similar to: CDR destination when user presses '#'

Hi, If I do this in extensions.conf exten => 411,1,Dial(IAX2/hhandresen@iaxtel/18005558355@iaxtel,40,rS(10)) the line is cut of in 10 sec., thats fine, but in CDR I got dst as T, and not 411. How can I handle this so I still get kicked of after 10 sec., but get 411 as dst in my cdr ? -- mvh. Hans-Henrik Andresen

Hi, Have a problem in my extension.conf: I have: [sip] exten => _333.,1,wait,3 exten => _333.,2,Answer exten => _333.,3,AbsoluteTimeout,7 exten => _333.,4,Hangup I wanted to test if * is executing this dial plan by calling 3335254255 for example. The problem is as follow: It waits, it answers but it does not seems to see the Absolutetimeout: call goes forever. What's wrong? Am

Is there a way to get the que logs to go to a postgresql database? And/or are they any documents that explain the layout and field meanings? The logs look like they will be very helpful. Especially to show how long calls are held and who is taking them. I would like to make simple web reporting tool to show a daily summary status of queue activity. Also, there is not a manager command to do

Hi, I just wanted to point out that I have added a patch to Bug #1420 (http://bugs.digium.com/bug_view_page.php?bug_id=0001420), which adds Manager Events and Queue Log records for the AgentCallbackLogin application. Since the Bug Summary starts with "[Request]" and the Severity is "Trivial", both of which I can't change, this patch might be easy to overlook. Would

Hi There I am trying to get the following setup going. Two PCs (gateways) running Linux FC2 both have aDSL connections running. And both run the Asterisk PBX software, I defined the IAX trunks in the iax.conf on both sides, the systems seem to call eachother but never get the handshaking completed ... The consoles keep reporting errors on both sides. Is there a example for doing this ?? --

Hi everyone, Could someone help me on make my Asterisk registers to a Gatekeeper. I have compiled the chan_h323.so and it seems to be working. What I want to know is how can I "route" my SIP clients to a single account on a remote Gatekeeper. I have tried a lot of conbinations but nothing happend. For example: my account number: 123456789 ;extension.conf exten =>

I am trying to turn up DID trunks with our local phone company but do not know the correct format of extensions.conf to do this.

Hello I have contacted my line provider who is saying that in order to get my 0845 or 0870 number to id as the incoming number on a landline that i may call i need the following. User must provide - NPI set to E.163/E.164 User must provide - TON = "national or international I have had a good search around and can't seem to find a good answer to this. Does anyone have any idea where i

Due to the format of the message coming from the H323 channels included w/ Asterisk we were unable to use our gatekeeper. For a quick solution we tried the OH323 channel drivers and can receive inbound calls from the parent gatekeeper. We are trying to do a dial to gatekeeper... I am trying exten => 5551212,1,Wait,2 exten => 5551212,2,Dial,OH323/5551212 But I am not sure if this is the

I have need of a very simple callback function - when any call is made to a special SIP DID, the call is not answered but Asterisk then calls a pre-determined number - no need for CallerID to capture the calling number. Does anyone have a simple script to do this? Chris -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Hi We have a few conference numbers and all use MeetMe using the D option. We have noticed sometimes that the server is picking up more key presses than were actually done, i.e. the user presses 1234 for the pin and in the logs we see something like Created MeetMe conference 1022 for conference '12234' or Created MeetMe conference 1022 for conference '112334' Has anyone else

In our IVR we have a user enter a 6 digit number and information if returned. Our problem is that no all of the digits that the user presses are being recevied correctly. Its not as if the first digits are being cut off or the last, its just some digits aren't coming through. Our setup is that we have an IAX channel coming in from VoicePulse going to our Asterisk box via the T1 coming in to

I have verified this on two machines, but it would be helpful if others out there can reproduce it too. Also, I do not know if it is Xorg or the FreeBSD keyboard drivers, since I see no way to reproduce on the console (i.e. turn off repeat). In an xterm, type: "xset r off". Then try some multiple-key combinations (i.e. keep holding first key(s) when you type the next one): po (o does

The subject says it all. I'm betting there's a way to do it, but so far I haven't found the dialplan runestone via web searching. Thanks. b.

Hi, Does this report ring a bell here? I didn't check, but the image should carry an isohybridized 4.02 version of isolinux. The working (lenny) version is 3.71. Thanks, Feri. -------------------- Start of forwarded message -------------------- Subject: Bug#604245: Syslinux fails Date: Sun, 21 Nov 2010 14:13:41 +0100 From: Gyorgy Jeney <nog.lkml at gmail.com> To: submit at

Hello, I am using macros to give the ability to a call-receiver to 'accept' a call. However, any keypress connects the caller. Anyone have any suggestions about how to re-engineer this so that the receiver can deny the call, or press other keys to do other actions, without connecting to the user? Thanks, Jason Wolfe

Hi, I have the problem that WaitExten is not responding to key presses. Here are the sections from my extensions.conf: [globals] incoming_call=0 menu=0 announce=0 [internal] exten => 777,1,Goto(hotline,${EXTEN},1) [hotline] exten => _X.,1,Set(CALLERID(name)=Hotline) exten => _X.,n,Set(original_extension=${EXTEN}) exten => _X.,n,GotoIf($[${announce}=1]?4:10) exten =>

The way gnutar "preserves" atime is by noting it before the read, and setting it back after the read, thus wiping out a legitimate setting of atime occuring during that interval. Yeah, the netapps mess with unix times. Did you notice that mtime and ctime always match? Now that I know you're on a netapp, though, your problems are solved. Snapshot and sync from the snapshot, then

Dear All I am using Asterisk 1.4.17 in a calling card application. Following description explains the usage: A call/request hits asterisk from an ip xxx.xxx.xxx.xxx which opens a channel for this ip (Lets call it Channel A). Asterisk answers the call and play IVRs first asking the PIN and then destination number in an AGI making use of radius server for authentication/authorization. Once done

On 20/04/15 17:45, john wrote: > > > > Is this wheezy ? if so, it might be an idea to use backports, this > will get you 4.1.17 which is still in development, 3.6 is now EOL > OK, I understand a bit better where your problems lie. I would still use backports, supported code is (hopefully) better code :-) > > > I'd be willing to do that if it got me support