similar to: LCR

Okay, I've read as much as I can, and I think i've followed instructions, but I'm still having problems with * and firefly... I can get outgoing to other freshtel working, but not incoming (I get the "not available" voicemail), or outgoing to landline. I'm using the debian asterisk package (0.9.1-RC1-4) My iax.conf has in general (under my FWD register, which

Hi, I'm having some problems getting calls to go out via freshtel. There dosn't seem to be any specific information on how to get it working anywhere. The only information I've found is here: http://www.voip-info.org/wiki-Freshtel and that dosn't give you any idea of how to actually get it working. I've tried adapting information from other IAX2 provider examples but have

I found version 2.0.0 of Firefly on the Freshtel site, but it only has the network setup options for the Freshtel network, despite the final statement on the page http://www.freshtel.net/firefly/download/ that says: ----------------- Standalone SIP / IAX mode: If you want to use Firefly on our network (with your own voicemail etc.) you will need to register a Firefly number. However, you can

Hello, Anybody have experience on this matter? "If the source file has been deleted and recreate at the next sync rsync will move whole file without apply delta transfer even the destination file exist" Please reply Thank you, Remo Angeloni ********************************************************************************************** *** PRIVILEGED AND

Ok I have spent the last week working on getting my small PBX to work. I will in the end only have 4 SIP extensions being either softphones of IP phones. Currently only 1 SIP config for testing. And at the this point it should be all fairly easy with all inbound and outbound to PSTN will be going Via Firefly/Freshtel.net in Australia via IAX. Inbound does work in it's current basic state.

Hi, I'm looking at connecting an analog fax to asterisk via an FXO card. The plan is to send faxes thru freshtel. Has anyone done faxing with freshtel? Cheers, -Shaun

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

With a service like http://www.freshtel.net/?show=home that uses IAX and has servers in Australia, is it possible for the audio streams to take a different path than the call setup and control? In other words can it work like SIP with canreinvite where the two endpoint negotiate audio streams between themselves rather than though the FreshTel server? Thanks

Hello everyone, this is my first email and I hope I do not say stupid things:) I'm trying to synchronize two folders on two different servers via a satellite link (rsync can be useful for its properties) only that I have a problem, some files are updated on the source, but rsync transfers the entire file on the target, not the differences. This is' the configuration: - Source: rsync

Why don't you just try it? From what others have been reporting, it shouldn't take you long to get it running. You can use speexenc and speexdec for testing. On Nov 12, 2008, at 2:26, "Vincent Burel" <vincent.burel at vb-audio.com> wrote: > ok, thanks for these precision, and do you have some measure about > CPU load > ? > i really would like to get a

Hello All, I have been looking for a commercial version of an IAX2 Softphone for Windows but the ones I have came across (i.e. Iaxcomm, Iaxphone, Diax) do not seem to have an updated version since April 2004 in some cases. We looked at Firefly but we sent emails to Virbiage/Freshtel with questions and could never get a response from them. Has anyone got any recommendations for commercial

Okay, you've answered part of my question, which is "What value equals silence?". I assume then that a (decoded) frame of silence would be a buffer the size of my frame (320 bytes) full of 0's. Passing this frame (a frame of all 0's) through the encoder causes it to blowup though.. In response to your answer below, I don't think I want to overwrite the decoded

Ok I have a challange that I can't seem to find a way to fix it. My Voicemail in * timesout after 30secs without fail everytime no matter what I do. I have incomming calls comming in through Freshtel IAX2, if it goes to SIP extension when it is online it can hang on for what ever time the call goes for. If however it goes to the Voicemail it will timeout at 30sec and I can't seem to

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

How to set-up LCR ? a. which companies can be used with LCR? b. how to set-up & maintain LCR? c. multiple connection to one gateway? Example: +886223456789 could be reachable via a. ENUM free b. Dundi free c. Voipstunt free d. Voipbuster free e. Nufone $ f. Voipstunt $ g. others with 4 concurrent connections $$ h. others with 3 concurrent connections $$ I am looking

Hi, After a few extra days of hard work, debugging, and many coffees, I am pround to announce that Asterisk::LCR has been released on CPAN. Asterisk::LCR is an open-source, Perl-based collection of tools to help you manage efficiently multiple VoIP providers with your Asterisk installation. It is capable of importing providers rates from multiple providers, comparing these rates, and

Hi all, I am planning to implement LCR routing on my already running asterisk server. Uptill now i have found out that asterisk has no support for lcr, i have to do something about it myself, for example using the AGI. Im looking for ideas here. Whats the best way to start implementing lcr in asterisk. Should i use agi and start implementing my own lcr script or is there any plugin available which

Hello, I short question: I want to connect Asterisk to OpenBSC with mISDN, mISDNuser and LCR. Do I need chan_lcr? I have: Asterisk 1.8 mISDN .v2 integrated in Kernel 3.0.22 mISDNuser lcr 1.7 HFC-E1 Evaluation board from cologne chip I tried to configure Asterisk with <./configure --prefix=/usr/src/lcr --with-gsm-bs> and it runs without errors, But make and make install didn't run

I have got Areskicc installed with AMP and I can't stress out how good and excellent this software is. I must say the author desire every credit for this software and I would like to say Thank you to Arezqui who wrote the software. Now there is one last thing before my whole asterisk calling card become prefect. And I am wondering whether anybody may know the answer of this. I

Hi, This is what I want to do: 1. Asterisk to answer calls via DID's, currently using SIPGATE 2. Provide a menu, and allow users to dial out. 3. According to the country and area they dial, the call should connect via one of up 4 carriers depending on cost. 4. If the carrier is busy it should go to the next one in line and so forth. I have tried to set this up, but it never answers the