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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

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

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

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 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 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

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

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

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 having a problem where the recall button doesn't work If i press recall before I dial numbers it disconnects me which is what I would expect, but during a conversation if I want to transfer the TDM 400 just ignores the recall Any advice would be gratefully received Thanks Robb

Nearly got an SPA922 phone working behind a NAT, the phone registers, and I can dial out and have two way speech, on an incoming call the SPA922 rings I answer and the SPA922 shows "Anwsering" but never does and the far end continues ringing until the voicemail answers, this then show as a disconnected call on the SPA922 I'm on the lastest firmware 6.1.5(a) Thanks in advance

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

Could someone point me in the right direction for setting up the mysql cdr function Thanks robb

Here is just one example of a warning when compiling asterisk on Ubuntu 8.10 manager.c:1760: warning: ignoring return value of ?read?, declared with attribute warn_unused_result is this anything to worry about? can i safely ignore it? 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

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-type: dtmf-relay > >> Application-signal: 16 > >> >

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

Hello users, I have been working with a conference setup. My setup includes: 1)There will be an interface number provided to the user which might be a DID number or A Toll free number When user calls the number it asks for the conference room number and the user pin . on successfull authentication he will be participated in the conference 2)by didaling the same DID number the

Hi Just recieved the above phone Does anyone have sip.conf and extension.conf example for the SIP phone working with the FXS w100p and the FXO tdm400d any help would be appreciated Thanks Robb