[asterisk-users] Compiling 15.2.0 and 15.2.1 Fails Others are Fine

Hi All, When 15.2.0 was released I tried to upgrade as I do when new versions are released but it failed to compile. I figured it may be a bug so I waited for the next release but 15.2.1 also fails in the same location. I can download, and compile 15.1.5 no problems at all. I'm not sure if

Re: [asterisk-users] Digium G100 and CID Dropping First Digit.

On Mon, Jan 15, 2018 at 10:41:27PM +, David Klaverstyn wrote: > port1 < Calling Number (len=12) [ Ext: 0 TON: National Number (2) NPI: > ISDN/Telephony Numbering Plan (E.164/E.163) (1) > port1 < Presentation: Presentation allowed of > network

[asterisk-users] Digium G100 and CID Dropping First Digit.

Hi All, I have installed a number of Digium G100 devices in many countries like South Korea, Japan, Singapore and Australia. I have just installed two in New Zealand and both sites are having a problem with Caller ID. Incoming calls are dropping the first digit 0 from the caller ID. I was

Re: [asterisk-users] Requirement of DAHDI

You do not require DAHDI Linux or Tools if you do not have any TDM devices unless you want to use MeetMe instead of ConfBridge. Regards David. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of luke devon Sent: Saturday, 8 June

Re: [asterisk-users] Help with GotoIf Command

Hi All, Thanks for all your suggestions. It turns out that the CID Name was sent with a space at the end of the number and that is why there was never a match. Regards David. -- _ -- Bandwidth and Colocation Provided by

[asterisk-users] Help with GotoIf Command

) exten = s,3,NoOp(they are different) The wisdom of this group is much appreciated. This is an example only. I know the code will display both NoOp lines if the name and number is the same. Regards David Klaverstyn

Re: [asterisk-users] Extensions routing

The way I accomplish this is by having an active/passive cluster. The two or more servers have individual IP addresses and running heartbeat creates a clustered IP address. The active server uses the cluster IP address. If the active server should fail then the cluster IP address moves to

[asterisk-users] MeetMe or ConfBridge live meeting Streaming to the internet.

media player. Regards David Klaverstyn -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello

[asterisk-users] Help with Codes and Polycom Phones

that match to in Asterisk? Regards David Klaverstyn -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

[asterisk-users] Asterisk 10 and DUNDi, Extended Support?

When I compile Asterisk 10, under PBX modules I see DUNDi listed under extended support. Does this mean that DUNDi will be depreciated in the upcoming branches and if so what is replacing DUNDi? Regards David Klaverstyn

Re: [asterisk-users] Anyone have a reliable T.38 Solution

I'm using the Linksys PAP2T and the Grandstream with SpanDSP and tx_fax and rx_fax on multiple installations with no problems. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Darnell Sent: Wednesday, 4

Re: [asterisk-users] How to make SIP guest call

Hi, The point of SIP guest calls is that there is no username and password required to make calls. If you have enabled guest calls then whatever extensions you have allowed in the allocate default sip context anyone will be able to dial. If you have in your sip.conf file context=from-vsp

Re: [asterisk-users] dahdi_tool missing

Thanks Alec. Doing a make clean does not fix the problem. I removed the folder as you suggested, extracted the tar.gz file and recompiled. All good. From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com

[asterisk-users] AGI with PHP

Hi All, I'm having problems getting results from a PHP file. This is what the CLI is showing. -- Executing [...@internal:1] AGI(Console/dsp, GoTalk.php) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/GoTalk.php [Dec 14 11:57:25] ERROR[20260]: utils.c:1019

Re: [asterisk-users] ${REASON} not getting set.

I believe it may be because you have not told what context the local channel should use. Try using: Channel: local/15...@mycontext Obviously change the mycontext to the name of the context that you want to use. That may work for you. -Original Message- From:

Re: [asterisk-users] LDAP Get for Asterisk 1.6.x

I'd appreciate it if someone could give me an answer to using LDAP in Asterisk 1.6.x From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Klaverstyn Sent: Thursday, 20 August 2009 4:01 PM To: 'Asterisk Users Mailing List' Subject

[asterisk-users] LDAP Get for Asterisk 1.6.x

What is everyone using in Asterisk 1.6.x to retrieve data from LDAP. The version of app_ldap I have only works with Asterisk 1.4.x http://www.mezzo.net/asterisk/app_ldap-2.0rc1.tgz Without a way to get data from LDAP stop me from using Asterisk 1.6.x Regards David.

Re: [asterisk-users] Anonymous Connection form IP to use specific Context

Hi All, I never saw a reply to this question. Is anyone able to assist? Regards David. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Klaverstyn Sent: Friday, 19 June 2009 2:28 PM To: 'Asterisk Users Mailing List - Non

[asterisk-users] Anonymous Connection form IP to use specific Context

Hi All, How can I force an anonymous SIP connection from a certain IP address to use a specific context rather than the default one defined in sip.conf. I am using Asterisk 1.6.0.9 Regards David Klaverstyn ___ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Free Fax for asterisk

Hi All, Sorry to hijack this post but I am confused. What is the advantage of using this Digium Fax For Asterisk product when you can use Asterisks' 1.6.x module app_fax or Asterisks' 1.4.x agx-ast-addons with the app_txfax and app_rxfax modules? Regards David. -Original Message-

Re: [asterisk-users] Asterisk and HUD server

Hi Marco, Try this: http://yum.trixbox.org/centos/4/RPMS/hudlite-server-1.4.32-1.i386.rpm Regards David. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marco Sambo Sent: Thursday, 23 April 2009 7:29 PM To:

[asterisk-users] Changing menuselect values from CLI and not TUI

Hi All, I'm in the process of writing an install script and I would like to change some settings for the install process but I don't want the user to go into menuselect and make the changes manually. Is there a way to make the changes to menuselect from the CLI? As an example, selecting the

[asterisk-users] FOP with Asterisk 1.6. No call Information.

Hi All, For some reason the Asterisk Flash Operator Panel is not working since moving to Asterisk 1.6 from 1.4. I did a complete install onto new hardware. FOP will show an extension and trunk offline when it is offline. It will also show a call in progress to MeetMe but it will not show

[asterisk-users] Asterisk 1.6 call files Disposition=NO ANSWER

Hi Guys, Since moving to Asterisk 1.6, whenever I am using call files the call is always logged with a disposition of NO ANSWER even though the call is connected and answered. The duration displays the correct time. Can anyone explain as to why when using call files the disposition is not

[asterisk-users] Call Files

I have successfully created call files and I can get Asterisk to make calls based on those files. The problem I have is that it seems you need to use a Channel for the first leg of the call file. This means I have to use either a ZAP, SIP or IAX2 channel. What I would prefer to do is send the

Re: [asterisk-users] Asterisk installation

Nothing changes except for the files. /etc/zaptel.conf becomes /etc/dahdi/system.conf /etc/asterisk/zapata.conf becomes /etc/asterisk/chan_dahdi.conf -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Sent: Saturday, 1 November 2008 8:49 AM To:

[asterisk-users] Enter Value and continue dialplan

Hi, What function or application do I use to get people to type digits into the phone and store the value into a variable? The application WaitExten is not what I want that will jump to the new extension. I want users to enter a number into the phone and store it as a variable so I can

Re: [asterisk-users] Asterisk 1.6 CDR no Clid information

I found the problem. In the file /etc/asterisk/cdr_mysql.conf the default setting is to change clid to callerid. I remarked the line and it is all working. [aliases] start=calldate ;callerid=clid From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Klaverstyn Sent

[asterisk-users] Asterisk 1.6 CDR no Clid information

Hi All, For some reason since moving to Asterisk 1.6. my CDR records are no longer displaying the Clid field. The CDR records contain the Source field be for some reason not the CID details. I am logging CDR to mysql. Is anyone able to help? Regards David.