similar to: Asterisk on Mini-Box M300

Does anyone know of a hardware adapter that can take ISDN BRI frames (I.430) and encapsulate them in Ethernet (any form, but TDMoE would be really cool), in much the same way that the redfone does for PRI? (yes I have asked this before in looser terms, but it was a while ago :) Would anyone find such a device useful? Thanks James

Dear Forum Members, I just purchased two VoiSmart GSM cards. Tried to install one of them on my Fedora Core 5 system, The compilation was not smooth, as expected, but after a small fix, it went through. Then I put two SIM cards in the card's slots. Then I loaded the modules. Then I started the Asterisk. After all I configured the vgsm.conf file according to my settings, that is just changed

Hi all, I have a "Cologne Chip Designs GmbH ISDN network controller" and I want to terminate voip calls via this ISDN card. My question is: How I must to wire the ISDN equipment with my ISDN card? With normal cable or crossover? How I can to check if ISDN card is linked with ISDN equipment? In this moment I have 1:1 cable between ISDN's, the mISDN is installed and "misdn

hi does anyone have an idea how it could be possible to do email -> fax gatewaying with asterisk + app_txfax, but still keep track of who sent the fax? i've thought a little about smtp auth, but it doesn't look too easy to integrate smoothly with asterisk.... roy

Hello list, I'd like to share something u all , so that i could understand whats going on into my Asterisk box. i have a setup like this client(ip phone) -----ip network------- [Asterisk]----ip network -------[Service provider] i have configured A2biling in my Asterisk box. so when client call to my Asterisk A2billing's ivr respoce , my client authenticate there pin and call . all

Hi all, I hava SER with many clients (sipura SPA2100). One of these is an Asterisk which have others clients (sipuraSPA2100). I also have a Cisco GW which give me access to the PSTN. I make calls to all IP phones in my network, but I can't call PSTN numbers. After I dial, I hear 2 ringbacks but at the same time Asterisk says: Called pstn_number@SER_ip_address SIP/SER_ip_address-ec75 is

Hi, I tried to find a reference in terms of size but got back a bunch of tech documents and couldn't get the idea of wav49 format. wav49 format is supposed to be half the size of a normal wav right? so, how much disk space takes to save one minute of audio in wav49? I trying to do some capacity planning for a voicemail server. -- ------------------------------------------------------------

Hi List, I would like to know if there is a way to detect gateways which time out (because of network problems or hardware failure for instance) when you send traffic to them. So when you do: Dial(SIP/number@gateway) If a call couldn't get through because the gateway has timed out, i want to do something about it. The idea would be to suspend gateway which time out for 60 minutes,

Hello everyone. I've got a HFC ISDN card that I'm using with chan_misdn and it basically behaves like crap. Echo is waaay worst then echo I get TDM400 card, sound is "choppy" (there other side is allays complaining about sound interruptions) and to top it all it detects fake DTMF's all the time. Is this a chan_misdn problem or is it a card problem? I really need to get

Dear Asterisk Comunity, I'm thinking about developing a callback application based on the following scenario: 1. Customer Calls the outgoing number which is a PSTN line connected to my Zap channel 2. Asterisk captures the Caller ID and calls back the customer. 3. As soon as the customer picks up the phone, asterisk plays a promt to enter the Destination number. 4. Asterisk Connects the

Hi group, i'm brand new and i would like to ask about soekris hardware. I read along the web but i have some doubts that i think can be solved here. My question are the following: 1) does the Digium TDM400P fit in the soekris box with a 4801 SBC or a bigger box is needed? Any suggestions about where to pick up another box? 2) does the Digium TDM100P (already discontinued) fits fine in a

Hi, I'm trying to setup a system where incoming faxes are received using SpanDSP and then send on to another (remote) fax machine. The SpanDSP part is working excellently, however I dont seem to be able to get the forwarding part to work. Heres what I put into my extensions.conf: exten => s,4,Answer() exten => s,5,Set(FAXFILE=/tmp/fax-${UNIQUEID}.tif) exten =>

Hello Masters Here i going explain what Iam doing and where i need help .. Iam running Sip Express Router ,Asterisk, on same box (for testing) my Sip express router is working fine and i can accept global register requests with valid account and in front of Sip express router (SER) Iam using Mediaproxy-1.4.2 which is handler to rtp/rtcp streams between nated clients

Hey everyone, A few employees have noticed some problem here and there when trying to make outgoing phone calls. After it happens, they try again, and are able to call through. The dial plan for outbound calling looks like below. Which I know they are getting to the Congestion part (which explains the busy) but what I can't seem to figure out is the cause for why they are getting sent

The place I currently work at has a Panasonic Key system with 9 extensions, and no voicemail. It services 2 PSTN lines. I am hoping to use Asterisk to host voicemail (I would like to use the IVR also, but I don't even know if or how it would work). Do I need to use a PRI between the two, or is there a simple solution? I would like people to be able to answer the phone and

can someone overthere help? the server specs are as follows HP DL380G4 Dual Intel Xeon 3.2GHz processor with 4GB RAM, running fedora core 3 asterisk-1.2.5 ss7-0.8.3d. using sip as advised to receive calls from another gateway in US. using g729 in transcoding way. however, I noticed the call hit the 51 active calls which is 102channels, I run "top" to check the system resources usage

Just to close the thread. The problem was that I was using an old version of the code. If anyone has the same problem, you can download the code from here: http://www.thetechguide.com/howto/asterisk/bluetoothfiles.tar.gz Good luck, Danko ----- Original Message ----- From: "Danko Miocevic" <danko@santirso.com.ar> To: "Asterisk

So I took a chance with an X100P knock-off on eBay. I'm running Asterisk + FreePBX on a PowerMac G3 (beige desktop) using Slackintosh 10.2 and kernel 2.6.16.16. Everything has been fine up until now. I compile the 1.2.5 Zaptel drivers without a problem, get the udev configuration in, modprobe zaptel, and finally modprobe wcfxo. At this point, I get the message: ZT_CHANCONFIG failed on channel

I've seen the embedded posts. Is anyone running Asterisk on the MINI ITX? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956

Is there anyone who has successfully put together a "high-powered" mini-itx ZFS box that would be willing to post their system specs? I''m eyeballing the KI690-AM2... http://www.albatron.com.tw/english/product/mb/pro_detail.asp?rlink=Overview&no=239 ...but am having a difficult time locating it and a suitable case currently. I was in negotiations with a hardware source,