search for: vladyslav

HI All. I'm using tiff-v.3.5.7 and spandsp-0.0.1k with latest * cvs on Slackware-10.0. Here is debug messages from * console. Please advise. Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 0ms Get at 9600 Changed from phase 3 to 5 Fast carrier up Training failed (sequence failed) Fast

Does anyone configure that or is that possible ? Thanks in advance -- Best regards Vlad

Hi Ppl. Once, couple weeks ago when I have updated * from CVS-HEAD something happen and I could not send a fax anymore. After that I have tried previous * CVS versions with different versions of spandsp (0.0.1, 0.0.2pre4, 0.0.2pre10) but without any changes. I have tried that on Fedora Core 2 with libtiff-3.5.7-16.1 and libtiff-devel-3.5.7-16.1. Everything compiles smoothly, but when I try to send

Hi list, Please tell me why sometimes RSymphony.dll crashes on Windows 7 (may be on other Windows too). I saw that it depends on input data. For example this code bring to crash: library('Rsymphony') mat = c(0, 1, -100.37967, 0, 0, 1, 0, 0, -200, 0, 0, 1, -0.4, 0, 0, 1, 0, 0, -0.4, 0, 0.5, 0, 0, 0, 0, 0, -1, 0, 0, 0) mat = matrix(mat, nrow=6, byrow=TRUE) mat dir = c("=",

Good day All. Is there a way to pass DTMF signals to AGI script during conversation ? Actually here what I want to make: Users are usually dial using dialplan and when someone press *4 (during conversation) I want to have agi script to deal with that, but those users should keep talking and even didn't notice that one of them press something. Is there a way to do that or it's complete

Hi All! Have a problem with remote call pickup via sip. When 1 sip phone is ringing and I'm trying to pickup a call from another sip phone by dialing *8# I'm getting: -- Sent into invalid extension '*8#' in context 'from-sip-post' on SIP/ciscok-8d39 such configs: zapata.conf ------ context=inbound-analog callgroup=2 channel=2 ------ sip.conf ------ [ciscok] type=friend

Today morning cvs server checkout problem: cvs server: Updating asterisk-addons/format_mp3 cvs server: failed to create lock directory for `/usr/cvsroot/asterisk-addons/format_mp3' (/usr/cvsroot/asterisk-addons/format_mp3/#cvs.lock): Permission denied cvs server: failed to obtain dir lock in repository `/usr/cvsroot/asterisk-addons/format_mp3' cvs [server aborted]: read lock failed -

HI ALL. Please help. Problem: video calls drop after 15-20 seconds all the time. Use * latest cvs. from sip.conf [1102] type=friend username=1102 host=dynamic callerid=Veo webcam<1102> canreinvite=no disallow=all allow=gsm ;allow=ulaw allow=h261 allow=h263 from iax.conf [peer2] ; 192.168.0.7 type=friend port=4569 auth=md5 secret=second2 context=local host=dynamic qualify=yes trunk=yes

There is option automon => *1 in features.conf As I understand when *1 pressed during conversation => recording should begin. But unfortunately it doesn't work for me. I use CVS-HEAD-01/27/05 Does anyone has that feature working? Thanks -- Best Regards VladK

Just wondering whether we have a resolution to iconnect incoming problem, which started few days ago. Cheers SW -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040808/ecc99c4a/attachment.htm

Good day! Have a weird problem with HT-286 and Conference room. I use Asterisk CVS-HEAD-06/04/04. Here it is: When HT-286 get into the conference room first and nobody in that room everything seems ok (with any codec HT286 allowed), but when HT-286 get into conference room when somebody already there, have got different HT behavior: 1. When HT use GSM codec => it connects to conference room,

...now how to > >> fix it > >> > >> > >> May be you can find the solution in my post: > >> > >> http://lists.digium.com/pipermail/asterisk-users/2004-August/ > >> 058014.html > >> > >> Raj > >> > >> --- Vladyslav <vladk@azhelp.net> wrote: > >> > >>> Try to comment out in your sip.conf > >>> ;qualify=yes > >>> > >>> > >>> On Mon, 2004-08-09 at 06:52, Sathya Weerasooriya wrote: > >>>> Just wondering whether we have a res...

Hi there, we are looking for an opensource or commercial * based Call Center. Full ACD, call monitoring, multiple queue, IVR, voicemail, management, reporting, CDR, etc is needed. over 100 seat can be the initial target and will grow in a very short time. SIP phones will be used and multiple E1 lines incoming, so to provide full failover a cluster of * machines or some other form of redundancy

Good day, does anyone have pulse dialing working ? http://voip-info.org/tiki-index.php?page=Asterisk+zaptel+pulse+dialing At the link above there is a statement: configuration for European telephone lines will look like: make_time=63 break_time=37 pause_time=800 So where these pamameters should go to ? zapata.conf ignore them. I have tried on both version of asterisk (cvs and stable) with

Good day ALL. Could anyone tell me is there a way to get fax debug output into the file when running safe_asterisk ? ------------------------------------------------------ V.8 capable Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter, V.29 and V.17 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm or 255mm Recording length: A4 (297mm)

Good day list, I have a problem with ATA Handy Tone 286. It has been unsuccessfully downgraded via HTTP. Seems like during downgrade there was a problem with connection, because now it's not responding at all. There is no way to get to it's voice menu via phone (by pressing button on it). The button is just flashing ... Could anyone suggest any way to bring that device back to life.

Hi All. I have notice a problem with FM feature (screen macros) on Asterisk CVS version. When call goes via IAX and calling part "accept the call" on Dial command with option M, in macros context it's setting MACRO_RESULT=CONTINUE, but anyway it hangups both channels. If anyone faced with such problem please let me know. I need to know whether it's bug or just configuration