search for: gamehouse

I am trying to run "scrufs" one of gamehouse games. As all gamehouse games, at start it shows screen where user should register or to choose play free trial. I instaled mozzila plugin or something. This screen shows, but when I try to click on links like "start" or "register" i get an error in terminal Code: >err:mshtm...

...hear a half second of noise, the line goes dead and gets hungup. In my case, Zap/1 is the first channel of a 24 channel T1 line to a Fujitsu PBX. All seems configured right as I can dial out and the cell phone will ring. This is very frustrating, can anybody help out w/ this? -- Jesse Keating GameHouse -- Systems Engineer

...1, where if I call VoiceMailMain from a sip phone, and then either put in incorrect passwords or just hang up, I never get a Spawn Extension that hangs up the call, and my sip phone is not capable of making any more calls until I restart the daemon. Can anybody help me fix this? -- Jesse Keating GameHouse -- Systems Engineer

...art Asterisk. In IRC, a developer seemed to think it had to do with me using switch => in my dial plan. Basically I never see the calling extension get the -1 signal. Can somebody help me figure out why this is happening and how I can fix it while still using switch => ? -- Jesse Keating GameHouse -- Systems Engineer

I tried following the Wiki page regarding the Polycom 501 and having the same extension appear on all 3 line buttons (just like my cisco) but I'm having no luck. Has anyone else had success in doing this? Perhaps someone who has been successful can update the wiki? Thanks, Matthew http://www.voip-info.org/tiki-index.php?page=Polycom+Soundpoint+IP+501

Hello, Sorry, I know I read this somewhere but now I can't find it when I need it. I'd like to force a call to go out one line if we dial '9' first and then the number. Same for '8' only I will force it out a different line. There is a parameter or a method to allow the dialtone to come back after pressing the first 9... but I can't remember how to do it. Anyone know?

Hello, Can anyone point me in the direction of software to monitor channel usage on voice T1s? Using a TE410. The wiki documentation seems geared to SIP channel usage.... Thanks Charles -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051004/8450d0c2/attachment.htm

I have installed Asterisk and i have configured with two SJPhones; i am able to make calls between these two phones.I am planning to develop a application basically web based application from which the administrator able to trace the call logs or call summary, i mean from which user agent to user agent call is going , and what is the staus, if second user tranfers the call to the third

Hi folks, New to the list. Just updated the bootrom and app firmware on a Soundpoint IP 501 as per: http://www.voip-info.org/wiki-Polycom+Phones Updated from: to: APP 1.4.1.0040 1.5.2.0054 BootROM 2.6.1.0003 2.6.2.0032 After I did this, it appears that the Web interface for the phone won't change the settings, nor will it actually reboot the phone now. What do I

...ran into an issue. WHen I try to add a new contact into the directory, I am not able to. A window blinks really fast but the entry isn't saved. When you exit the Contact Directory system you get a 'Busy! Please try again' window. What the heck could be going on? -- Jesse Keating GameHouse -- Systems Engineer

Is there a way I can have "voice mail check" calls coming from my internal users automatically get to the right extension, without having the user enter their extension? I'm thinking that I could have the local SPA boxes translate, or have each user live in a context where the extension in question exists uniquely per user, but both of these seem kludgey. Thanks in advance for