similar to: How to configure 2 asterisks

Does any one have any recommendations on an off-the-shelve PC for an Asterisk Server? This is for a proof of concept, so it needs to be inexpensive. I have tried 2 different PC's and had problems with the sound cards. I am thinking of PC's I can buy from local dealers like Best Buy, Office Depot. SO a cheap HP, Compaq or eMachine would work fine for me. --Steve

Hello - sorry for my bad english. I will make a list of all sound files on asterisk and i'll record then on professional studio. the french prompts from sineapps sounds bad... sorry for her... tell me if their is many peoples want it ! thank's. en francais: dites moi si ca vaut le coup que j'investissexr dans l'enregistrement des messages en francais. La voix sera la "voix

Hello, I am a newbie in Asterisk IP PBX but I am very impressed by its functionalities. I have read that It can work over IP network and across the PSTN. I am not very sure how it works over the PSTN.. In case if people have not yet the Internet or SDL access, I would like just to know if it is possible to interconnect two IP LANs using the traditionnal analog Network and Asterisk PBX? In such I

Skype is very succesfsfull and get more and more users, ... we can ignore them, accept them or do something,... My suggestion is that we try to do something, ... If we would peer to each other, than we get soon also a great amount of users together, and than our service becomes more valuable, ... Let's discuss advantages and disadvantages! bye Ronald -- Ronald Wiplinger (CEO of

I have been impressed with the quality and meagre bandwidth of the G729 codec from Digium. I am in a testing phase of our roll out, we are using 5 Asterisk PBXs in various countries to provide connectivity for our employees, owners and family. As we are testing, and our setup is somewhat complex due to the peculiarities of our connectivity, there has had to be a lot of changes to servers, cards to

I have a customer looking for an automated way to provide his customers information. He found some windows software called Active Call Center - but I believe that he did the 40 day trial and it crashed his windows machine to much. So he wants something that can do a similar task running on linux. The Users expierience should go something like this One of his customers calls in the IVR asks

Hello All * users. I have been looking for a way to allow GSM termination through Asterisk to occur. I am not a Telekom fundi and I have set up IAX2/IAX/SIP on asterisk with the ZAP channels via the Digium TDM 400P. I am unable to find any place that can tell me the cost of the VoiceBlue with a currency to I can calculate the cost of buying one. Alternativly - or just out of interist - I

Slightly OT, but I think this is of possible interest to many of you, I need to get a UPS for my asterisk box. They are rated in VA but I can't quite figure out how that converts to real life. I have a PIII-800 box with two X100P and one TDM400P plus graphics adapter, an IDE hard drive etc. Will a small 400VA box be enough for this? tia

Hello, I'm traying to link 2 * servers using SIP and the following errors was show: "SIP/AsteriskA:AsteriskA@10.0.0.121/100") in new stack Dec 13 22:46:57 WARNING[8767]: chan_sip.c:1398 create_addr: No such host: 10.0.0.121/100 Dec 13 22:46:57 NOTICE[8767]: app_dial.c:759 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy/congested at this time Dec 13

Hello list we are installing 2 new servers (to run asterisk) shortly, for a "stand alone" service. Ignoring our current naming convention, we'd like to name them something.. but we are not sure what. a consideration is that on the screens of the phones it shows extension@hostname (eg 3001@telephony) (all extensions are numeric) so the users will see it everyday i'm not

IMHO! I just see a skype channel as something good for asterisk. Skype has broad coverage. I can't imagine that skype wouldn't be interested in selling corporate accounts "skype trunk lines". Imagine having unlimited or X amount of continious calls coming in on SkypeIN and out on SkypeOUT from Asterisk. Internal Phones would all talk IAX or SIP to asterisk and use all PBX

Good morningI was wondering whether you could help me. I installed sipp on my Asterisk server but I don't really understand how does it fonction! Has someone ever tried it?If you can explain to me the principle, I would be extremely grateful.Thank you very much in advance. _________________________________________________________________ Lancez des recherches en toute s?curit? depuis

I am trying to test the call parking, but It doesn't fonction :(these are my config files.extensions.conf:include=>parkedcallsexten => 4000,1,Dial(SIP/4000,60,tT)exten => 4001,1,Dial(SIP/4001,60,tT)exten => 4002,1,Dial(SIP/4002,60,tT)In features.conf:[general]parkext => 700 parkpos => 701-720 context => parkedcalls [featuremap]blindxfer => # disconnect => *0

Hi List, Yesterday night after a power off due to a faulty UPS my asterisk doesn't want to start anymore. Here is what I get on the CLI: Asterisk Ready. *CLI> Disconnected from Asterisk server: Bad file descriptor. Executing last minute cleanups == Destroying musiconhold processes Asterisk uncleanly ending (0). I use 1.2.7 I think on a debian sarge and cdr_pgsql too. Any ideas?

I am using Aserisk as a SIP server to interconnect differents PBX in differents sites. I am now looking for a tool that can test the performance of this solution: I mean is there a tool that enables me to test the capacity of this SIP server in terms of simultaneous calls that could be treated, the comsuption of bandwidth.. or any thing like this? I am in urgent need to such a tool, If anyone

Hey guys. If my Asterisk box connects to the PSTN using SIP and IP over ethernet and doesn't require any authentication, what sort of a trunk would need to be created? Thanks, -- Nick e: nick.hoffman@altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality

RAD appear to have bucketloads of products which bridge between various interfaces (E1, BRI, POTS) and their own TDMoIP protocol. The attractive thing about them for me is their availability in Australia. The voip wiki says not much about it (http://www.voip-info.org/wiki/view/TDMoIP), and certainly nothing about if there is any way to get Asterisk to talk TDMoIP. Despite the name, TDMoIP

Hi List, Being very new to Xen I have a few generic questions for the list, I hope to grab some useful advice and pointers to documentation. I am evaluating Xen to consolidate a few existing servers into one appliance (mainly in order to reduce power consumption, heat, and hardware failure risks). I plan to have a SER router, an Asterisk LCR router, a voicemail server, a calling card server

Hi, I need to have some SIP extentions on remote places where the latency from my asterisk box with public ip is 1~1.5 seconds. What codec will work fine on this sceneary? I'm planning to use iLBC, is a good choice? Regards, Guillermo.

Hi List, I am working on least cost routing code on the moment, and I am stumbling on a problem. Say you have provider A having: Prefix XXX 0.10 Prefix XXXYYY 0.20 And provider B having Prefix XXX 0.15 You're stuck, because you cannot decide if provider B's "XXX" prefix also covers XXXYYY numbers or not. If it doesn't, it would be a waste