similar to: SERVICE CODES

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

Hello - I've read Asterisk should be able to activate a do not disturb feature to turn off the ringers on extensions. I checked the wiki and can't find documentation for how to do it. Here's my attempt, added to extensions.conf: [dnd-on] exten => _#78,1,Answer exten => _#78,n,Wait(1) exten => _#78,n,Macro(user-callerid,) exten =>

Hi all, Running bristuffed 1.2.4 system with solely Cisco 7940/7960 phones with SIP. I'm using also op_panel 0.25 (snapshot). I'm using * queues. I want to properly implement DND via *78 and *79. I'm using op_panel's documentation RECIPE 1 solution with astdb and dnd variables and this is fine for FOP. The DND works in normal cases, since I catch it with my Macro dialsip, HOWEVER

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 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 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

I have these very phones and took me a while to figure this out myself. The phone considers each line registration to be a line with a second line. So, call line while someone is on a call and another instance will appear below. That means you only need one registered instance for the phones to get two incoming calls. If however you want to have a second registered extension rung if the first

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

I'm trying to keep the DND status of my Snom phones and the astdb in line but I'm stuck on integrating my gui DND button which talks to * using the manager interface (actually it uses Astmanproxy as the gui host is on a different network to asterisk and can't see the Snom's across the network). All's working fine in my Dialplan; when someone dials the code for DND-on or

Hello all, this may be slightly offtopic :-) I have some Cisco 7940 phones with SIP firmware, connected to an Asterisk 1.2.18-BRIstuffed-0.3.0-PRE-1y-g (HorstBox Pro with custom extensions.conf). On some of the phones, two lines are configured, one for business, one for private calls. When forwarding a line to another destination (e.g. to voicemail), we can't use the phone's own

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

This depends on your application. As you say you are able to do everything you require in dialplan at that is great. I have used AGI fairly extensively becuase the stuff I want to do can't be done in dialplan alone. For instance i have written a auto attendants that can be dynamically controlled by a non-techie user with real time and in call reconfiguration. Also i have written IVR apps that

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 am running asterisk 11 and i would like to have features access codes in my system such as call waiting(all types) (enable/disable), call forward (enable/disable) and DND. my dialplan is pretty simple and it is the following [DefaultPlan]exten => _XXXXXXXXXX,1,Dial(SIP/VoipGate/${EXTEN},120,Tt)exten => _XXXXXXXXXX,1,Busy() exten => _4XX,2,Answer()exten =>

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

Robb, I wrote up a small tutorial on setting up the standard tftp server for linux check it out on my site. http://asterisk.titaniumsoft.net/ Mitchel -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Robert Boardman Sent: Thursday, May 13, 2004 2:44 PM To: asterisk-users@lists.digium.com Subject:

Hi, Is there a way to poll the DND status of a Linksys SPA9xx/5xx phone? The reason I ask is that I'm trying to implement DND + BLF on asterisk. However, the DND softkey on the Linksys phone does not send any feature codes to asterisk. On the flip side, if you disable the Vertical Activation Codes on the phone, then dialing the feature code doesn't display 'Do Not Disturb' on the

Hi Seems a lot of people are still downloading 0.6.0 and using it - my concern is that people will find it buggy and incomplete and give up on wxruby altogether. I think we''ve got wxruby2 to a point where it''s all round better than the old series, even though it''s not release quality, and we''re missing the DnD classes. So for the next release,

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