similar to: Re: call waiting beep on PSTN and TDM400P FXO linehook flash

I have been looking for the answer to this question for a while. Google-ing and reading the archives of Asterisk-Users has not enlightened me. It seems that this question has been asked many times, and many times it has gone unanswered. I have call waiting and three way calling on my PSTN line from Verizon (the local telco). This is connected to a FXO port on a TDM400P. I also have

I've setup * with TDM400P w/1 FXS, 3 FXO modules. I've one analog phone connected to TDM400P FXS module, 1 PSTN line to one of the FXO module(ZAP) , and 2 analog phones connected to Sipura 2000 (SIP). The calls between SIPs and zap phone (fxs) are OK. But 2 issues cannot be solved: 1. To dial to PSTN via zap phone, the setup in extensions.conf with the following exten =>

Is there anyway to turn off the call waiting beep in the grandstream and/or cisco ata186? I have a dial statement in my extensions.conf that rings 5 phones at the same time by combining them with the & in the dial statement. i.e.) exten => blah,blah,Dial(SIP/GS1&SIP/GS2&SIP/GS3&SIP/ata186a&SIP/ata186b,25,t) If one of those lines is being used, then the user gets a really

We are experiencing problems with FXO modules on TDM400P. From time to time they stop responding to incoming rings although they work fine if we use them to dial out. It's been verified at least in two different installations (using different mainboards) in two different locations. The only solution to the problem is to stop asterisk, then unload and reload kernel drivers. The problem appears

Hi all, A while back, there was a short thread on using the FXS interface from a Primus TalkBroadBand device (a DLink ATA) as a incoming line for the FXO interface on the TDM400P: Primus <--> DLink ATA FXS <--> TDM400P FXO <--> Asterisk In that thread, a couple of people suggested that this does not work reliabley, and the ATA FXS <--> TDM FXO link 'goes

Sorry in advance if this is slightly off topic. I'm selling a Digium TDM400P with 4 FXO ports for $300 plus $20 shipping costs. This is an authentic Digium TDM400P. It's brand-new and was only installed into a system once and then removed (we decided to go with T1 spans instead). Please email me off list if you're interested in purchasing it. Thanks! -- Kenneth Shaw ExpiTrans, Inc.

Not to sound like a broken record but...... Is there any word on the availability of the FXO cards for the TDM400P? I have an application that would benefit. If it has been dropped please let me know. Thanks Gene Kochanowsky

Since the irc channel wasn't any help, I will try posting my problem here. I have two TDM400Ps less than a week old in a PC. All of the FXS ports work fine, and all of the FXO ports worked fine up until thisafternoon. If I try to dial in, I get a busy signal, if I try to dial out, all I hear is a very scratchy, very crackly dialtone. If I swap the first FXO module with the second on the card,

Hi Guys, I recently got a TDM400P 4 FXO for use in the UK, this at the time seemed like a good idea as I had good results with an X100P clone. Installation went great and call clarity is excellent no echo like I had on the clone card. My problems start with detecting hanging up the line. If a person calls into the system and speaks to me on a SIP phone when I hang up the call clears

hi, all I have configure tdm400p analog fxo card. that's ok. but how to chek that is working properly or not. i chek with ztcfg -vvvv and zttool . that's ok. i want to dial from my fxo port to another extesion. zaptel.conf ------------------ fxsls=1,2,3,4 defaultzone=in loadzone=in zapata.conf ---------------- context=mycontext signalling=fxl_ls group=1 channel=1-4 thanks' in

I have a TDM400p with 3 FXS mods and 1 FXO mod. I have all set up with what seems to be correct settings (according to digium and asterisk wiki). As soon as I plug in my POTS line into FXO mod the line goes into offhook state (whether I have power to the card or not). Should this happen? When I try to call * box all I get is busy signal. I've installed stable version, cvs version, change

Dear all. I have a tdm400p with an FXO module in slot 4 and an FXS module in slot 1. I have not configured the FXS port in an attempt to keep things simple. The problem is that when I call the POTS number (assigned by phone company) asterisk is seeing the call but then not doing anything with it. The verbose output from asterisk is as follows:

Hy, I would want to "test" the FXO and FXS channels of a TDM400p with the asterisk. For example, when I want to Know if a sip peers is correctly register, I use ChanisAvail, but if a module FXS hasn't a phone connected, How could I Know it?. Note, that I don't want to Know if the channel ZAP is correctly configurated. This information is given for Chanisavail. Thanks for

Hellow everyone, here's an other newby question. I've got a * configured with the card in the subject line. At times Asterisk fails to notice a disconet from the incoming line going into one of the FXO ports. Consequently it just keeps the line off-hook for ever and that causes my provider to mark the line aut of order. Is there any way to "help" Asterisk notice the disconect?

Thanks for the suggestion. I have busydetect=yes in zapata.conf. You refer to "Martin's algorithm". Can you provide more details please? ----- Original Message ----- From: "Soner Tari" <list@kulustur.org> Subject: Re: [Asterisk-Users] FXO module in TDM400P (UK, BT) - Hangup detection failing To: "Asterisk Users Mailing List - Non-Commercial Discussion"

>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

I have a TDM400P with some FXO ports, and I wanted to connect the two POTS lines from my Pipeline-75 ISDN router into the FXO interfaces on my Asterisk server. Hooked it up, seemed fine, called in and it answered. The problem is when the call is hung up on, the FXO port never drops. So of course then the P75 just holds the line off hook and you get a busy. So it's good for the first

Hi there! I'm having a problem with TDM400P and Caller ID on Call Waiting. Normal Caller ID works quite well, but I can't get CIDCW to work (tested against Siemens phones). I hear the tone, but a message on the console appears telling that the phone doesn't support CIDCW; when it does (it's used every day against the telephone central, a Siemens EWSD). Any ideas what could be

I'm a complete Asterisk novice and have an installation based on the Asterisk@HomeCD. I've installed my TDM400P with 2 x FXO & 2 x FXS, but every time I try to dial out, I get a message "No circuits available". Can someone confirm the pinouts for connecting the FXO's to a UK BT Line - I have RJ11 connectors on the back of my TDM400P card, so ideally I'd like to

Any details yet? -Dan