similar to: Minimal modules.conf (e.g. with autoload=no)?

I'm considering that board as a mail and voip gateway for home use. In view of all those statements about how little resources asterisk needs, did anybody already try running asterisk on it? Thanks, Bruno.

After having been toying around with asterisk and various VoIP stuff for a couple of weeks now, I want to recommend a preferred protocol and softphone to friends and family for calling me up. As SIP and H323 are such a mess to set up in NATed environments, the only reasonable protocol option right now seems to be IAX. After looking at http://www.voip-info.org/wiki-Asterisk+IAX+clients and trying

Hi folks an issue I don't understand. I'm running * stable 1.0.3 on public internet, with following iax.conf / sip.conf entries: iax.conf [100] type=friend username=Foo context=default auth=md5,plaintext,rsa secret=secret host=dynamic callerid="Foo" <100> qualify=no sip.conf [10] type=friend username=Bar context=default callerid=Bar <10>

Talking * 1.0.12 here. Problem: I'd like to avoid retries with queue, i.e. if members choose to ignore a call they should not be bothered again. On the other hand, when a call times out according to the Queue(...) timeout, the call should proceed to voicemail. Setting retry in queue.conf to a high value unfortunately doesn't solve the problem. More specifically, the timeout t given to

Hi folks last weekend, I tried Windows Messenger first time and was stunned by the little latency it gives. Until now, I've been using softphones on Linux exclusively, like iaxcomm, linphone and sjphone, and they all give me about 1, at times even 2 secs delay. Whereas Messenger really seems to be in the millisec range. Of course, I'm now curious why there is that difference. Clearly,

Hello , I learn handbook-draft.but I think I don't understand asterisk. where I can find some learning book about asterisk? thank u. B.R. John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?24? 7:51 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5,

Bruno Hertz is believed to have said: >Hi Aldo > >don't know about Suse, but I have a working setup with asterisk 1-0 >stable, chan_capi 0.3.5 and fcpci-suse9.1-3.11-02 on Debian Sarge, >though not prepackaged but all self compiled. > >Looking at your log messages, chan_capi obviously is installed, but the >load of app_capiCD.so fails due to an undefined symbol

While setting up my first dial plan, I find that notions like remote ip, network, or incoming network interface seem to be totally lacking regarding calling parties, where * still seems to fully rely on the easily spoofable caller id. Especially, allowing only certain ips or networks to enter a specific context in the dial plan is apparently not possible, at least in the h323 world. Don't

Hi All, Is it correct to say that by design, asterisk wont make use of any cards hardware dsp capabilities ? I don't think that any of the hardware cards currently supported have any dsp capabilities, but I wanted to know if for example, in the future a driver was written for a card that did have dsp capabilities, would asterisk be able to make any use of it ? I am only just starting out

Assuming I'm using a VOIP provider of some sort, what kind of bandwidth requirements / line should I expect to have in place? I currently have 8 traditional voice lines, and a FAX line that doubles as my DSL source. Ballpark, what do I need to have in place to move everything to asterisk? Dunc

Hello , I'm a student in Shahed university in IRAN . Now I'm become a member of our uni IT-Center . We are going to have an organized network with a en-bloc user authentication and proxy server. In comparison between MS AD and Samba , i want to choose Samba server ( cause of my belief in Open Source ) . Our plan have a Forest(root : shahed.ac.ir) and trees for faculties (like :

Hi, I want to autoload ide-scsi.o on each boot. In debian there is a file /etc/modules in which I can iclude the names of the modules that are to be loaded at boot time As I remember in gentoo there was a similar /etc/modules.auto or something ... is there an appropriate place in a redhat distribution or I should write my own rc script? sasoon

I am having problems getting any form of call transfer working. I have reconfigured blind transfers to be #1 and assisted transfers to be *2 but these are not working. Looking at the wiki (http://www.voip-info.org/wiki-Asterisk+cmd+Transfer) it it does not mention IAX so I assume I have to use the native IAX transfer supported by Diax? I have tried using Diax but am getting a problem that after

(REPOST, sorry if you get this more than once.) Hello all, (Not sure if this is more appropriate for user or dev list) Does asterisk have any sort of "standards based" api that can enable an application to do call control on the switch ? For example, if I am developing a call center application using asterisk, I would like to be notified of inbound calls and then be able to route

Hi All, I am wondering if I will be breaking the GPL, if I write for example, a channel driver or make some modifications to the astrisk source code, to interface at RUN TIME, through sockets, with a proprietary system. Eg. 1. I write chan_xxx + modify asterisk source (make changes + new code publicly available) 2. chan_xxx supports hardware by XXX Corp. 3, XXX Corps interface is

I keep running into this problem where I don''t see my changes getting reflected in the browser. Take me a few minutes to realize that I just changed a module and Webrick is not going to reload that file. Any way to tell Webrick about these other files? Thanks, Dave -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear list, I tried to install the CAPI support RPM from the SUSE 9.2 disks. YaST tells me it is happy with the process. The Asterisk release I am using is the one that comes packaged in RPM format, also included in the distribution. Still starting asterisk with the usual asterisk -vvvc I see that something goes wrong. [app_capiCD.so]Dec 23 19:21:45 WARNING[1076850816]: loader.c:242

Hello All, I am new to *, and this is my first post on the user list. I have had success with making / receiving calls to a SIP hardware Phone and the Console Channel Driver. My SIP phone *requires* that I register with a SIP server. For this, I used the NIST sip presence server (a version that I downloaded almost a year ago). I have a problem with getting asterisk to register properly,

Dear list, In a mixed model I selected I find a non positive definite random effects variance-covariance matrix G, where some parameters are estimated close to zero, and related confidence intervals are incredibly large. Since simplification of the random portion is not an option, for both interest in the parameters and significant increase in the model fit, I would like to collect

I just installed a new TDM400P with one FXO interface in slot 4 (how it came from Digium). This box is running Debian with a 2.6.10-rc2-mm3 kernel. After the make linux26 and make install in /usr/local/src/zaptel, I can see contents in /dev/zap but any attemp to touch for example /dev/zap/ctl gets a no such device or address ... Any suggestions?