Not an Asterisk question, but…A bunch of our 8xx numbers started playing this recording when dialed. Our provider (Inteliquent) says its not them.Does anybody know who is playing it and what..
Asterisk 13.3.2I change the allowed codec from ulaw to g729 in sip.conf and enter sip reload on the console, but calls continue to use ulaw until restart.Before reload:lc10*CLI> sip show settings Global Signalling Settings:—————..
Asterisk 13.3.2The console command sip show settings shows the allowed codecs in the Global Signalling Settings but does not include the packetization setting.Similarly, both core show channel and sip show channel will show the codec(s), but not ..
Now that the g729 patents have expired, how do we use g729 in Asterisk?Will Digium be releasing a g729 codec for free use or do we download the free codec off the Internet now that we can use it without moral or legal res..
If I have a SIP endpoint defined in sip.conf using a host name instead of an IP address, do I have to reload sip to get Asterisk to re-resolve the host name if I change the IP address in my DNS?Does the answer change if the host name in sip.conf resol..
I googled about a bit without success, so…Is there a version matrix available?Something that would say: for kernel version w, you can run up to version x of Asterisk, DAHDI version y, and libpri version z?For example, I have a bunch of remote ho..
Are these incoming calls copper or VOIP?If you only accept copper calls, make sure Asterisk is only listening to 127.0.0.1 and enforce this policy with another layer dropping any incoming SIP packets at the firewall.If you only intend to accept ca..
I must have missed the memo, but the repos vanished on 2016-09-25 causing all of my yum updates to fail.Am Im the only one using ..
If you are happy with the way blacklisting with the Asterisk database works, how about a shell script that loads all of the entries in your blacklist file?Other alternatives involve modifying your dial plan. If you are comfortable with that then ..
I have a problem where SIP calls from some providers are dropping at 15 minutes.The topology is: Client sends calls to a host running OpenSIPS, OpenSIPS sends calls to an Asterisk server.Below,Client is the IP address of the clients host (running FPBX-2.8.1(126.96.36.199)OpenS..