Archives : March-2015
I wasnt able to get much out of babytel, beyond the fact that I was, apparently, sending options which is why Im not getting 200 OK.How can I, generally speaking, ping/telnet or otherwise test the connection to get more data?A connection to this p..
Is it possible to log the raw signaling of Dahdi channels to a..
list,Im hoping that you could read through this mail and give me some tips on how to improve my setup (functionality, security, really anything). Its my first Asterisk installation and meant for simple home use.I installed Asterisk 11 on an OpenWrt Barr..
All;Im running Asterisk 11.6-cert9 and am trying to play a pre-recorded audio file to extensions using the Page() command. The dial plan looks like this: exten => s,n,Page(${AVAILCHANS},A(${AUDIOMSG})) and the paging by itself works great. However, w..
Hey i have an interesting topic to discuss here. The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support . the problems that i faced w..
Hello.I have plain text password for endpoint with outbound registration (someone elses server). My aim is to write it in pjsip.conf.md5 means that I know realm. I do not always know it.Is where any way?Dmit..
Hello!As I see there isdsp_drop_silence switch in confbridge. Could you tell me how asterisk detects silence?Is it possible to change silence level, so, lets say some not loud enough background noises will be recognized as silence and only loud eno..
Hello.Voice quality when calling – this is one of the most important in the PBX. You need to record the quality parameters for each call to improve.Because the overall quality of a call can only be determined upon completion, I did it in the HangUp hand..
, this is a bug?ERROR[361]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection available when trying to connect clie..
Kindly guide with debugging TLS issue in asterisk 11.16. Compiled from source and works all ok !Added the below to sip.conftlsenable=yes tlsbindaddr=0.0.0.0:5061However asterisk doesnt even listen to port 5061sudo netstat -anpKindly guideThanks Be..