similar to: [Dragon Age Origins] Official DLCs "Unable to load area"

I have managed to install DAO:Awakening. It is working in the sense that you can get into the game and play the game but a lot of item text is missing. For example, many new items have no description and appear in my inventory as gxa_im_(code) . Also I have unlocked battlemage by buying the book, however I cannot level up as battlemage, I cannot click the specialization button, nor does any text

Hello. I've read through the dragon age wine appdb page, installed all the requisite stuff through winetricks, used a fresh prefix and all that, but my dlc in dragon age is always showing up as unauthorized. I'm pretty sure the dragon age updater service is running, it just can't seem to see my account as being authorized to use the dlc. I have in fact bought a couple of dlc's such

Im currently working at it, and got as far as being able to resume a saved game, but only for a few seconds before it crashes. You can see my full progress in the AppDB. Since, despite many helpful posts, there doesn't seem to be anyone actually running it on SLeo over there, I was just wondering if ANYONE has been successful in that task, or if I'm the only lonely wanderer in the dark...

Hello, I read with disbelief on the Dragon Age wine appdb page that people actually have working dlc in Dragon Age Origins. This flies in the face of absolutely everything I've seen in my experience with the game. I've installed via Steam, installed the prerequisite windows crap through winetricks (dotnet30, physx, vcrun6, vcrun2005 vcrun2008, and everything else). I've manually

Mandi! Rowland penny via samba In chel di` si favelave... > yes, Provided you use the right attribute to search on ;-) Ah! ;-) Just i'm here, i test three condition in account flags, eg: UAC=$(ldbsearch ${LDB_OPTS} -b "${BASEDN}" "(&(objectClass=user)(sAMAccountName=$1))" userAccountControl | grep "^userAccountControl: " | cut -d ' ' -f 2-)

Hi all, we have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone. After that the SIP call session has been established (INVITE and 200 OK) we have the following codec situation: UAC ASTERISK UAS | ASTERISK UAC PROVIDER g711 <----------------------> g711

I hopefully cleared all SAMBA files and set up a fresh ADC using: samba-tool domain provision --use-rfc2307 --domain=UAC --realm=UAC.MGR --server-role=dc --dns-backend=SAMBA_INTERNAL --targetdir=/srv/files --adminpass="secret" --option="dns forwarder=172.16.6.11" The provisioning seemed okay, i.e. nothing hints at any errors and I see a DOMAIN SID as the final entry as

I need to do some testing, but before to hit by head on a known wall, i ask here. My AD domain get used (via PAM/Winbind) to give access to some other dervice, most notably here dovecot. When password expire (or users change it) the MUA try the old password some times, then ask for a new password; users cleraly get scared, press randomly 'OK' or 'Cancel', but if they press 2-3

Hi guys. I've installed Dragon Age : Origins, and patched it to 1.02. I visited the AppDB for it ( http://appdb.winehq.org/objectManager.php?sClass=version&iId=18283 ) and followed all the steps (minus the sound fixes; as there's no sound issues). However, the intro videos do not show, and when it reaches the main menu, the mouse cursor is stuck in the center of the screen and cannot

Another week, another Winetricks. Version 20110317 supports two more games: cojbib_demo "Call of Juarez: Bound in Blood Demo" (2009) luxor_ar "Luxor Amun Rising" (2006) and fixes a few bugs in Dragon Age: Origins. For bug tracker, mailing lists, and wiki, see http://winetricks.org In particular, http://code.google.com/p/winetricks/wiki/FeaturedGames has screencasts

Hi I'm sorry. I mailed the same question again. because, it cannot be yet solved. any ideas with asterisk? [Aug 20 14:40:12] WARNING[29315]: chan_sip.c:11806 check_auth: username mismatch, have <aaaa>, digest has aaaa at 192.168.0.1[Aug 20 14:40:12] NOTICE[29315]: chan_sip.c:20479 handle_request_register: Registration from 'aaaa <sip:aaaa at 192.168.0.1>' failed for

Going through exactly the same stuff. Currently researching down this path: fixme:winsock:WSAIoctl SIO_GET_EXTENSION_FUNCTION_POINTER: unimplemented TransmitFile correlated with the DragonAge_1.log entries that look like this: DA Download Manager (DADM) Attempting to connect to DADM Control Server! DA Download Manager (DADM) Connected to DADM Control Server! DA Download Manager (DADM) Failed to

Hi, is it possible to use Asteriks for translating SIP to H323 and vice versa? I am looking to implement the following Setup SIP UAC <-> SIP-Server <-> SIP/H323 Proxy <-> H323 Server <-> H323 UAC Basicly i want SIP fones to talk to H323 fones and and SIP Fones to access PSTN Gateway(s) in a H323 network. Anyone got something similiar running? Any ideas? best regards,

Hello, I have such a problem. I have an * configured as a peer connected to the gateway to PSTN. While calling to the switched off cell phone, the gateway sends to the * the SIP message 180 with the SDP part, and also a lot of rtp packets containing the operator's in band info. But * forwards the 180 to the UAC without the sdp part and also without the rtp stream. Is there any way, how

Hai, small note here.. > > Setting up the Admin user is interesting. I will try that. > But I could as well add myself to the domain admins group. > The name is arbitrary. Yes the name is but now you are working with Admin rights. Never ever do that. Please dont, use the UAC. It's so easy to get infected if you work as admin so dont.. Personaly here, at the office, i

>> The call that gets dropped had a retransmission of INVITE from UAC >> to UAS (and therefore retransmission of 200 OK from UAS to UAC). >> There is nothing wrong with the re-transmission as such, but I >> noticed a potential bug in Asterisk in the way it responds to an >> INVITE retransmission. Asterisk is bumping up the session version >> number in

I have the following scenario: g729 gsm UAS <-----------> * <-----------> UAC I am using sipp to generate the calls between the UAC and the UAS and sending some rtp from the UAC, I want * to do transcoding but as I see it is not. As long as I know 'Attempting native bridge' means only passing-through the rtp ?Am I wrong? The UAC and UAS are

We have Samba setup for our shared drive. I have pasted the smb.conf file below. Everything is working well accept when we try and run an EXE file using Windows Vista. When we run an EXE file it first ask for UAC control then it pops up the username and password prompt. You must then type your username and password in again before it will run. I think the issues is that UAC is now running the

Hello All, I have a little complicated question about the Dial command. I use OpenSIPs to loadbalance Asterisk Servers, and Users are registered on Asterisk servers. Asterisk use the Reg. Contact entry to reach the UAC via the OpenSIPs server. Everything works except for trunk numbers: For each peer on Asterisk, "Addr->IP" is IP of the Proxy and "Reg. Contact" is the IP

hey all, am wondering if anyone has successfuly done a SIP attended transfer using the REFER method (after an INVITE obviously) and the Replaces: header. we're writing our own SIP UAC and the asterisk code seems to support it, but we're not really sure if this is so. we plan on the following call flows: 1. incoming call from exten 1111 is sent to our UAC with Dial() 2. our UAC makes