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I have a really weird issue which started when I installed some truetype fonts on my system. So far I've been able to verify this with Wine 1.1.3 and am about to test with Wine 1.1.13. doneill at mandy /World of Warcraft $ winecfg doneill at mandy /World of Warcraft $ winecfg err:module:attach_process_dlls "gdi32.dll" failed to initialize, aborting err:module:LdrInitializeThunk Main exe initialization for L"C:\\windows\\system32\\winecfg.exe" failed, status c0000005...

I am running Asterisk on our LAN with Cisco IP phones and softphone clients, SJPhone and Firefly. Everything on our LAN works fine and the quality is good. We have recently registered other callers not on our LAN who are using SJPhone and Firefly as well. The audio quality is fine, but I consistently hear a buzzing sound in the background. We have experimented with the line in and microphone

I currently have my Asterisk set up to "monitor" (record) all audio in my conference room on meetme. However, Asterisk will record an "____in.wav" and "_____out.wav" file for each user that joins the conference. Is there a way to set my extensions.conf file up so it only records when user when extension 1234 calls, for example? I'm assuming that the

I previously had Asterisk 1.0.7 running on a Linux 2.4.x kernel with ztdummy. I was able to do things like meetme and music on hold. I recently installed Asterisk 1.0.9 on a different machine with a Linux 2.6.x kernel running ztdummy. I installed and configured everything the same way, but when I try to call into a conference room I get the error message stating, "that is not a valid

I have Asterisk running on my LAN with softphone clients (SJPhone) and Cisco 7940/60s, all using SIP. I also have a few remote sites connecting to my Asterisk server. I am getting an echo back of my voice when talking with one particular site. The caller does not hear an echo on their end. All calls on the LAN or to other sites do not produce an echo. When the caller places his SJPhone on

I have Asterisk running on a Linux 2.4.x box with ztdummy. Once I did a modprobe on ztdummy I was able to enter into a conference room using my softphone clients. I'm using SJphone and Firefly. I have noticed a significant delay (1 to 3 seconds) while talking within the conference room. I have tried both clients, SIP and IAX protocols and various codecs. I have also tried it from different