similar to: Sometimes calls are silent

Hi all, I was excited to see the announcement on the list regarding the fix for the echo problems on Digium FXO cards! I have 2 X101P's, TDM400P with 4 FXS modules and couple of XLite softphones. A few months back,I had gone thru the recommendation on the list to remove echo from the SIP phones(I never did have any echo on the TDM400P FXS phones), and had removed about 90% of the echo.

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

Hi, I am running Redhat RHEL 4, authentification is via kerberos against and AD server, usernames are supplied via ldap service running on another redhat box - winbind has been seg faulting repeating when accessing samba - always the same error message... see logs below - can anyone tell me whats going on? Mar 14 16:12:45 firefly winbindd[14752]: [2008/03/14 16:12:45, 0]

I'm having a problem configuring asterisk to send incoming calls to Firefly. I can make outgoing calls from firefly through asterisk without any problems at all. The firefly client does this when it's on the same IP subnet without a firewall, or from a NAT'd environment. Can anyone tell me where I'm going wrong? Here is output from iax2 show peers: Name/Username Host

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

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

I have an asterisk server up and running, using Firefly in IAX mode works great, even with Firefly behind a NAT (as expected, since IAX works really well with NAT). Now I'm trying to get X-Lite and/or Firefly to work in SIP mode from behind the NAT, and I can't seem to get there. At this point, the phone will successfully register with Asterisk, and the Asterisk qualify messages get

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>

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 -

Just a quick announcement for Firefly users that Firefly 1.9.5 is out. Mainly just a bug fix release as we get ready for Firefly 2.0. One notable feature added is DTMF via SIP INFO. http://www.virbiage.com/firefly/download/firefly-thirdparty.exe is the URL As always, send me any bugs, features or suggestions. -Adam

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

Quick question from a newbie, I have asterisk configured to dial IAX extensions (which works). When dialing from one IAX extension (using Firefly) to another IAX extension (also using Firefly), the Firefly client rings on the receiving end and gives the option of accepting or denying the call. However, when I dial in to Asterix using a VoicePulse number and dial the same extension Firefly

Dear all, Question: How to merge two data frames such that new column are added in a particular way? I'm not actually sure how to best articulate my question to be honest, so i hope showing you what I want to achieve will communicate my question better. Lets say I have two data frames: > DF1 <- data.frame(cbind(Show=c('Firefly', 'Red Dwarf'), Measure=1:2,

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

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, I was messing around with FireFly last night and got asterisk to crash hard. It looks like the bug is a division by zero in chan_iax2.c. I reproduced it and here are some infos I got from gdb: [Switching to Thread 245775 (LWP 23251)] 0x41154918 in calc_timestamp (p=0x816b710, ts=0, f=0x424eef24) at chan_iax2.c:2896 2896 int diff = ms % (f->samples / 8);

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.

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

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

Dynamic DNS updating is failing (which is bizarre, because I could have sworn I'd had it working before). Help? Setup: Samba 4 DC running bind 9.9.2, Samba 3.6.3 member The output of "net -d10 ads join" is attached, compressed. Interesting portions of named.conf: options { (no allow-updates section) ... tkey-gssapi-keytab "/var/lib/samba/private/dns.keytab";