In old 1.4 you can kill -SIGINT # and stop the process. I dont seem to be able to do this in 1.8 any more. I know about/usr/sbin/asterisk -rx core stop now but in case it doesnt respond. How do I kill it . It seems to respawn itself. Even on a kill..
list, (Resending this email due to a typo in previous copy) I need to do E1 to T1 conversion for a project, and was wondering if there exists a card with both E1 and T1 on it. Or is it possible to use two separate cards in an asterisk box, one for..
I have been experimenting with CEL in a trunk version of asterisk for some time and have upgraded my test machine to 1.8.0 today. Made a few calls and it looks like the eventtype field is missing in the CEL insert query when using ODBC. I see the follow..
list, I need to do E1 to T1 conversion for a project, and was wondering if there exists a card with both E1 and T1 on it. Or is it possible to use two separate cards in an asterisk box, one for E1 and one for T1? (Please dont mention aculab or adtr..
I am helping a friend on one of his sip trunk and couldnt find the way to resolve his problem. His asterisks problem is like this: 0. When incoming call to one of his sip trunk, Asterisk reply with 488 Not acceptable here. So the call get dropped…
Everyone, For some reason a few phones connected to a pfSense box cant make or allow in OpenVPN in port 1194 UDP. So, I established the VPN tunnel on 1194 TCP and it works fine. I would like to know if there is any disadvantages to using TCP over ..
When upgrading-downgrading from one asterisk version to another, you may need to edit config files. For instance, when upgrading sip.conf from 1.4 to 1.8, it is tempting to use a single common file and add some #ifdef-like statements in it so that 1.8-speci..
Hi Im using asterisk 1.4.17 and have recently found an odd issue. When processing the CDR data on outbound calls Ive been using the channel field to extract which sip extension has made the call as I was under the impression that the channel name ..
How can I let asterisk immediately dials a trunk when off..
I wonder if I may freely use the default soundfiles that came with asterisk
(fpm-world-mix, fpm-calm-river and fpm-sunshine) on production server?
Are there any official sources of royalty free..