Archives : January-2019
Ive just read in [1] about SIP MESSAGE addition to both chan_pjsip and ConfBridge. It seems very interesting addition as it brings the capability to mix voice, video and text in conferencing.On an other hand, there are some softphones (Zoiper, Br..
all,we are working on aA to B basic Call scenario with early media. On that scenario we get a call from a PJSIP endpoint and we place a new call using ARI. On the created channel we receive a 183 Session progress where we have an announcement regard..
Is it possible to find real domain names instead of IP addresses in SIP URI?For instance, in a book dedicated to SIP (Understanding the Session Initiation Protocol), Im reading an example of a SIP INVITE that looks like:INVITE sip:411@salzburg.at;user=ph..
These questions crossed my mind this morning :In general, are anonymous international calls allowed (ie calling from one country to a number in an other country while hiding your own caller id) ?Are there special rules in Europe for this ?Be..
, imagine a ConfBridge conference with 10 participants. Now, one of them suddenly starts to yell and scream. Is there any built-in functionality (maybe not in ConfBridge, but in Asterisk itself) to identify the loudest caller? Or maybe already built..
When I last looked into this a couple of years ago, simple one-word speech recognition was rather complex and slow.At the moment, I use Google Speech Recognition which uses no local processing power, and is very accurate and fast, allowing me to ..
Page [1] gathers information on how to configure Asterisk CDR Radius backend. Im not familiar at all with Radius in IP Telephony.1. Would a Radius database and its associated tools allow live call accounting data displaying of an Asterisk instance powe..
Tony Ok, got this solved. I discovered my AMI message was corrupt due to a bug in our third party dialer app we wrote ourselves…! E. g. this worked on Asterisk 1.8: ActionID=12edad43-e817-427b-aa21-31a9659f86e1 &Action=Originate &Channel=SIP/local/3035@lo..
Ive just gone through the process of cross-compiling Asterisk 16 for ARM. I thought it would be as easy as calling the ./configure script with the appropriate host parameter, but it turned out to be more complicated. Im wondering whether I did someth..
Guys Ive run into a weird problem on Asterisk 13. Again something that worked fine on 1.8 but is now broken on Asterisk 13. I have an extension 3015. Im trying to originate a recording playback call on it via AMI by sending Action: Originate Action..