search for: boardman

Hi I have looked through the wiki and search the mailing list, but I cannot find a way to intrude on a call, can asterisk do this feature? if so how? Thanks for your help Robb

2008/10/5 robert.boardman at gmail.com <robert.boardman at gmail.com> > Kevin P. Fleming wrote: > > Olivier wrote: > > > > > >> 2. R Hook-flash key is now available to transfer calls. > >> In s450IP web management server, its defaults settings are : > >> Application-typ...

Hi All I'm having problems posting to this list, no bounces the mails just dont show any advice how to get the postings through is there filtering? robb

...L L R L R L I want to analyse runs of turns but would like to automate how this is processed and end up with a table that displays the frequency of the sequences (L, LL, LLL, R, RR, RRR etc). Is it possible to construct something like this using R and how would I go about doing it? Thanks Simon Boardman [[alternative HTML version deleted]]

Hi, can anyone recommend a Linux based TFTP server to go on an asterisk box? Thanks in advance Robb

Hi All I cannot seem to find a way to stop atserisk inercepting DTMF tones and regenerating them even on a zap to zap bridged call is this possible? Thanks Robb

Hi there: I have this configuration: |-----[Server 2] | [Internet]--------[Router]----------[Switch]------------ [Server 1] | |-----[PC1] | |-----[PC2] | |-----[PC3] Server 1 has IP 216.251.XXX.XX1 Server 2 has IP 216.251.XXX.XX2 PC1 has IP 216.251.XXX.XX3 PC2 has IP 192.168.XXX.1 PC3 has IP 192.168.XXX.2 How do I configure shorewall in SERVER 2 to block to/from the Internet Port 22

Hi I have a problem with a new DECT phone I have bought The key pad works like a Mobile phone where you dial first then pick up the line, but it seems to dail too fast or spuriously, ie 012826736464 show on thew Asterisk console as 0012282677, could any one offer advice how to fix? Also when doing a ZAP bridge to this phone from an outside line the call is very echoy, but not an internal

I have two questions about incomming ring and extension ringing 1) When an incoming call is detected by asterisk it takes 2 or three rings before the internal phone ring does anyone know how I can fix this? 2) All internal phone ring on an incoming pstn call but after the call is answer all the other phone ring for a couple of tinkles how can I stop this from happening? Thanks for your

Hi I have a requirement to bridge Digital ISDN call through an asterisk box but no matter what I setup in the dial plan the second leg of the zap bridge is always set to Transfer Capability of SPEECH, I wondered if any one has come across this and managed to fix it? Thanks in advance for your help Robb

Hi I have had a Centos 5 installed with asterisk and zaptel for a couple of weeks, I had to reboot eh machine today, and when it rebooted it got stuck at "Starting udev" if I remove thew tdm400 it boots OK, but no zaptel has anyone seen this , and can offer any advice? Thanks Robb

Don't you just hate it when something was working and when you come to use it in anger it's broken :-( Something in the, fairly, recent series of Asterisk updates has broken DIGITAL call passthrough. I've an ISDN PBX behind my Asterisk Box (PRI ISDN comes into port 1 of a Digium Wildcard and the PBX it connected to port 2 via an ISDN crossover cable). This PBX used to be able to

Hi List I'm trying to update to the lastest kernel but I have a dirver that is not inculded in the distrubution, and I had to use the driver disk when installing centos 4.4 in the first place, The driver megasr .ko works fine with the installed kernel but I cannot find on for the updated kernel, any adive would be appreciated. without the updated driver there is a kernel panic on boot due to

Hi All BT are providing a SIP gateway for PSTN through the BT communicator with Yahoo Messenger, I have done an ethereal trace and found that the BT Communicator side of the software is using SIP, so in theory I could add more PSTN lines to Asterisk for BT using SIP, but I am having problems deciphering the trace so my question is has anyone else tried to get BT Communicator work with

Hi All Just been looking at stats for one of my sites, and I'm conserned about the number of error cause codes being returned from the telco for example 12000 calls processed 131 are cause code 31* normal. unspecified.* 139 are cause code 28 * invalid number format (address incomplete).* 112 are cause code 1 *Unallocated (unassigned) number. *this adds up to about 3% of calls not

...guide! > http://www.R-project.org/posting-guide.html > > ---------------------------------------------------------------------- > -- > > Nedbank Limited Reg No 1951/000009/06 > Directors: WAM Clewlow (Chairman) Prof MM Katz (Vice-chairman) ML > Ndlovu (Vice-chairman) TA Boardman (Chief Executive) CJW Ball MWT > Brown RG Cottrell BE Davison N Dennis (British) MA Enus-Brey Prof > B de L Figaji RM Head (British) RJ Khoza JB Magwaza ME Mkwanazi > JVF Roberts (British) CML Savage GT Serobe JH Sutcliffe (British) > Company Secretary: GS Nienaber 16...

...nks Tobias Brandt P.S. Apologies if this appears on the list twice but I suspect an earlier post was blocked since it was in html format. ******************** Nedbank Limited Reg No 1951/000009/06 Directors: WAM Clewlow (Chairman) Prof MM Katz (Vice-chairman) ML Ndlovu (Vice-chairman) TA Boardman (Chief Executive) CJW Ball MWT Brown RG Cottrell BE Davison N Dennis (British) MA Enus-Brey Prof B de L Figaji RM Head (British) RJ Khoza JB Magwaza ME Mkwanazi JVF Roberts (British) CML Savage GT Serobe JH Sutcliffe (British) Company Secretary: GS Nienaber 16.08.2005 This email a...

Hello everybody: This could be a stupid question, or may be not; I'm not sure 'cause I have not a very wide experience working with Asterisk, actually I just started last week. I need to make an IVR system work and I choose working with AGIs, written in Perl. The available documentation I've found show it as a very simple proccess, but it doesn't work for me... and I

Hi I'm Using Dev Kit TDM, every think seems to of compiled OK and I'm using the devkit lite sample configs from the Digium FAQ, but when I pick up the phone I get Dial tone OK But as soon as I press a button I get busy tone and Asterisk -VVVC shows -- Starting simple switch on 'Zap/2-1' NOTICE[245774]: File pbx.c, Line 1164 (pbx_extension_helper): Cannot

Hi all I ahve been having problems loading the wcfss and wcfxo drivers I get an error message invalid argument and something about post install insmod failed the currently load modules do show the drivers loaded but asteris won't start lsmod root@slackware:~# lsmod Module Size Used by Not tainted soundcore 3332 0 (autoclean) wcfxo