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I've got wine 0.9.57 and installed directx 9.0 but it seems doesnt work. I've installed new ati drivers, new audio drivers. When I try to play at Assassin's Creed i got an error: Code: fixme:d3d_shader:shader_glsl_load_constantsI >>>>>>>>>>>>>>>>> GL_INVALID_OPERATION (0x502) from glUniform4ivARB @ glsl_shader.c / 279 there is no

Hi to all, to manage properly a call center for multiple companies is possible to let the X-lite/X-Pro softphone to display the number or context called from PSTN to let operator answer with the correct name of the company?? I explain better. If a call come from PSTN to Number A for company A i want the operator recognize it and answer "Good Morning, I'm Operator of company A"

Hi, i read the * support ringing multiple devices at the same time, i inserted this line on my configuration on default context: exten => s,1,Dial(SIP/260&SIP/261&SIP/262&SIP/263|30) exten => s,2,Voicemail,u260 exten => s,3,Hangup And i have both 4 clients in sip.conf . The problem is that if i call it fall immediately in the Voicemail if the client 260 is not registered .

Hello Samba List, I am currently running samba 3.3.2 joined to ADS and I am consistently getting this error in the winbind log: no backend defined for idmap config DOMAIN Also, I seem to lose association between user and uid. I think it happens when the winbind cache expires. Here's my smb.conf: [global] workgroup = DOMAIN realm = DOMAIN.COM server string = server

Hi to all, i saw that in chan_sip there is the possibility to let the * to take the number from the Remote-Party-ID header field on incoming calls from gateway. What about to let the * to generate the Remote-Party-ID on outgoing calls? this is is useful for us to let the users to have their outgoing number hidden but let our switch to get the correct record for accounting. I think that If i hide

Hi to all, we have a community of people on an * box that use SIP softphones to talk each other. Can you suggest me the quickest and simple way to let someone know who is online without have to call one by one the persons to look if they are present or not?? Something the user list in Microsoft Messenger. I was thinking on some sort of web page that can check the registration of the sip

Hi all, We've seen several examples recently of performance opportunities on POWER if we can improve the location of save/restore code for callee-saved registers. Both Nemanja and myself have discussed this with several people, and it seems that there are two possibilities for improving this: 1. Extend shrink wrapping to make the analysis of callee-saved registers more precise. 2.

I''m really newbie in roby on rails. I''m studing but I can''t find an answer to this question: I''m looking for the classic tamplate that I can found on every ruby-on-rails generated sites: I would like to use in my site I can found some already listed? -- Posted via http://www.ruby-forum.com/.

Hello, I am attempting to dynamically create user shares when they connect to the server based on their username. I cannot use [homes]. My reasoning for this is that the users require a $ at the end of the share or it becomes confusing to them(long story). What I'm seeing is that some Windows XP clients will connect to /home/<username> but other clients try to connect to

Hi, i'm using * with SER and a cisco 3725 as Gateway. I noticed that the reinvite is not working if i use SER and if i don't use IT (*---->Gateway) the reivite works so the * server is able to let the RTP direct from gateway to SIP Clients. Do you know in which way can i let it work with the SER too. Becouse i need SER to manage other VOIP communities but if i'm not able to use

Hi to all, can you suggest to me the best way to avoid problems in the CDRs for anonymous sip calls? I have some peoples that set Send Anonymous : Yes in their Grandstream phones and i don't receive the username as phone number that i use to make billing. It is empty. The only place where there is the phone number is in the peer name where it write the name of the peer that in this case is

Hi, notice that i have Grandstream phones and i have the problem if i activate the Send Anonymous function on them. If i do not activate that option the ACCOUNTCODE is correctly populated. SO i think it may be a bug of asterisk. I'm using Asterisk CVS-HEAD-10/07/04-18:07:25 . Thanks, Bye, Marcello

I got HD audio baised on Realtek, I have installed appropirate drivers from realtek website, they work in Slax, but they don't work with WINE. When i go to AUDIO card in WINE its says no_audio, what could be wrong?

I try to launch AssassinsCreed but it doesn't work on WINE (ver.0.9.57), I got something like this: fixme:system:SystemParametersInfoW Unimplemented action: 59 (SPI_SETSTICKYKEYS) fixme:system:SystemParametersInfoW Unimplemented action: 53 (SPI_SETTOGGLEKEYS) fixme:system:SystemParametersInfoW Unimplemented action: 51 (SPI_SETFILTERKEYS) What Could be wrong?

Hi to all, i already searched the archive without finding a solution to my problem. I have asterisk installation 1.2.18 to support multiple virtiual PBXs. I use SIP peer in the format <ID>-<EXT> to let every virtual PBX to share the same numbers of EXT. Ex. (PBX ID 10 Extensions) 10-101 10-102 10-103 (PBX ID 20 Extensions) 20-101 20-102 20-103 I use some rules in the dialplan to

Hi, How can I integrate Asterisk to Nuance TTS engine instead of Cepstral? Has anybody done this? How is the architecture and can Java AGI be used to communicate between them? regards, Vela Sivasankaran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091021/56254a0e/attachment.htm

I have just got a Cisco 7941G and am experiencing the exact same problem (phone is requesting .tlv file from TFTP server and never asks for .cnf.xml file). The phone originally had SCCP on it, but I downloaded and flashed with the latest Cisco SIP image (8.4(3) released 2009-01-13). In reading your message below, it looks like you were going to try an incremental upgrade?did you have any

Vuoi Davvero Guadagnare con Internet? Bene, salva su disco questa pagina per averla a portata di mano anche se il tuo PC non ? connesso a Internet, poi copia quanto segue in Word o in Blocco Note e stampalo, cos? lo potrai leggere con pi? attenzione. Questo Sistema ? diverso da tutti gli altri, quindi non essere precipitoso nel valutarlo senza averlo compreso a fondo, ma ti assicuro che

Hi to all, following the suggestion of Matteo Brancaleoni to solve my problem of anonymous call in CDRS i implemented the accountcode in any peer that i have in the sip.conf so i can use the SetCallerID to accountcode to identify the caller. When i make the sip show users the accountcode is correctly displayed but when i try to access the variable ${ACCOUNTCODE} from the extensions.conf to

Hi to all, i'm wondering to realize a dynamic macro that can take the number of extensions to RING,the ring type and all the parameter in a dynamic way. I have done this code to test it: macro pbx-ring-group-ael(pbx_id,num_int,ring_type,timeout,ext_string) { //; pbx_id = Id of PBX in the DB //; num_int = Quantity of extensions to ring //; ring_type = Kind of RING (C=contemporaneous