similar to: Example of TDM20B

Dear all, I am happy to tell you that I received a Digium's TDM20B card for my Asterisk box today. but the phone jack is RJ45 instead of RJ11. So Please I need precise instructions to connect a phone to this card. please, assume that I have a phone (a normal analouge phone connected to the one end of a cable with an RJ11 jack (at the phone side). and now I want to connect the other end to the

Hi all, I am going to have a TDM20B FXS card and sit in it in my Asterisk box, I need out going calls only. I am quite a newbei of it, I hope your tips and tricks with valuable links. Thank you Kumara

I have a TDM400 with 2 FXS cards, and a Samsung 6100 DECT base station connected to one of the ports. The base station appears to only support UK (BT) style CLIP Caller ID signalling. Does anyone know if it is possible to configure the ZAP channel to send in this format, or alternatively can you recommend a GAP compatible base station which will handle both BT and Bellcore signalling? A search

See: http://lists.digium.com/pipermail/asterisk-users/2004-September/063348.h tml -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kumara Jayaweera Sent: Tuesday, June 14, 2005 1:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RJ45 instead RJ11 in Digium's TDM20B card help meplease

I have a TDM20B and asterisk compiled fine. The drivers have been loaded (wcfxs and run ztcfg -vv, I see the drivers loaded with lsmod). Asterisk starts up fine. I am using the default configuration files that are made when you do a "make samples". I was wondering if someone had a link or website that stepped someone through this kind of setup. What I want to do right now, is use a

Hi guys. I have a XP100P Clone , and the busydetect dont work for me.. PSTN---Asterisk---Sip---Asterisk----PBX Any call from pstn side dont disconnect ... I have no disconnect supervision and busydetect dont work... Please Help me. Zapata.conf [channels] echocancel=yes usecallerid=no hidecallerid=no rxgain=0.0 txgain=0.0 signalling=fxs_ks callprogress=no context=entrada channel=>1

No matter what I do, I've tried now three different motherboards, my TDM400P keeps giving 'Freshmaker failed register test' after emitting a list of xx != 00 errors. Anyone have any clues for me? I'm wondering if I got a bad module or what. TIA -- Undocumented Features quote of the moment... "It's not the one bullet with your name on it that you have to worry about;

>what is the wire combination/sequence >for a successful installation? http://www.google.ca/search?hl=en&c2coff=1&q=rj-11+rj-45+site%3Alists.digium .com&meta= hth

I have posted this to the Asterisk Forums, but got no response yet. Sorry if you are reading this for the second time. What fax hardware do I need for a T1? Ideally, I will switch my T1 to a digital PRI (not CAS I'm told, which is not as good) coming into the building. My CLEC said I can do this switch no problem. I have an analog T1 coming in now. >From the Asterisk box, I will connect

Hello. I am a spanish student, so excuse my English. I have this HW: - 2 X100P PCI with two analog lines plugged in. These lines are two extensions of a panasonic PBX. Zap/1 = X100P <-- analog line --> extension #237 PBX Panasonic Zap/2 = X100P <-- analog line --> extension #245 PBX Panasonic - 1 TDM20B with two analog telephones plugged in. Zap/3 = TDM20B

I have a box running *@Home 1.0 and I noticed today when I went to change some settings to do with the DID's that it is no longer detecting the different lines. I have a Digium 4 port line card and Im pretty sure that the DID's used to work when I used fxs_ks signaling on them. However I changed to fxs_ls signaling because the ks wasn't detecting when people were hanging up properly.

I tried adding wwww to my dial string... It appears to not made any differnece. I see by watching show channels that the w's are there in the dial. I have this box connected to bell south down in georgia. I have an identical box connected in indiana. The box in indiana works fine when dialing out. When in GA dialout is getting "you must first dial a 1 to place this call". show

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.

Im having a problem getting the 2 cards working together. I added the TDM with 3 FXO and 1 FXS board. The TE405p is working fine. THe TDM registers and identifies the FXO's and FXS but cant get them to go off hook. I hope someone can help. Asterisk seems to THINK its dialing it sends manager status as dialing, and it just sits there with no tones, on the phone dialing out. If I dial,

Hello, everyone, I am currently trying to get the asterisk server to talk with a 3COM NBX with T1 connection. My asterisk server has a T100p, TDM20B, a couple of sip phones. Now the sip phones are calling 3COM NBX phones fine, however, the analog phone has problem when dialing the NBX phones. The connection is established and the NBX auto-attendant picks up the call however the NBX end says that

I have a customer running Asterisk 1.2.13 with Zaptel 1.2.11 that is having calls dropped. Sometimes you can stay on the phone for a long time and sometimes the call is dropped within a minute. There are 9 lines connected to 3 TDM04B cards. The Panasonic Pbx we replaced did not have this problem. There are 8 SIP phones and 16 analog phones connected to two Astribank-8 units and everyone

Hi all, With the gamut of FXO cards out there, I'm looking for a recommendation for home use. I have a nicely working Asterisk 1.4 system that just requires an FXO card to connect my NTL PSTN to it. My previous X101P clone seems to have kicked the bucket. Any suggestions would be greatly appreciated. Regards Kyle -- Kyle Gordon kyle@lodge.glasgownet.com http://lodge.glasgownet.com

Hello all, Been a while since I've strolled this way. Apologies in advance if this is a common line of questioning. I've just bought a new Intel 865G based board with a P4 Hyperthreading processor. I believe I've gotten SMP set up correctly: in the menuconfig I specified SMP and told acpi to enumerate processors. Did I leave out anything? Anyway, the dmesg looks good and the

I just purchased a T100p from digium and a Carrier Access Access Bank 1 channel bank (12fxs/12fxo). I have the setup partially working thanks to some help from IRC. However I still have the following issues I can't seem to resolve 1. When calling into the system from the PSTN call hangup is not detected. * leaves line in use until it is shutdown. 2. When calling an analog phone connected to

Hi All, I've got an MD3200 modem which is working as a Zaptel FXO interface for Asterisk (X100P clone I believe). It seems to work, but on incoming or outgoing calls I can hear the other party ok but when I speak, I hear my voice echo back at me (quite quietly but it's distracting!) on everything I say. The other party doesn't seem to hear the echo - they just say I sound a bit quiet.