Re: [asterisk-users] LUA

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Taylor Sent: Thursday, July 18, 2013 11:05 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] LUA On 07/18/2013 08:56 AM, jacob.e.mi...@l-3com.com wrote:

[asterisk-users] LUA

I am attempting to setup my server to use Lua for the dialplan (extentions.lua), but I am unable to get the asterisk configure script to find the installation of Lua on my box. I have downloaded the Lua sources from the www.lua.org site, and I have installed via the make linux install command. I

Re: [asterisk-users] LUA

I am attempting to setup my server to use Lua for the dialplan (extentions.lua), but I am unable to get the asterisk configure script to find the installation of Lua on my box. I have downloaded the Lua sources from the www.lua.org http://www.lua.org/ site, and I have installed via the “make

Re: [asterisk-users] AMI timeouts

Are you using a 3rd party java library such as asterisk-java (https://github.com/srt/asterisk-java), or are you doing your own Java AMI connector? I use asterisk-java and it has been working great. Jacob -- _ -- Bandwidth and

Re: [asterisk-users] Cisco SSCP to SIP

Even with the Cisco SIP firmware on the phones you still have to provide the XML configuration files to the phone via the TFTP. You need to have the XMLDefault.cnf.xml and SEPMAC ADDRESS.cnf.xml at the least... Jacob -Original Message- From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] Cisco SSCP to SIP

This is for signed XML files, some of the newer models require signed files for security. Is there a reason to use SIP? There is a really good SCCP module for asterisk (chan-sccp-b http://sourceforge.net/projects/chan-sccp-b/ ). Usually you have to set in the SEPMAC ADDRESS.cnf.xml what

[asterisk-users] Skinny directmedia

Asterisk 11 CentOS 6.4 Cisco 7971 phones Does chan_skinny support directmedia? Jacob Miles Software Engineer jacob.e.mi...@l-3com.com 903.457.4422 image001.png-- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Joining an astablished call

The best way I have found to do this is to use ChanSpy/ExtenSpy and then use the wisper/barge modes. Jacob From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Monday, May 06, 2013 9:46 AM To: Asterisk Users Mailing

[asterisk-users] Caller ID is not persisted when using Channel Redirect

Is there a work around for Caller ID information not being persisted when using the CLI or AMI Channel Redirect. A calls B (caller id is displayed), B transfers call to C (no caller id is displayed on phone c). Jacob Miles Software Engineer jacob.e.mi...@l-3com.com 903.457.4422

Re: [asterisk-users] extensions.conf / test DID

I am trying to make sure my DID and SIP account details are working properly and engaging the extensions.conf and dial plan. I have a successful SIP session registered: Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922) Asterisk*CLI sip show registry Host

Re: [asterisk-users] How to install in /usr/local/sbin instead of/usr/sbin ?

If this is the case then doing make install DESTDIR=../local/sbin should install in the /usr/local/sbin directory. It looks to be using a relative path starting in /usr/sbin/ Jacob From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On

Re: [asterisk-users] How to get louder voice ?

Not sure if this is what you want but you can always set the TX and RX Gain values via the dialplan. Jacob From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Tuesday, November 13, 2012 4:46 AM To: Asterisk Users Mailing