similar to: Recall doesn't seem to work

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

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

Hello. I'm trying to transfer calls from an analog phone (Zap/1, TDM400P card) to some other terminal connected to my Asterisk PBX. If I make a flash hook pressing the phone hangup button quickly it works as expected, I get a new dialtone and the other side is put on hold. But I would like to use my phone's "R" key instead for some different reasons (it's quite easier to use

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

Is there anyway to change the "flash" time on a TDM400 phone port (a user has a phone that seems to generate a short flash which isn't being picked up). Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN steve@gbnet.net Euro Tech News Blog

Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 <-> PSTN gw)? - Asterisk ip: 192.168.1.10 - h323<->PSTN gw: 192.168.1.20 I've tried: exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) or exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) but it does not work at all. If my h323 client

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

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

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

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

Hi All I would Like to run a macro in a meetme conference when a user presses a certain digit sequence, but I cannot seem to find how to do this , is it possible? if so how? Thanks for you help Robb -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091119/9ff3f816/attachment.htm

Hi I was wondering if there are any special settings that I need to be able to transfer calls. Whenever I press the 'recall' button, I just here a click, and no ring-tone to transfer. in my debug log I get this : -------------------------- Feb 24 09:09:27 DEBUG[19216]: Exception on 10, channel 1 Feb 24 09:09:27 DEBUG[19216]: Got event Pulse Start(14) on channel 1 (index 0) Feb 24