similar to: Problems with IAX Clients, HELP ME PLEASE.

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

I have two units at customer locations in the Caribbean registering to a server in the US. Both units are connected to the Cable TV company's internet feed. If I run mtr to the units I see clean internet and low latency, but when I watch the CLI, I see constant problems. The audio quality is terrible, but I can't see why. Sep 3 16:42:46 NOTICE[20423]: chan_iax2.c:7014 socket_read:

All, I have an issue with IAX that I can't comprehend. Approximately every eight minutes my servers go unreachable. They stay unreachable for exactly 10ms. I have two servers running IAX and it happens on both servers simultaneously. I have searched the archives and see similar issues, but not the exact same one. I am on the current CVS stable version of *. Also, during IAX calls,

I just downloaded the NT version of R and cannot find anything concrete on loading packages. There is alot on how to do it under Unix. The windows FAQ briefly mentions it but does not explain it. I have donwloaded windows binaries of the package "clust.zip" and unzipped it into my c:\rw0632\library directory... So I have c:\rw0632\library\cluster Now what do I do ?

hi all, i surely overlooked something. but i am currently trying to do some benchmarking stuff with llvm. in my thesis i was using llvm-1.5 (i did some extension work, which i would have to frontport now). But unfortunately 1.5 is for some benchmarks unstable (at least on my box) such that llc sigsegvs sometimes, etc. So I wanted to give 1.9 a go and build that on the my intel mac book. I did

I have a ATA 186 with SIP firmware 3.1 when I changed the configurations to use the g.726 codec I received many erros and the calls doesn't work. I changed the fields: - LBRCodec: 6 <- the code for g.726 - TXCodec: 6 - RxCodec: 6 The errors: Jul 9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to calculate samples for format G726 Jul 9 13:15:37 NOTICE[1192491824]:

hi just donwloaded the 1.6.1 branch and made configure & install. so far so good. after staerting asterisk with: asterisk -vvvvcr Could not load features.conf == Registered application 'ParkedCall' == Registered application 'Park' == Manager registered action ParkedCalls == Manager registered action Park == Manager registered action Bridge == Manager registered

What's the importance of the impedance matching in a FXO interface ? Kind regards, Miguel

You need to use -emit-llvm http://llvm.org/docs/GettingStarted.html#tutorial4 -Tanya On Thu, 21 Dec 2006, Jakob Praher wrote: > hi all, > > i surely overlooked something. > but i am currently trying to do some benchmarking stuff with llvm. > in my thesis i was using llvm-1.5 (i did some extension work, which i > would have to frontport now). > > But unfortunately 1.5 is

Hello all, I'm new to nut, i've donwloaded and installed without problem apparently nut 2.0.3 on a slackware 10.x (1 i think). /usr/local/ups/bin/upsdrvctl start seems to work correctly. but when i do: /usr/local/ups/sbin/upsd i've got: Network UPS Tools upsd 2.0.3 stat /usr/local/ups/etc/upsd.conf: Permission denied but upsd.conf is 0640 and set with root.ups as stated in some

Hello ! I am having issues since the upgrade in debian from zaptel 1.4.7.1 to 1.4.8. I did m-a a-i zaptel to upgrade what i still get is: cat /sys/module/zaptel/version 1.4.7.1 It doesn't seem that it wants to load the new driver, any ideas ? Kind regards, Jan -- Jan Prunk <janprunk AT SPAMFREE gmail DOT com> http://www.prunk.si GnuPG: 1024D/00E80E86 77C5 156E 29A4 EB6C 1C4A 5EBA

So I'm attempting to build Wine from the the source files in the Git repository. After running ./configure I took note of the dependencies that were missing and went about trying to download, build and install them. I then re-ran ./configure and it still couldn't find any of the libraries I had just downloaded. My initial thought was that Macports was building the libraries as 64-bit.

I have set up FWD via IAX service. I have tested the IAX service with 613, echo test, and 612, saytime. It all works well. However when ringing a FWD user, I got this error all the time: Connected to Asterisk CVS-v1-0-02/01/05-09:34:45 currently running on chat (pid = 8282) chat*CLI> Verbosity is at least 3 -- Executing SetCallerID("SIP/1001-a1fb", ""David

Hi, Have just been testing asterisk 1.8.0, 1.8.0-rc5 and a trunk version from about half an hour ago. IAX Friends (Zoiper Softphones) don't stay registered for more than a few seconds they start out with status unknown and quickly become unreachable, I am using realtime with postgresql, with tables and configuration that have worked fine for IAX in 1.6 and a trunk release from a few months

I have a problem: When I'm in a call and a second call arrive (call waiting) I can't transfer the first call. If I press flash the line change to the second call, if I press flash again the line change to the first call. How I can transfer a call in this kind of situation ? Kind regards, Miguel

Powball gives every indication of starting up correctly and all, but there's a problem... Before it'll start, it wants to be fed a keystroke. Typing at the console has no effect, nor does typing with the focus on it's window. Is there any way to pass it a character via the command line invocation or something?

Hi, I had a few problems with IE regarding either Java script or active X under wine so before I started wining I upgraded to latest available version on Gentoo dist 20041019. I saved my config file and made sure it was not changed after installation. (it should not have, I use emerge as root). Now whenever I start IE I get: $ wine ~/.wine/fake_windows/Program\ Files/Internet\

I have downloaded Ububtu abd used Add/Remove to install Wine. I can not get it to work. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://www.winehq.org/pipermail/wine-users/attachments/20070906/86803328/attachment.htm

Dear members, I am trying to share the bandwith for all users fairly. because some users heavily uses p2p programs and a few hosts consume most of bandwith. when we limit each flow it does not work as one client has many connection at same time. Firstly I tried to write a script that use the ipfm(which accounts traffic usage) logs as follow .extract ip addresses heavy bandwith users from

Anyone out there have the AudioCodes MP-108 working with Asterisk? I am able to get the channels to registers with Asterisk, but anytime I try and send a call I receive these error messages: Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:771 __sip_ack: Stopping retransmission on '117801284512845hUxv-9991110061--17708185305@63.201.117.76' of Response 20587: Found Jul 6 15:12:10