similar to: Grandstream GXP-2000 headset

hi, i''m trying move from file: based disk to tap:aio but things don''t work i have centos4 dom0 with centos4 domU xen 3.0.4-testing changeset: 13138:d401cb96d8a0 self compiled [root@xen linux-2.6.16.38-xen]# grep XEN_BLKDEV_TAP .config CONFIG_XEN_BLKDEV_TAP=m config disk = [ ''file:/var/lib/xen/test.img,hda1,w'',

please look at http://www.voip-info.org/wiki/view/Asterisk+SRTP and try compile&run clients with srtp (linksys,gxp-2000, minisip, twikle, ...) --------------------------------------- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA - http://lcna.slu.cz =======================================

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Greetings Everyone! Version 1.0.6 of Asterisk, zaptel, libpri, and Asterisk-addons has been released. There is also a new tarball for Asterisk-sounds. They are available for download on the digium FTP site: ftp://ftp.asterisk.org/pub/asterisk/ ftp://ftp.asterisk.org/pub/zaptel/ ftp://ftp.asterisk.org/pub/libpri/ ChangeLogs are available with the

For those of you who may be interest.... IAX2 loads are now available for the standard builds... http://www.aredfox.com/edownloadsiax2.htm Just a word of caution... Remember to change the ports over to 4569 from whatever. And don't forget to grab the palmtool from http://www.aredfox.com/download/tools/PalmTool.zip My own testing of IAX2 with both a phone and an ATA is that IAX2 is

hello, is there some way to notify people in the same pickup group about call from caller to callee? i.e. i have call from 111 to 222 there are 222,333,444 in the same pickup group 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup the call with *8 siemens have this on their sip openstage phones. how they do this? thanks -- --------------------------------------- Marek

hi, i have rh72 systems + updates libvorbis, libogg, vorbis-tools (xslt,xml2) recompiled rpm from rh8.0 ices2 klient celeron 1.Ghz 512RAM icecast2 server duron 700Mhz 256RAM 100Mbps network 4 streams 128 kbs ogg from playlist(random) i have noticed memory leaks in ices2 (randomly) what type of info do you need to correct this? (im newbie to debugging) --

> Olle E Johansson wrote: >> >> 23 apr 2007 kl. 19.55 skrev Russell Bryant: >> >>> John Todd wrote: >>>> To morph this into a -dev thread: if this patch were to become (again) >>>> useful and error-free, is there any objection or usefulness in adding it >>>> to TRUNK? Personally, I think there is, if there is a method by which

Hello folks, I'm having trouble configuring Asterisk to play an "invalid extension" message to anyone dialing an undefined extension. First I tried using the 'i' pseudo-extension, but it didn't work at all; searching the wiki I found that page: http://www.voip-info.org/tiki-index.php?page=Asterisk%20i%20extension where it basically says that the 'i'

hi, do you someone test this http://www.grandstream.com/y-gxv3000.htm? video works? (it's have H264 video codec) i want this topology gxv-3000 - nat -{Internet}- Asterisk -{Internet}- nat - gxv-3000 --------------------------------------- Marek Cervenka LCNA - http://lcna.slu.cz =======================================

The feedback we are getting so far has been excellent! As more is decided the list will be updated, if you'd like to be involved in helping, please join us on the IRC channel, #asterisk-uk on irc.freenode.net. If your company would like more involvement with the event, please email me directly. I would really like to hear from people/companies who would like to: - # Exhibit a product or

> hi, > > i have rh72 systems + updates > libvorbis, libogg, vorbis-tools (xslt,xml2) recompiled rpm from rh8.0 > ices2 klient celeron 1.Ghz 512RAM > icecast2 server duron 700Mhz 256RAM > 100Mbps network > > 4 streams 128 kbs ogg from playlist(random) > > i have noticed memory leaks in ices2 (randomly) > > what type of info do you need to correct this?

I upgraded from OpenSSH_3.0.2p1 to OpenSSH 3.4p1. Starting SSHD or ssh-keygen I'm getting the "PRNG is not seeded". I have verified that prngd is running and "egc.pl /var/spool/prngd/pool get" runs just fine reporting 32800 bits of entropy. My platform is Solaris 8 (sparc) and I downloaded binaries from www.sunfreeware.com. My guess is the build of OpenSSH 3.4.p1 is

Hello puppet users, I´de like to switch from md5 checksums to md5lite to save a few cpu cycles on my (overloaded) puppetmaster. so it is just replacing md5 to md5lite within my manifests and I´m done or do I have to keep things like the local clientbucket in mind? are there any procedures/recommondations regarding this? I´m running puppet >= 3.1.x on my puppet minions and master. bye ,

dear folks, when I run smbclient -L <hostname>, I've got the problem ; Added interface ip=10.1.1.32 bcast=10.1.7.255 nmask=255.255.248.0 Server time is Thu Jul 9 15:30:47 1998 Timezone is UTC+7.0 Domain=[AIFILE] OS=[Unix] Server=[Samba 1.9.16p9] SMBtconX failed. ERRSRV - ERRaccess (The requester does not have the necessary access rights within the specified context for the

I think I have 4 options. 1, Modify chan_sip.c to update a new field in sipusers realtime table with the status of the sip peer/user. Then use agi to dial sip calls. Check the status field if OK then dial the fullcontact from the sip table. If not goto voicemail or where ever else I want the call to go.. The UA would only register to one server, so only one server *should* be writing to the

Have you looked at his website, www.netenable.co.uk ? Looks like he pays bills the same way as he answers followups ;-) g -----Original Message----- From: asterisk-users-bounces@lists.digium.com on behalf of Peter Bowyer Sent: Sun 03-Dec-06 8:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] VoIP GSM Gateways Not very good at answering followups to

Fixes some of the more egregious problems with v1, and also applies properly to the head of git without needing any other patches. Rich.

Mark, While these samples are pretty good they do not work "out of the box" - there are a couple of issues: 1. the samples are 44100 samples/second and Asterisk needs them to be at 8000 samples/second. This is what happens if you prune out all of the Amercian voicemail prompts and substitute yours: Asterisk 1.2.7, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark

XXX No documentation. Only handles one disk. Network cards? Do we need to escape YAML format? What firmware types does kubevirt support. --- v2v/Makefile.am | 2 + v2v/cmdline.ml | 21 ++++++++++ v2v/output_kubevirt.ml | 103 ++++++++++++++++++++++++++++++++++++++++++++++++ v2v/output_kubevirt.mli | 24 +++++++++++ 4 files changed, 150 insertions(+) diff --git

Hi Everyone, I can connect the Jabra GN2124 + GN2100 (smart cord) to the Aastra 53i receiver port and I get a tone. But when I connect it to the headset port there is no tone. I am running firmware 2.4 and I can't seem to find that DHSG, EHS or whatever the setting maybe called to enable to get this headset work with the phone. Can anyone quickly tell me where the audio options are on this