similar to: Asterisk VS. Skype

Hey list (again - annoying bastard I am) I've played with Firefly/* for a while and I have yet to find a way to have * send voicemail notification to Firefly. It appears possible using SIP (no clue whether Firefly supports it) in the sip.conf file, but there's no mention of anything voicemail-related in the IAX.conf file. I'm using IAX with Firefly, so that might just be the

Hey, Can anyone comment on the difference between the 7905 and it's upgrade, the 7905-G ? Has anyone used these phones in a configuration? Is SIP well implemented in the 7905 ? Thanks in advance, Mat

After many months of development, I'm pleased to announced Firefly - an IAX soft phone and network. The firefly softphone - free, runs under windows, allows connection to multiple networks, skinable interface, connection to firefly network, IAX2 protocol, (SIP in next release), codecs supported - iLBC, G.711 ulaw/alaw, GSM. - contact lists, selectable ringtones. download from here -

As Promised, I've released a new version of Firefly (ver 1.8) with IAX & SIP support back in. Get it from Virbiage site or here's the direct link http://www.virbiage.com/firefly/download/firefly-thirdparty.exe If it crashes on startup, export your Firefly tree from the registry (current user -> software -> firefly), then delete tree from your registry. If that fixes it, send

Just an FYI FireFly no longer works with anything but the FireFly network. No more SIP, No more IAX. It was a damn good IAX client... too bad its crap now. bkw

Hi We've recently set up and are using with success 1.0.7 using a Junghanns quadbri card to BRI ISDN, and Firefly with IAX2 protocol as softphones Works very well, however we're getting cases where sometimes the call just drops. >From setting more verbose modes we get a log which is shown below. The problem seems to be the maxretries message which comes from chan_iax2. We are using

How come FireFly doesn't give me an Inband DTMF option? Only RFC2833 and Info. RFC2833 is the default, so I left it that way. I also unchecked all the codecs except g711ulaw to force that codecs usage. However, when I go to place a call, I get this: Nov 3 13:18:44 WARNING[53641241]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 Nov 3 13:18:44

There's a new firefly release out for those who are using firefly with your lovely asterisk / SIP server. http://www.virbiage.com/firefly/download/firefly-thirdparty.exe the main changes are improved GUI fixes (mouse wheel works now :) ), few url parsing fixes, mic volume control and improved compatibility with SIP servers (namely SER). Send me all bugs, problems and suggestions (even

Hi Does anybody know how to build the LibIAX2 from Virbiage? It has some nice features when using Firefly (Messaging, Status Indication). The source can be downloaded here: http://www.virbiage.com/3rdparty/. It does not contain any directions how to compile. Any hints? Thanks! Reto

Hey list! Some people are auctioning Cisco CP-7905 phones on Ebay, and I was wondering if those supported SIP. I think I saw something on voip-info.org about this, but I would like someone who has the same phones to confirm this. Anyone can? Good quality? OR If anyone knows of a place that sells those phones for a good price, I'd be interested. Buying a 250$US Cisco 7960 from Canada is

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

Good day, I've got around to installing an X100P card in my computer to try out asterisk. I noticed (and people who were testing with me also noticed) that when dialing from my SIP soft phone to the PSTN, the ringer tone changes after 2-3 seconds, precisely when the Zap channel takes over the call. Is it possible to eliminate the first ringing? Is there a reason to this

http://www.x100p.com/products_2.htm Anyone ever use this box? How's it compare with the Iaxy? I'd like to buy one or the other.. The Iaxy is appealing because to me, it seems less "no name", but this one says that it supports using hostnames, whereas apparently the iaxy only supports IP addresses?? That's appealing to the dynamic DNS guy in me! :-) Any

Hi all I'm looking for a solution to this problem. *box--------internet-----------nat-----------softphone We have potential customers who will be travelling the world with laptops/pda's. They need to be able to connect to the asterisk box via ip wherever they are and will have no control over nat whatsoever. I have read that STUN offers this service, but cannot picture in my mind how

Is there a softphone for windows that supports IAX? I can't seem to find anything out there...maybe im looking in the wrong places... Jeromy Grimmett VoipEmpire.com jeromy@voipempire.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050523/e668136a/attachment.htm

Anyone have any suggestions on free sip phone software for windows?? Only have one IP phone and want to have one other computer hooked up to my Asterisk box for testing.

I still having problem with remote SIP client, trying to use IAX client instead but i've to specify TCP port 8080 (because of firewall). I did this on server in bindport=8080 in iax.conf but i cannot find a soft client that allow to set wich server port to use. Any idea? Thanks, Antonio

Hello anyone who can help I have two Asterisk boxes with identical hardware (Dev & Production). I recently rebuilt the DEV box using Fedora Core 3 and the latest CVS Head. The hardware is an Intel CA810e, onboard everything with a PIII processor. The config is pure VOIP using IAX2 & ilBC with Virbiage Firefly soft clients. I also use Ztdummy which seems to be working ok - no error

G'Day, I have a bit of FireFly problem that hopefully someone has seen before. What happens is if I make to or receive a call from the FireFly network the call will connect successfully. However, around 10 seconds after I answer the call I am disconnected. The weird thing is same thing happens if I make a call. I've had a look at the * console and I can't see that my * PBX drops

Hi all, im Luis from argentina i started to setup asterisk in my network but i just cant... so i have a few questions, if some one can help me with examples and or explanations would be great! My Setup: Asterisk 1.0.7 (on 10.0.0.254) Gentoo 2005.0 1) Can i have LOCAL users on my lan? 2) what should i use AIX or SIP 3) can my local users have a phone number? (lets say im 1112 and i can call to