similar to: presence and video conference

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 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 nouveau Yahoo! Messenger

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, is there an open-source implementation 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 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, 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.

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 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

I have created shell script for MRTG statistics of droped/rejected packets: ftp://slovakia.shorewall.net/mirror/shorewall/mrtg/ http://slovakia.shorewall.net/pub/shorewall/mrtg/ rsync://slovakia.shorewall.net/shorewall/mrtg/ example: http://slovakia.shorewall.net/pub/shorewall/mrtg/example/ It is not based on /var/log/messages (syslog), but iptables counter. A lot of packets are droped/rejected

Hi, When I ran wine for the first time, font metrics information is built. It doesn't take a lot of time, but when you are forwarding X session over half a world, this can be pretty slow and annoying. Well, I know that there is probably no way how to avoid this, but I think there is a way how to avoid this when you reinstall wine. So, question is, where is stored font metrics information

Is Asterisk capable of handling video conference? I am wondering if there is anybody in the list who tried it with NetMeeting(s). If it is possible, is the * required to register in the GK for this purpose? or making it as h323gw only is enough. -------------- next part -------------- An HTML attachment was scrubbed... URL:

This Friday's edition of the weekly VoIP Users Conference call is all about wideband audio (aka HD Voice) and conferencing. The guest for this call is David Frankel, CEO of ZipDX a commercial service that specializes in wideband conferencing. We expect an interesting call touching on many aspects of VoIP going beyond the traditional phone service, conference bridges, technical standards,

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!" | +-------------------------------------------------+

Hello, I've got a problem with connection to my SIP provider. In general, everything works, but I get lots of these messages: Oct 17 19:10:06 DEBUG[29707]: chan_sip.c:11148 handle_request: That's odd... Got a response on a call we dont know about. Cseq 42710 Cmd SIP/2.0 Oct 17 19:10:06 DEBUG[29707]: chan_sip.c:11148 handle_request: That's odd... Got a response on a call

Hello, I am working on my PhD thesis and it would be really helpfull if someone could advise me, whether can Virsh create snapshots of VMs using copy-on-write. Thanks for reply. Juraj

Thanks for your answer, I am trying to make a snapshot of whole virtual machine (disk, CPU state, memory). But I need to make this snapshot in matter of seconds. I have already try to create snapshot of disk, which is not problem. I use qcow2 format, and create new disk image using original disk image as backing file. But still I am not able to assign state of vm with new disk. I have found

What's the status on using eyebeam with Asterisk, does it still require a patch to Asterisk to support the video component? I'm intererested in starting to use Video and audio telephony but wary of anything that requires patches. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM:

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