similar to: call waiting beep on PSTN and TDM400P FXO line hook flash

I had a similar issue with sending a flash to the PSTN for call waiting. I found that my dial plan in the extensions.conf file was not allowing me to dial *xx. Once I corrected my dial plan I was able to dial *0, *69, *78, *79, etc. Training the wife on how to actually use it was an entirely different issue. I have not tried to enable 3-way calling via the PSTN. "Jeff Otterson"

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 =>

>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 one FXO module and two FXS modules in it. I also have a Wildcard X101P. After trying hard to get things working on various Intel computers, but having echo problems that made it not really usable, I decided to try it on some older PowerPC (Macintosh) hardware running Yellow Dog Linux. Things started off smoothly. Both zaptel and asterisk seemed to compile okay, and both

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

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"

Hi , I have an asterix (1.2.7.1) box with 3 TDM400P and 10 FXO modules, and zaptel (1.2.5). The problem is that 6 modules can't be initaized and i get the folowing error: May 7 09:17:37 WARNING[7739] chan_zap.c: Unable to specify channel 9: No such device May 7 09:17:37 ERROR[7739] chan_zap.c: Unable to open channel 9: No such device here = 0, tmp->channel = 9, channel = 9 ztcfg output

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.

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

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

I have been having some troubles with the zaptel channel on what appears to be the inbound process. The box is running the stable CVS code and has a TDM400P 4-port FXO card in it for analog connectivity. Channel 1 is the only active port on the card at the moment as we only have one analog line. What has been happening is that it looks like Asterisk has been detecting an inbound call even though

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

My set up is quite simple: Redhat 9, Asterisk CV-HEAD from 07/11/04-13:04:22; TDM400P card, FXO card (X101P now unused) and a Fritz! AVM pci ISDN card. It works. But...just got an adsi PT390 phone and with some help from the nice people at Sayson got some basic programming into it. The programming came from dialling over a traditional phone line however and not from Asterisk. Whenever I

Hi I know this gets covered fairly regularly, but I've had a search through the archives and can't find anything dealing with this specifically - apologies if I've missed it though. I've got a TDM11B with the fxo port plugged into a standard UK BT PSTN line, loading wcfxs with OPERMODE=UK. All's working well, except if I get an incoming call through my bt line, and the

I just received a Wildcard TDM400P by FedEx yesterday. I noticed that the FXO/FXS modules use connectors similar to Ethernet. Now, i want to connect the TDM400P to the PSTN connector in the wall, and also to a regular analog phone. Both the PSTN conn and the phone use smaller connectors, typical for analog phones. I searched the "official" docs and the Wiki, there's good

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, 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

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: