similar to: Hamachi with Asterisk

Hello Wine Users, I know that my question has already been posted once. (http://forum.winehq.org/viewtopic.php?t=2130&highlight=hamachi)But that thread did not really help me. Maybe I did something wrong [Rolling Eyes] This is the situation: I want to play "The Lord of the Rings: The Battle for Middle Earth - The Rise of the Witchking" with wine in a hamachi VPN. I am online in the

Hey all, I know that there are several posts on this subject but there seems to be no actual answer, however there are alot of things to try. Reason I've created a new post is that the others go so far then no 1 answers :P. Ok so I've got hamachi working fine, pinging and all with a linux native for obvious reasons. What I'd like to get working is Command and conquer 3 kanes wrath and

I was attempting to install Hamachi 1.0.3 via Wine (1.1.17) and got the following error message: wine: Call from 0x7b82fcc0 to unimplemented function setupapi.dll.SetupDiDeleteDeviceInfo, aborting wine: Unimplemented function setupapi.dll.SetupDiDeleteDeviceInfo called at address 0x7b82fcc0 (thread 002a), starting debugger... wine: Call from 0x7b82fcc0 to unimplemented function

Hi all, Got a very frustrating problem here, I've been trying to use hamachi to play a few games over it's virtual lan connection. With all the games in question, be they native or run via wine, they all seem to not be able to pick up on the hamachi network. My /etc/hosts file has the following entries. 5.86.189.163 sirbubbles-desktop 127.0.0.1 localhost 127.0.1.1 sirbubbles-desktop I

hey McFlow, this theme actually important for me now )) Me and my friend have the same problem in Command & Conquer 3 Kane's Wrath if you still there (or somebody else) please, tell us how did you fix it??? :( I have XP on my PC, my friend using Linux Gentoo. we make a chanel in hamachi, enter it. We pinging each other good but in the game I don't see him... and he sees me and my

You can try this, or check multiple solutions on the bug report URL I posted here. Lan play seems to be different from online play and you might have luck. With hamachi, however, no matter what I do it'll always fail.

Does anyone have hamachi running on Centos 4.3. I keep getting a Illegal instruction error. Maybe a security feature I don't know about? Thanks, Dan

Hello to all I've been playing Command and Conquer: Red Alert 3 (LEGAL version of the game) with wine and it has been working very well. However, I decided to try and play LAN (linux-to-linux) and did not manage to do it. There was a time where I could see my name twice on the list (should see once) plus my other pc's name. So this meant that: PC A Sees PC A TWICE and PC B ONCE PC B

Dear all, I have tried to run an asterisk instance together with XLite on a single machine (a PowerBook). The intent is to take advantage of IAX connections to easily cross NATs while traveling. While the IAX setup proved 'easy', just having to fiddle a little with working configs at both sides, I did not succeed so far in getting XLite to connect to the local Asterisk server, AND be

Hello! I bought a virtual IP line to my ISP to use with my asterisk but when I try to connect it to my ISP tells me I can not use and I can only use with a softphone that gives me, xlite ready configured. I use ngrep to see what information sent on xlite for communication, the User-Agent was changed so I change the User-Agent to my asterisk to the same as saying the xlite but still does not work.

Hi, I'm wondering if anyone has experienced an issue with the XLite softphone and asterisk accepting dtmf? I can listen to my voicemail perfectly from my hardphone. However when I dial the voicemail number from my XLite softphone and enter the password at the voicemail prompt, an error appears vm-incorrect and I get an "Unable to read password" message on the asterisk console. Has

Hi, I have an Asterisk box behind a NAT address and also a Xlite 4 soft phone behind a different NAT network. Asterisk -> Nat -> Internet -> Nat -> Softphone. I can register my softphone to the asterisk box ok via SIP but the RTP stream from the asterisk box is addressed to the private non-routeable address of the softphone when I turn on rtp debuging. How can I configure the rtp

Helo List, First I would like to apologize for my bad spelling as well as that I did not search the wiki first. I only have email access at the moment. I am having trouble setting both variables and global variables thru an extension. I am using Asterisk 1.2.4 with Ztdummy on CentOS 3.4 with an Xlite softphone. I have two xlite phones on diffent computers. One logs in as xlite1 and the other as

Asterisk version 1.2.14 I use snom190 and xliteV3 as sip phones. asterisk send the rtp stream never to the xlite softphone. Any hits for me? *CLI> rtp debug RTP Debugging Enabled -- Executing Dial("SIP/xlite-007918f0", "SIP/snom") in new stack -- Called snom -- SIP/snom-00797110 is ringing -- SIP/snom-00797110 is ringing -- SIP/snom-00797110 answered

Hello I have few questions : - what's the difference between a subscribe request et a register request ? - in asterisk 1.6 allowguest=yes or no param does it work ? if yes, please someone could explain how doest it work because I think i'm a little bit confuse. - if I configure a sip terminal in sip.conf like this [john] type=friend username=JOHN secret=mypassword host=dynamic

I have an asterisk server behind NAT - no audio on the test external calls I have tried making so far. Read http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions - No solution evident from there, sounds like I have case 9. I would have thought that all I would have to do is port foward and have the external IP on the asterisk server, which I have done I have fowared 5060UDP, 8000UDP, and

Hi guys, I'm testing the sending of a URL to an XLite softphone when a call is in queue. See the output of the CLI below: -- Executing Queue("Zap/69-1", "q_sample|tT|http:// www.google.com/") in new stack -- Started music on hold, class 'default', on Zap/69-1 -- outgoing agentcall, to agent '1000', on 'Local/ 1000@agents-1b94,1'

Hi, Asterisk 1.2.5 serves as a pstn gateway to voip users (xlite clients) by means of a X100P (a small test system) The pstn side has clear sound but i can hardly hear the other person (voip side)talking. The voip side (xlite xten softphone) has crackling sounds and also a very low voice volume (i can hardly hear the pstn person). Xlite connects using GSM to the asterisk machine, it's on the

I have build asterisk from latest CVS HEAD-05/09/05 with H323 support as described in README file. Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2 kernel-2.6.11 I tested it with following phones: -- XLite (SIP softphone) -- QMix SIP IP phone (PA168F) -- SJPhone (H323 softphone) -- QMix H323 IP phone (PA168F) -- FireFly (IAX2 softphone) Everything works fine except a problem

Hi guys i signed up to iconnect a few hours ago to try do some cool stuff. but im having a few problems. Im running asterisk and xp/xlite softphone.. both xp box and asterisk box are on public ips. The problem is that when i ring anyone in the world it'll ring they'll pickup and i can hear them 100% perfectly/clearly.. but they cant hear me.. occasionaly they can hear something like a