similar to: Small office setup with Asterisk @home, IAX and analog termination

I've got an Asterisk @home setup that mostly works. I'm using Polycom SIP phones and have PSTN termination through an IAX provider. I say "mostly works" because, if I'm on the phone, any additional incoming call goes straight to voicemail rather than signaling call waiting on the phone. I've tried dialing *70, but that doesn't seem to fix things. I've googled

I'm trying to setup a demo where my Asterisk box with a TDM01B (FXO) answers an incoming call and forwards that call to a SIP softphone (X-lite.) Seems all is built/installed okay: # ztcfg -vv Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. I'm pretty new at this and the extensions.conf file is eating my

I am having a problem with an Asterisk server, in that when it is receiving a call from another Asterisk server using an IAX2 trunk the phone rings for 10 ms and then there is a hungup from asterisk and then the phone rings again before another hangup. The funny thing is that after I really hang up on the calling phone it repeats this as if I am still trying to call. Any Ideas?

The TDM01B is 4 port capable but has only 1 FXO module. I'm running asterisk 1.2.7 and zaptel 1.2.5. I cannot seem to get the TDM01B working. When I do the zttool it shows 4/1/0. I can dial out from a POTS phone up to the point that the cable plugs into the card. Here is my /etc/zaptel.conf loadzone=us fxsks=1 and here is my /etc/Zapata.conf [channels] language=en #include

Folks, For the last few days I've been trying to experiment with a Panasonic PBX and an X100P but have run into quite a few problems which I am not sure if they can be solved with this type of card (how about TDM01B?) 1) I wanted to use *'s IVR capabilities, so I routed the calls to the extension where the x100p was connected to. Asterisk should answer the call, playback a message,

Guys, I need to use Asterisk to call out PSTN numbers via an analogue line. I understand Digium manufactures these kinds of cards, but can someone tell me which model number it is. I really only need a card with one or 2 analogue ports max. Walid -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi: I am practicing with the attached shapefile and was wondering if I can get some help. Haven't used 'rgdal' and 'maptools' much but it appears to be a great way bring map data into R. Please take a look at the comments and let me know if I need to explain better what I am trying to accomplish. library(rgdal) library(maptools) library(ggplot2) dsn="C:/Documents and

I'm attempting to use an asterisk box with a Digium TDM01B as voicemail for an existing Meridian/Norstar PBX with an ATA-2 adapter. We're having problems where hangup is not always (but sometimes) detected. It's not detected probably 70% of the time or more. (The users transfer callers to an extension--caller then has to navigate a menu to get to the appropriate user). After

Hello, I am trying the setup the TDM01B card. 1 FXO. I connected it to a regular home line. in the /etc/zaptel.conf, I have fxsls=4 In the /etc/asterisk/zapata.conf I have: signaling=fxs_ls language=en group=1 context=default channel => 4 When I start asterisk, I get this error: ERROR[10376]: chan_zap.c:6584 mkintf; Signaling requested on channel 4 is FXO Loopstart but line is in FXS

Hi Everyone. I have a client that uses an Altigen system. I am really new to PBX systems so all this is totally foreign to me. They currently have 5 inbound trunk lines and about 20 analog phones. >From what I can gather they are using the Altigen Quantum cards that support 8 extensions and 4 trunks. >From what I can gather the solution is a TDM04B and TDM01B to bring in the lines from

Hi, I'm trying to set up a basic FXO <> SIP gateway. That is, I want calls from my SIP phone to simply be dumped onto the POTS line. My (entire) extensions.conf is: [from-sip] exten => _9NXXXXXX,1,Dial(ZAP/1/${EXTEN}) and my zaptel.conf is: fxsks=1 loadzone=us defaultzone=us and my zapata.conf is: context=incoming signalling=fxs_ks echocancel=yes

I have a couple SIP phones on a PIII 1Ghz 256MB * server with a TDM01B connected to the PSTN. Calls between SIP phones are clear. Calls to the PSTN are quite noisy. The other person does not hear noise but I hear quite a bit. It is not an annoying sound but definitely much noisier than typical PSTN or even cell phone calls. I believe I have a TDM400P REV H card. I definitely don't

Dear list, We have an weird problem with our FXO card (TDM01B). When I made a call using this channel, all goes well, the called phone rings but when the called phone answers the call. In me handset I can hear an weird sound like a "Clack". I tryed diferents TDM cards and modules, and my zapata.conf is like, language=en context=from-zaptel switchtype=national usecallerid=yes

Hi Folks, Recent installation with a simple TDM11B (one FXO, one FXS) that I've set up (at home). I am receiving callerID fine from the telco, as it shows up in my call detail records, AND on 2 SIP phones. However, I'm not reliably receiving it (that is, very seldom does it come through) on the analog phones. Any ideas on where to check configurations, etc? I haven't encountered this

Hello, I have configured my TDM01B Card (1 FXO Port ) as follows (below) but it will not pick up an incoming call. Any suggestions/tools to see what the problem is? I have looked at zttool where this line changes but I don't understand what it means (The last digit changed from 0 to 1) Total/Conf/Act: 4/ 1/ 1 /etc/zaptel.conf fxsks=4 loadzone=nl defaultzone=nl

Hi i have buy a Digium TDM400P with 1 fxo modules TDM01B for connect my asterisk to a french analog line. In my zaptel.conf, i have: loadzone=fr defaultzone=fr fxols=3 when i load the module, i have in logs: Nov 24 06:13:40 gw kernel: Freed a Wildcard Nov 24 06:13:40 gw kernel: Zapata Telephony Interface Unloaded Nov 24 06:13:40 gw zaptel: Removing zaptel module: succeeded Nov 24

Hi all, I'm trying to figure out a way to create a data graphic that I haven't ever seen an example of before, but hopefully there's an R package out there for it. The idea is to essentially create a heatmap, but to allow each column and/or row to be a different width, rather than having uniform column and row height. This is sort of like a Marimekko chart in appearance, except that

I had to return my TDM11B because it put the PSTN line 'off hook' the moment I plugged it in and wouldn't hang it up. The new card seems to work because I can actually make an outgoing call from the FXO port to my cell phone, so I'm pretty happy about that. But Asterisk doesn't recognize incoming calls from the PSTN. If I dial my home phone from my cell phone asterisk

Hello, my config : aah 2.6 (asterisk 1.2.4) , centos 4.2, 1 TDM11B (1 Fxo / 1 fxs ), 1 phone, 1 softphone I'm in France When someone from PSTN calls and hangs up before the call is answered, internal extension keeps ringing until timeout occurs. PSTN line keeps busy. Hangup detection doesn't work. I've played with different paremeters (callprogress, busydetect, busycount,

Hi all, Your help would be greatly appreciated, I have been struggling for days with inst / config of TDM01B. I have installed TDM01B. Using Asterisk 1.2.9.1 on RH Ent 4.00 cat /proc/interrupts shows card wctdm on int 10. Green LED is on (module is in port 4). I have my /etc/zaptel.conf which includes fxsks=4 zttool show my card. ztcfg -vv shows my card. zap show status shows my card but I