similar to: Re: AGI and fast-entered DTMF codes (SOLVED)

Hi R People: I am looking for some objects: >objects(pat="f") [1] "dufus" "f" "f1" "fake.df" "ff" "fm1" "one.df" "one1.df" "x.df" "xf" >mode(objects(pat="f")) [1] "character" > I would like to determine the mode of these objects.

Hi all, I am trying to indentify a problem: I have 2 machines, one with Asterisk 1.0.11, the second with Asterisk 1.2.17. Both running with the same zaptel (1.2.16). Asterisk 1.0.11 running on Sarge with AMP's dialplan and the 1.2.17 running on Etch with FreePBX's dial plan. Now on both machines, I have some FXS connected (yes, I am talking about Astrinbanks...). The problem is that

Senior Quantitative Research Analyst Based in Irish Life Investment Managers, Lower Beresford Place, Dublin 1 Irish Life Investment Managers (ILIM), the investment management company within the Irish Life & Permanent Group manages assets of circa EUR30bn and provides fund management services to a large domestic and multi-national client base. We provide investment products to both

Hi, I know it sounds weird, and this is part of the reason I have not reported that sooner. As I upgraded from 1.6.2.x to 1.8.x several months ago I am experiencing this problem. If a call is initiated from a DAHDI extension after no DAHDI extensions were used for some time, arbitrary DTMF digits are skipped and the call fails. If the call is redialed it goes through. Normally just one (1)

http://bugzilla.mindrot.org/show_bug.cgi?id=1054 Summary: Nmap Causing SSH Session to Prematurely End Product: Portable OpenSSH Version: 3.8.1p1 Platform: All OS/Version: Mac OS X Status: NEW Severity: normal Priority: P2 Component: ssh AssignedTo: bitbucket at mindrot.org ReportedBy:

I have a Toshiba DK424 connected via T1 E&M to a TE110P card. Intermittently when a user dials a number I am getting 'getdtmf' errors on the Ast server and the calls do not go through. If they dial the number once or twice more, it works fine and I receive no DTMF problems. On another note, end users are complaining about intermittent disconnects. T1 is ESF/B8ZS - 24 chan. Other

Hi, I'm new to R. I'm using the R Gui and have tried to install additional packeges. When I choose Packages/Install package(s) I get the following error... utils:::menuInstallPkgs() Warning: unable to access index for repository http://cran.ma.imperial.ac.uk/bin/windows/contrib/2.15 Warning: unable to access index for repository http://www.stats.ox.ac.uk/pub/RWin/bin/windows/contrib/2.15

Hi all. I have a DTMF detection problem by my new analog card (ATCOM 2 FXO port). When i`m playing a voice with 'GET DATA' AGI command, sometimes asterisk do not receive DTMF from caller while the voice is playing. But if user waits to the end of playing voice, there is no problem. I`m using Asterisk 10.3.1, dahdi-2.6.1 on CentOS.6.4. Could you please help me? Here is my configs:

Yes and No The D500 is a terrible thing First problem: it sends some horrible DTMF, so if your voicemail is configured to send #H and *H, it will not work, configure it to send numbers, like 8H and 9H (H is a placeholder for the extension). I also managed to use the MWI (message light), it's a perl script that is in voip-info.org, but with a little correction because the wiki distorted it. If

Hi, I am using a channel bank and zaptel hardware. I have a credit card machine on one of the channels that appears to be dialing "too soon" for asterisk, every complete number recognized by asterisk is missing the first 1-4 numbers. This is a serious problem for me, anyone have any ideas on whats going on? The pstn picks up on the dtmf tones just fine. I was able to get it to

Hi, Can anyone please advise if there is a neat way to 'share' libraries of (previously developed) S+ code with R? Specifically, within S+ I can use the attach(what = "<Chapter Directory>", pos = 2) command, to retrieve previously developed functions and variables to position 2 in the search list. [Some may be intentionally masked by what's in the current working

The British and Irish Region of the International Biometric Society would like to bring your attention to a half-day meeting on Statistical Software Interoperability on Thursday 14th November 2013 at the headquarters of the Royal Statistical Society, Errol Street, London, UK. Please see http://bir.biometricsociety.org/events/interop for registration, and more details. Closing date for online

Hi, I''m developing a Rails app which uses dynamic forms. To implement these, I''ve used AJAX and Rails''s built-in form observers. Unfortunately, I''m having a problem. This is how the form starts: * Country select box [OBSERVED] (Republic of Ireland selected) * County select box [OBSERVED] (none selected, list of Republic of Irish counties in the drop down)

Hi guys, I am trying to use DISA. The scenario is - I call my home number (where X100P seats) from mobile phone, enter the password, enter international number and get connected via voiptel. It works perfectly when I call extension setup with DISA from X-PRO SIP phone, but when I dial into Zap, It seems that it does not detect DTMF tones. Here is a log and config files Please help

Hello all, I'm struck with a very strange problem today. I've an AGI with some code subroutine snippet as follows: sub enable_sbc($) { my $carrier = shift; my $tmp = substr($carrier,1); my $jkh = $tmp; $server_port = $ast_agi->get_variable("SIPPEER($jkh,port)"); $ser_ip = $ast_agi->get_variable("SIPPEER($tmp,ip)");

When I enable callprogress I get this error message... (when I call, it will ring forever but asterisk is acting as if it DID pick up the line ... but it never did... ) I'm beginning to think that asterisk can't hangup the line when in voice mail and this, whatever action I'll take... Actually, callprogress doesn't seem to work at all... (in my case) Mar 24 23:37:28

Hi guys. I have a XP100P Clone , and the busydetect dont work for me.. PSTN---Asterisk---Sip---Asterisk----PBX Any call from pstn side dont disconnect ... I have no disconnect supervision and busydetect dont work... Please Help me. Zapata.conf [channels] echocancel=yes usecallerid=no hidecallerid=no rxgain=0.0 txgain=0.0 signalling=fxs_ks callprogress=no context=entrada channel=>1

Actually I am using Asterisk 1.2.7.1, zaptel 1.2.5, libpri 1.2.2 I ve tried many values for rx/txgain togeher with echocancel and relaxdtmf. The detection is not working with call file, manager originate and not with the dial command to the mobile. I have no ideas left. I got it sometimes to work if I use a specific channel (i.e. Dial(ZAP/14/...) But with the same vaules on a second call there

Well, I thought and hoped my issue of random hangups on our TDM400P were related to busydetect=yes in zapata.conf. The behavior of a call being hungup has not changed, however, since setting the busydetect option to 'no'. Again, the only affected user is my loud talker... What are some causes/solutions to seemingly random call disconnects on Zap channels that people have seen? I have

Hi all, Tiny, but very important question for me: what it can be when standard agi-test.agi script, like: print STDERR "Testing 'waitdtmf'..."; print "WAIT FOR DIGIT 10000\n"; my $result = <STDIN>; &checkresult($result); on the call from KPhone application does NOT return any DTMF code back (I use dtmfmode=inband in sip.conf): Testing