 71 Nov 27 00:37:56 dns kernel: Freshmaker passed register test Nov 27 00:37:56 dns kernel: Module 0: Installed -- AUTO FXS/DPO Nov 27 00:37:56 dns kernel: Module 1: Installed

I got this nice TDM22B with two green modules left and two modules right /var/log/messages shows: Nov 26 23:47:48 dns kernel: Zapata Telephony Interface Registered on major 196 Nov 27 00:37:53 dns kernel: Freshmaker version: 71 Nov 27 00:37:56 dns kernel: Freshmaker passed register test Nov 27 00:37:56 dns kernel: Module 0: Installed -- AUTO FXS/DPO Nov 27 00:37:56 dns kernel: Module 1:

Hey all, I'm having problems where there is significant static when making SIP -> PSTN calls. SIP -> SIP and SIP -> VM calls are totally clear and fine. Here's the setup: Polycom 601,501, and ten 301s. Digum 2400 TDM card w/echo cancelling, 12 FXO ports. The TDM card is on IRQ 5 with nothing else on it. Server Specs: Asus P4P800E Deluxe P4 3.0 Ghz 1 GB Ram 80 GB SATA HD -

Hello my friends... We are having some problems with the fax in our asterisk server... We have: Asterisk 1.4.21.2 Zaptel Version: 1.4.11 libpri version: 1.4.5 Digium Card TDM 410P This digium card has 3 FXO ports and 1 FXS port where we have a fax machine connected! The problem is that we can receive fax very good, but we can't make any outbound fax call, in fact, our asterisk get freezed

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

Sorry everyone, I know this has been hashed over a bunch of times but I can't find anything that pertains to specific cracking and popping on the FXO modules of a TDM04. This happens on inbound or outbound calls. This is the first install I have done with a TDM card for FXO modules so please, be kind if I am missing something really simple. Damn I wish everyone wanted t-1's or

I have put a box together using 2 X100P and 2 TDM400 4port cards. Using the simple setup that Martin posted a few days ago, I have asterisk almost up and running. /etc/zaptel.conf fxsks=1-2 fxoks=3-10 loadzone=nz defaultzone=nz (I have added NZ tone information to zaptel and Asterisk - I'll submit a patch soon). /etc/asterisk/zapata.conf [channels] context=incoming signalling=fxs_ks

I have a very strange problem I'm hoping someone has encountered already. I've scoured the internet for an answer to this one. My phone company provides no disconnect supervision. Hence I'm forced to use the busydetect feature. I have a TDM400 with two FXO ports. If I call from an internal extension to a landline and then hangup the landline Asterisk detects the busy signal

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 ;

I cant seem to get outgoing calls to be placed properly .. No matter what I try I get an error from the PSTN company saying that the "call can not be completed as dialed" or "you need to dial a one..." The asterisk debugging seems to show the correct number being dialed out of the zap interface... the "9" is being stripped and there is a "1" where it is

On Wed, 13 Dec 2006, Wireless wrote: > Hi Gordon > > I too have this problem with one of my two BT lines, very very annoying. I > was using a TDM400P and am now using a Sangoma A200 (lightning got the TDM), > I think the sangoma is very slightly better (less echo) but I might just be > kidding myself. Another think I've found it that using a faster CPU (now on > a 450

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

Yo! I changed callprogress to no, and in wcfxo.c source around line 334 i changed the value 32000 and -32000 to 10000 and -10000 because it had something to do with the DC voltage when it was ringing. I found reference here (http://www.voipuser.org/forum_topic_1791.html) with an interesting diagram of wiring that was incorrect for sending voltage to a phone or something like that. So put it

Hello, I have few questions about Asterisk. I installed Asterisk from CVS on FreeBSD and I made cvsup 2 days ago. 1.I couldn't find Asterisk version using "asterisk -V" command. How can I to find version information? 2.I am using Wildcard X101P (FXO) and Wildcard TDM400P REV E/F (FXS)on it. I tried Asterisk CallerID feature, but unable to get it. I tried callerid signaling V23,

Hello list , I?m putting to work a new asterisk box. I?m running * 1.0.3 with (one ) Wildcard TDM ( 2FXS* 2FXO) and (one) E-100P. Both boards are working well. ztcfg don?t show me any error. zttool list both cards as "OK" But when i run asterisk on verbose mode i get those errors : [chan_zap.so] => (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found --

Hi there, I'm experiencing an echo problem and dammed If I can sort it out. We're running Asterisk on Fedora Core 3 64bit, installed as per http://www.voip-info.org/wiki-Asterisk+Fedora+Core+3. These are the specs of the Machine ? 1 x AMD A64/3500+ CPU: Desktop Athlon64? Retail w/fan SKT 1 x Asus A8N-SLI Deluxe Athlon? 64 S939 NVIDIA nForce(r)4 SLI? PCI Express Req: 24pin ATX 1 x