[asterisk-users] how to add extensions and sip registrations dynamically
Hi, I have inherited some code that appears to implement a kluge-y way of adding and removing extensions, sip devices, and sip registrations dynamically. Yep, you guessed it - it modifies the extensions.conf and sip.conf files, and then execute script to ask Asterisk to reload the dialplan and the sip module. Is there a better way to do this? :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] getting results messages from CLI commands via -rx
Hi, I am issuing CLI commands via script, using the asterisk -rx method. Its working great. Now, I need to get the results of the command to look for error messages, etc. I've tried setting several -v flags as well, but I only get the Asterisk startup text (version, license info, etc), not the results of the command itself. Is this even possible? Thanx! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] strategy for measuring conference audio delay
Hi, I have need to measure the end-to-end audio delay in the MeetMe conference application. Currently, I have written a python program that connects to an Asterisk MeetMe conference via SIP, and pumps RTP packets into the conference. Another instance of the program dials into the same conference and receives the mixed RTP stream. I figure I can have both python programs running on the same machine - essentially creating an echo test setup. Then, all I have to do is measure the time delay between when I send the audio stream to when I receive it. I think I'm being a bonehead, but the trick appears to be in what kind of RTP packets to generate and how to analyze the mixed audio stream coming back from Asterisk. Preferably, I would send one RTP packet that gets mixed in a predictable way and then I can look for it in the receive stream. Anyone have any suggestions on how I can do this, given my setup? Thanx! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk phone conferencing performance
Yes, I am using the ztdummy and 1.6beta Asterisk. There is no load on the machine - hovers between 1%-2% CPU usage during the conference. Thank you. You shouldn't have any delays at all. Are you using ztdummy for timing? and what kind of load does the box have on it? On Sep 9, 2008, at 4:23 PM, George Williams wrote: Hi, I just set up my first Asterisk with MeetMe conference support on my local LAN. It works great, but I think it may need a little tuning - I am getting audio delays of up to 1 second. Should I expect better performance in this area, or is this to be expected? Thanx! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk phone conferencing performance
Hi, I just set up my first Asterisk with MeetMe conference support on my local LAN. It works great, but I think it may need a little tuning - I am getting audio delays of up to 1 second. Should I expect better performance in this area, or is this to be expected? Thanx! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
