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https://bugs.freedesktop.org/show_bug.cgi?id=95351 Bug ID: 95351 Summary: lockup on kde startup Product: Mesa Version: 11.2 Hardware: x86-64 (AMD64) OS: Linux (All) Status: NEW Severity: major Priority: medium Component: Drivers/DRI/nouveau Assignee: nouveau at

Hello, I patched asterisk cvs head sources with http://juraj.bednar.sk/work/software/asterisk/messaging/ and presnce patch without success. asterisk send "405 method not allowed" to sender. I use polycom ip300. Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le...

https://bugs.freedesktop.org/show_bug.cgi?id=76173 Priority: medium Bug ID: 76173 Assignee: nouveau at lists.freedesktop.org Summary: xbmc green screen with vdpau enabled Severity: normal Classification: Unclassified OS: Linux (All) Reporter: serafean at gmail.com Hardware: x86-64 (AMD64)

...ER, that supports the feature. Is there also support in asterisk? Any pointer to documentation describing this is welcome. One more question -- is there a video conferencing support (like meetme, but for video)? I also found some development pages, but without code... Thanks, Juraj Bednar.

...tation of G.729 codec for use outside of US? I know it's a patented codec, but since there are usually no software patents outside of the US, I don't care about the patent license. I could use open-source implementation of the codec, if there was some. Any ideas? Sincerely, Juraj Bednar.

Hello, I see, that Rails is quite english-centric. I am developing some webs, that are not primarily in English. I have a few questions: - besides turning of plurals, what should I take care? How to use utf-8 for all data and converting it from local charsets to utf-8? - how do I make my page multilingual (i.e. adding english support later)? Is there something like gettext support? Is

Hello, has anyone seen or created a Debian Sarge package for 1.2beta1? I saw some for Sid, but I prefer not upgrading glibc right now, as this is a production server (Asterisk on it will be for testing). Thanks, Juraj.

Is anybody running SAMBA with NT4 SP4 clients ? -- +-------------------------------------------------+ | Milan Bednar mxb@inel.gov 208-526-8640 | | MAILSTOP EROB M/S 3640 | | | | "FACT times IMPORTANCE equals NEWS!" | +-------------------------------------------------+

...ny howto regarding Samba 4, DNS and Active Directory? Better, I would appreciate some lines describing the whole process of migration or someone who has already done this. Thank you very much. -- s pozdravem / best regards, *Vladim?r Bedn??* <http://pccomp.eu> *PC COMP s.r.o.* *e-mail:* bednar at pccomp.eu <mailto:bednar at pccomp.eu> ?esk?ch brat?? 302 *mobile (personal): *+420 777 834 332 *office:* +420 468 000 308 566 01, Vysok? M?to *mobile (office): *+420 608 535 301 *green-line (office):* 800 303 800 *I?:* 27545512 *DI?(VAT):* CZ27545512 *ICQ: *322 790 689 / *Skype: *vx...

Hello, I am migrating one server to dovecot. The only problem is, that users have logins with @domain as part of their user name. I want to use pam auth (for other reasons, if only for dovecot, I would use mysql, but I need the same password db to be used for other services, like samba). Is there a way to allow this type of login? Just cut everything beginning with @. I can change the

Hello, I found, that in CVS Head, in chan_sip.c, there's some support of asterisk. I have one question -- how does it map extensions to sip user names? When my client "subscribes" to other extensions' presence, they see all users online, but it may be because of voicemail fallback. Is there a way to map extension to some sip user's presence? Any ideas are welcome.

Hello, I would like to ask, if there's a way to transfer a call from some external program? I would like to build something like Asterisk Flash Operator Panel, with the ability to transfer a call using drag and drop. So I would like to connect to asterisk command line interface and transfer one side of a call to someone else. Is this possible somehow? Thank you, Juraj.

Hello, I'm looking for a way to have G.729 codec working on Solaris/x86. Binary codec from Digium is not compiled for Solaris/x86 (only sparc). Are there any alternative (free or commercial) G.729 implementations, which would work? I saw something from Intel and got it to compile on Linux, but it was only for evaluation purposes, so we upgraded to commercial codec from Digium. I

...ve registrations) is working too (all of the accounts). I've been told by my voice provider, that they are also using Asterisk on their side. I've tried upgrading from Asterisk 1.2.10 to Asterisk 1.2.12.1 and it did not help. Any ideas or help would be greatly appreciated. Juraj Bednar.

Hello, I was again asked to try to add support for presence (SUBSCRIBE/NOTIFY) and IM using SIMPLE. I have few questions: a.) are there any, at least partial projects, patches, anything, that at least partly implements presence and/or IM to chan_sip? I don't care about presence on other channels, I have one SIP client per user. I've read this list's archive several times and

Hello, I have a working ocfs2 cluster with two nodes running on Debian Etch with 2.6.20 kernel and latest Debian packaged (1.2.3-1.3) ocfs2-tools. Everything works well, I just can't reboot machine correctly, it hangs up on "Will now reset" message. When I do reboot -f, it reboots correctly. When I press reset on "will now reset", everything boots up and joins the

Hello, I would like to implement a home GSM gateway using asterisk. What would you recommend me as a low-cost hardware for creating a gsm channel? I found 2n gsm gateway, that supports sip and chan_blue for bluetooth connections. Any recommendations? Basically, I want to end calls to some GSM number in my sip telephone and for some prefixes dial out using that same sip telephone. Also

Hello, I would like to use Asterisk with fayn.cz service. They should be using a standard H.323 protocol, but I have no more information about this. They provide a softphone and/or rebranded H.323 telephone, but I don't know any H.323 settings nor if the firmware in the phone is modified. Has anyone tried this successfully? They provide a Prague telephone number reachable from

Hello, I added queuing support (based on SQLite database to store the queue) for my SIP Messaging patch. Works with eyeBeam, probably lots of bugs, but it's at least something. I created page about installation on the tips and tricks of voip-info.org: http://www.voip-info.org/tiki-index.php?page=Asterisk+SIP+Messaging Any bugfixes are welcome. Yes, it's a huge hack and supports

Hello, I have a polycom ip600 and eyebeam. When I call from polycom to eyeBeam, everything, including audio works. When I call the other side (from eyeBeam to polycom), I get no audio. In both cases, eyeBeam shows the same codec: g711u. Also sip show channels shows ulaw codec for both sides and correct addresses. I have canreinvite=no. I don't know if it's important, but asterisk