similar to: Re: Incoming echo cancel

Hello ! Is there something wrong with the function HINT(<extension number>,n) ? Note the param n - it is supposed to get (as far as I understand the documentation) CALLERID(name) of the extension (via extension's hint). Example configuration: pjsip.conf: [10] type=endopoint . . callerid = Test extension <10> . . extensions.conf: [exts] exten => 10,hint,PJSIP/10 exten =>

Thanks for the information Lee ! Still, something is still strange to me, since this two Panasonic Fax machines are receiving at least 20 multi-paged faxes a day (they are in same office as both Asterisk boxes, and me :) ). Beside that, POTS lines in those two faxes are from same PSTN switch as line in X100P in one of Asterisk boxes, and the ISDN PRI lines in other Asterisk box is from same PSTN

> On Sun, 27 Mar 2005, Nenad Radosavljevic wrote: >> Only way I have managed to get Zap channel to reject a call on TE110P >> without answering it, is to dial number that is not handled in dialplan >> (I >> have a ISDN PRI with 100 number DID service, and about 30 of them are >> handled by dialplan). So far I didn't manage to reject call that are >>

Hello all ! Can anyone recommend me ATA device that REALLY has T.38 built in. So far I have heard of Telco Systems Access201, which seems to be impossible to bye in Europe (all resselers are droped Telco systems ATAs for some reason (tried in Germany and in UK so far)), and I have heard that SIPURA SPA-2100 should have T.38 built in into newer firmware, but I wasn't able to confirm that

Hi all ! I'm trying to make a automonitor generated filename to "make its way" into CRD(usrefiled), so I can keep track of recorded conversations in CDR logs. Looking how to do that, I have found cool (but almost undocumented) option of res_monitor: if you set monitor format in form of "format:<string>" (i.e. "wav:monitor"), res_monitor will prefix the

Hi ! On Cisco 7960 (with or without 7914 add-on module) when I press speakerphone button (or select line with line button - which automatically put second line on speakerphone) after about 15-20 seconds of dialtone Asterisk stable dies (seg fault). Tested versions of Asterisk are 1.0.2, 1.0.3 or 1.0.5, chan_sccp is newest form CVS of chann-sccp.sourceforge.net ). Firmware of 7960 is

Dear R helpers     Suppose I have a following data   y  <- c(9.21, 9.51, 9.73, 9.88, 10.12. 10.21)   t  <- c(0, 0.25, 1, 3, 6, 12)   I want to find out the polynomial which fits y in terms of t i.e. y = f(t) some function of t.   e.g.   y = bo + b1*t + (b2 * t^2) + (b3 * t^3) + ...... and so on.   In Excel I have defined y as independent variable, then defined t, t^2 and t^3 and using

I have minor echo on an IAX2 channel when using Firefly and a head set. I have tried various headsets and settings but still a little bit of the echo remains and I'd love to get rid of it. After some research I stubled on zaptel/mec2.h but it seem that it works only on the ZAP channel. Is there something I can do on the IAX2 channel?

Hi all, We have just installed a * server (CVS head) with a TE110P card and a IDSN20 line, we are using the GXP-2000 handsets running the latest firmware (.9). Some of the calls we are receiving have echo at the our end (we can hear ourselves speak). We have a traditional ISDN telephone system here as well and when I make a call from one of the handsets asterisk answers, directs the call to

Hi ! Does anyone managed to send multipage faxes (in single TIFF file) with app_txfax from spandsp package (i'm using 0.0.2pre18, libtiff 3.7.1)? If so, I'm interested in format of TIFF file that has been sent sent succesfully (tiffinfo <fax-filename>). I'm having problems with app_txfax, sending first page successfuly 99 % of the time, but never managed to send second or

Using TDM400's here and I have tried everything to cure the echo. I have used the Milliwatt test from the telco and from asterisk to tune RX/TX gain via a patched ztmonitor. What happens is I experience midcall echo. I turned on aggressive_suppressor and it seems to do great. The problem happens with misc. noise around the office will cause it to mute the other end of a phone call while

Hi. I'm working with i4l with asterisk CVS-02/21/03-13:59:12, plus i4l (chan modem i4l *dsp patched* and kernel 2.4.19 patched to disable dtmf). All seems ok (apart some echo issues that seems gone with mec2 aggressive suppressor), but outgoing dtmf doesn't work . or at least I hear the very first part of the dtmf, but then it seems suppressed. here's my modem.conf [interfaces]

I was fooling around with echo cans today. I found that I could reproduce this problem regularly t400p adit600 for fxs mec3 enabled Two speaker phones side by side within about 8" of each other. Put phone 1 offhook with speakerphone Dial phone 2 Phone 2 rings and is loud enough to be picked up by the mic in phone 1 By the second ring the echocan thinks it's figured things out, but the

Hi, I've got a small problem that has been driving me nuts over the last couple of days. I have a Samba (v3.0.11 on linux 2.4.21) server that is connected to the internet and has all smb and netbios ports firewalled. Access is provided via a OpenVPN tunnel interface with all traffic going through there. Clients get a /30 address with a route to the server. I have enabled WINS and the

Hi ! Did anyone had issues/managed to solve issues with DISA over Zap channels on * 1.0.X (STABLE) ? I have a situatuion where DTMFs that should be recognized in DISA work over SIP channels and do not work over ZAP channels (Zap channels are on TE110P) I have in default context: exten=> 299,1,DISA(no-password|default) and I have SIP extension 200 in [default] and I have Zap trunk which

Same problem here: if call come over ISDN PRI and it is for a SIP phone that equals to strong echo situation, at the SIP end. Interestingly this doesn't happen on all calls but it does on 95% of them. Asterisk load at that moment is insignificant - 1 to 2 calls. I have tried with all possible echo cancellers in zconfig.h, with and without MMX, and with and without CFLAGS+=-march=i686 in

I have Asterisk configured to record calls. Both in and out record ok but SoxMix fails to join the two files. The error from the CLI is as follows: Execute of ( nice -n 19 soxmix /var/spool/asterisk/monitor/Support-in.wav /var/spool/asterisk/monitor/Support-out.wav /var/spool/asterisk/monitor/Support.wav && rm -f /var/spool/asterisk/monitor/Support-* ) & failed. If I run exactly the

OK I see the ponit (although I never said that second page is interrupted - I said that in some combinations of resolutions and TIFF options receiving fax "spits" another blank sheet of paper beside the clearly received first page). I have read someware (some faxing tutuorial) that there is some kind of control code (6 EOLs I belive) that should be in TIFFs between pages and that

Hi everyone ! I have some aditional info on problem with TXFax and sending multi-page TIFFs. I have made 6 different multi-page TIFFs (Group3 1D with fillbits EOL aligned - 3 pages one TIFF in lowres and one in hires, Group3 2D -3 pages againg in both resolutions , and Group 4 - 3pages in both resolutions), and then tried to send them to Panasonic KX-F1100, Panasonic KX-F500 and SpanDSP

I was recently talking to a friend of mine who produces what's widely referred to as "techno", music that's heavy on electronically generated loops and effects, samples, and has plenty of bass. He doesn't like ogg for a few reasons - he sent me a rather long email, and I've stripped out the nonsense from it (he has some technically irrelevant reasons for not liking