similar to: How much to charge for Asterisk installations?

Hi, Is it possible to use asterisk with Free World Dialup (FWD) ? Did someone manage to make it work? how? Best, -P -- __________________________________________________________ Sign-up for your own personalized E-mail at Mail.com http://www.mail.com/?sr=signup CareerBuilder.com has over 400,000 jobs. Be smarter about your job search http://corp.mail.com/careers

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, I am using a TDM40B, and have managed to compile mpg123 and turned on MOH. Problem I am having is that it is choppy, staticy, and sounds like white noise pretty much. I have search the archives to see if this problem had been resolved, but I haven't found anything yet. Has anyone had this problem and resolved it? I am calling from

Hello all, I would like to document some things I am doing with asterisk, but would prefer to do this from a web interface. I am unfamiliar with any software that allows you to create online documentation from a web interface. Ideally I will be able to create documentation online from a browser, which then when saved, is immediately ready to be read online. Perhaps I can setup different authors

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 OK, so I have a local extension that a phone can call to take it to voicemail. I don't want it to exit out to a fast busy tone, as I would rather it allow the user to simply continue on and call a new number (without having to physically release the line first). The [intern] context is where everything goes by default (sip.conf for example has

hi all, I don't know how to setup asterix to work as PBX. If I want just basic configuration with 2 SIP phones (snom, ata), what all I have to write in the configuration files, or respectively in the configuration of ata and snom ? If there is any good documention available, send me URL too. All (ata, snom) are behind firewall (nat) and astrix is on the public IP, but I can move for

I had posted earlier asking about a Snom200 communicating with a C7960 and lots of noise in one direction. Turned out the problem was created by me removing the allow=all statement in sip.conf. Someone had suggested that statement is no longer needed, and using allow=ulaw, etc, had an issue where one or more deny's had to be used as well. By adding allow=ulaw in the sip.conf file, the Snom

Hi all. I have some questions: 1) Is there a way to get a full log of the calls (incoming and outgoing) 2) How is the intregation of Mysql and Asterisk. At witch Aplicattions. 3) And of the Extension a) I have a Support Call Center. Almost all the time all the extensions are busy, and some calls at hold. Is there a way that when some

Hi All, I have just compiled the newest version of mpg123 on a RedHat 9.0 system (mpg321 has not been installed) and I am using the newest CVS version of asterisk. Whenever I place any mp3 files in the /var/lib/asterisk/mohmp3/ directory, Asterisk crashes a horrible fiery death. If mp3s exist in that directory, then I can't even start Asterisk. If I start it without files then copy

Does anyone on this list know of any Linux based apps that will work with Dialogic or Brooktrout that provides voice-mail/Autoattendant only?? It seems that Panasonic, Avaya, and Mitel all use Unix/linux based OS on their firmware for their proprietary voice mails. My wish list would be; A software that provides all of the drivers for a dialogic or brooktrout board Voice Mail Messages in WAV

Hi, I'm just wondering how other people cost/charge for * boxes they have installed in small businesses? Asterisk is such a complex beast and is capable of so much it seems equally complex to figure out a charging model!? I'm contemplating installing a pair of servers at a small company with two separate locations, one main aim is to provide a IAX link between the two but I'm sure

I'm hoping someone can help me out with this. I am basically just trying to separate my outbound and inbound calls into separate contexts instead of having everything in a single context. Any help would be appreciated. Perhaps I've missed something really obvious.... Here is the network layout: <remote> <--TDM400P--> <nattedbox> <--IAX--> <liveipbox>

Hey all, I am in the middle of creating a new user wizard which will generate all the .conf's the new Asterisk user will require to get themselves up and running in Asterisk without having to touch a single configuration file. This is what I have come up with as a rough draft. It is far from complete, so I'm asking people to submit things that should be added, changed, removed

Thanks to ww and his patch on bug #104, I have successfully implemented Asterisk behind NAT without using STUN or anything crazy. It's quite straight forward. Until this gets tested enough and put into CVS, you will have to patch your chan_sip.c file to do this. I'm sure within the next few days this will get put merged into CVS if no one finds any problems. I tried this on chan_sip.c

G'day, When I connect to extension 600 to do an echo test, I can see my microphone sending, but the echo is simply digital garbage. I know it's not my Messenger client or anything like that, as I use it regularily with Messenger. I am using a linux box with RH9 and using a soundcard. I have a pretty stock install, so if someone know where to look for either alsa settings or anything I

Anyone in the Toronto area interested in getting together to share notes and swap war stories? -- Jim Van Meggelen jim@vanmeggelen.ca -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.9 - Release Date: 06/01/2005

I'm just curious if I was to place my * box behind a a FW/NAT box running linux, if my SIP calls will still work. Box right now is a RH9 computer using iptables as the FW. I wouldn't mind placing my * box behind it, but I'm wondering if anyone has actually gotten NAT working with *? Thanks, -- +------------------------------------------+ |Leif Madsen -

I'm sorry this is somewhat offtopic, but I do plan to use this to help me create documentation for the * project.. so I guess it is somewhat on topic :) Anyways, I am looking for some sort of document control system. It should act somewhat like a CVS where it keeps previous versions, allows people to submit documentation, keeps track of who has what document open etc.. etc.. The

Hi all, Here is a graphical diagram of what I am trying to do: <SIP> <---> <GW/NAT/*> <--IAX--> <*> <--TDM400P--> <Analog Phone> So I have incoming SIP calls go to the * on the GW, which I then want to forward over IAX to the second * box behind the NAT GW. If I was to place a call on the second * box, it should then forward to the * on the NAT GW

Hi, Has anyone managed to get X-Lite to work with Asterisk using the iLBC codec.. I have just tried updating the the latest version 1079 (BTW this new version supports up to 10 proxy configurations, Not that I can see a reason to have 10 proxies setup, I would rather have the ability to transfer calls).. I can make a call using iLBC but the sound that I hear is just a lot of pop's and

Hi, I am trying to create Bid/Ask for each second from a high volume stock and the only way I have been able to solve this is using loops to create the target matrix from the source tick data matrix. Looping is too slow and not practical to use on multiple stocks. For example: Bids Matrix (a real one is 400,000++ length): Bid Time 10.03 11:05:03.124 10.04 11:05:03.348 10.05