similar to: Hardware requirements

Hi, all Here is a something I found on the web: http://www.voipbuster.com And it works OK too. Now, I want to use it via asterisk, so I ccan use my normal phones instead of PC application. Did anyone try to connect astersisk and VoipBuster? Thanks, Rudolf

Hi, all Where can I get "Asterisk Developer's PCI Kit" in Australia? It is a TDM400P with 1FXS and 1 FXO module. I amight need an extra FXS module as well. Thanks, Rudolf

Hi, all I am still battling to connect * and voipbuster. What protocol does it use? Ethereal capture shows UDP traffic, but no SIP or IAX traffic when using their client. VoipBuster client connects to connectionserver.voipbuster.com on port 11112 for authentication. Call itself is placed on different server. I have tried to connect using SIP and IAX and it seems that no authentication is

Hi, all Did anyone tried TDM400 card on old main board (Intel 440LX chipset -- PII)? The reason I am asking is because TDM400 needs PCI2.2 and main board is PCI2.1 I do not want to upgrade yet. Thanks, Rudolf

I have a number of Polycom phones to setup with my * server. For my initial few phones I hand wrote configs. Does anyone here who uses Polycom phones have some form of management utility for automating their setup? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc.

> Still trying to get NAT working. Try adding a canreinvite=no. Nabeel

Hi, all I am in Australia and I have to setup Asterisk in few offices. There will be IP phones in each office and I must be able to call between offices. I need actual handsets. I need "standard" handsets to be used by people. Those must support features like CID, call forward, etc. --- your normal office feature set. Also I need some sort of more complex handset to be used by

Hi I seem to have installed the G729 codec properly but can't seem to get it to work .. in the Asterisk startup I see .. [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) == Detected 1 licensed G.729 transcoders WARNING[1074408160]: File translate.c, Line 218 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator

I'm using happily this cheap phones, but I still have a little problem. Configuring the phone is extremely easy on * and I've a couple of them perfectly working, except when i try to call some toll-free number (in italy 800xxxxxxx ). If the number called is an IVR system, often with GrandStream (but also with Cisco 7905.h323) it's impossible to make the menu choices via the Dialpad.

Hi all, We are in the process of selection IP Phones to work with our *new* Asterisk PBX. We want to buy 4 for something less than 1000$ but with a nice set of features to work with our mail box, lines, good sound quality, full duplex (and maybe speaker phone). Any suggestions for something with good voice quality and not much troubles to setup with Asterisk? Voici quality is the most

I need to configure an * box to connect to a primary * box which is attached to a PRI in order to make calls using both the same E1 connection. I know the best solution is to use IAX (altough i could connect the IP phones on the remote site directly to the main * box that is VPNed with the remote one). I've some doubt on how to config IAX to work in this situation. I think that: On the

Greetings, I need if possibile an explanation on how to manage the incoming callerid for an incoming call. Let me explain the situation: We have two different companies in this office that shares the same PBX (* box). Each company have its own number for the incoming calls. What i'd like to implement is something that, depending on the incoming line that is involved in the call, plays a

Hi, all I am doing "prrof of concept" system. I will have two IP phones connected to Asterisk box. Box itself will have 1 PSTN conenction and one analog phone conenction. A basic minimal configuration. At the moment I am planning to use an old PII-350 with 128M of RAM I have lying around. I can not test anything yet, as I am waiting for phones to arrive, so question is will that be

Greetings. Today I've installed a fresh new E100P on a EuroISDN PRI. It seems to work well, accepting calls, but, when I start *, I have the screen flooded with this message: PRI got event: 8 (or 6) If i look into /var/log/asterisk/messages i've this Error: File chan_zap.c, Line 5341 (zt_pri_error): PRI: Read on 35 failed: Unknown error 500 Repeated indefinitely. Also, i've a

Hi, all Can a normal PCI modem be used to provide PSTN interface? I have seen modems that have answering machine capabilities, so there should not be a problem sending voice through them. Certainly, modem will be cheaper option then dedicated cards. Am I missing something? /*******************************************************/ Rudolf Ladyzhenskii Senior Design Engineer Open Networks Pty.

Hi, I've been playing with Voicemail and Voicemail2 a bit for my users, and there are a few things I'm wondering about: - We can specify parameters to the mailbox (s, b or u) to select which prompts to play. However, if we specify 'b' or 'u' it plays that (customisable) message, but it also plays the voicemail instructions. For the dutch, it is customary that a user

Hi, I came back from a quiet weekend today and found my E400P box to have gone astray. Asterisk is loaded from inittab, and started crashing and reloading a couple of thousands of times, each time notifying my monitoring service :-P I remember there would be issues on old cvs stuff since the crash at digium so I made a clean checkout just now. Here is what happens when I load manually:

Is there any way to run asterisk without a fxo card? I am looking only run SIP and a single fxs card.

On Mon, 13 Aug 2018, Blumenthal, Uri - 0553 - MITLL wrote: > Lack of time on the Open Source projects is understandable, and not uncommon. > > However, PKCS11 has been in the codebase practically forever - the ECC > patches that I saw did not alter the API or such. It is especially > non-invasive when digital signature is concerned. > > Considering how long those patches have

I've to project and build a fresh new box with * on. Basically, i'll have this situation: The office is connected to the phone carrier with a PRI. I need to let users continue to use their analog phones in a office, and an IP-phone based solution on the remote office, that will call using outside using the PRI on the first one. PRI---> Asterisk ---> Analog phones |