Archives : July-2019
Im currently reviewing the Digium wiki on migrating from chan_sip to res_pjip and Im trying to access the script that is provided to help with conversion. https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip It would app..
I have a sip trunk between two asterisk boxes. I can call into the first box, hit 499 for example and the call goes to the second box and answers as expected plays me audio message just fine etc… My issue is that DTMF does not seem to be working.B..
Up till now, I have only used Asterisk versions 1.2, 10 and 11, on CentOS 4, 5 and 6, and have made extensive use of AMI and FastAGI connections to a multi-threaded backend written in C. For a new project, I am looking at trying Asterisk 16 with A..
I have two SIP extensions defined in sip.confregister => 4450@10.20.1.1/4450[4450]type=friend usernameD50host.20.1.1allow=all dtmfmode=inband context=incomingregister => 4451@10.20.1.1/4451[4451]type=friend usernameD51host.20.1.1allow=all dtmfmode=inb..
I have done additional testing and I havent been able to figure out why its failing.Since my original testing we now set the realm on the authentication section to match what we receive from NEC.Its of the format abc@xyz.com I have verified the md5_c..
The Asterisk Development Team would like to announce security releases for Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.These releases are available ..
all,If I use a SIP softphone and set to gsm codec clearly I get a 8K sample… if I change that to something like opus I get a much better sounding input…How do I get a better than 8K sampled input ?I desire to have that input be from a pipe. I h..
How do I get audio from a pipe into asterisk ?I can do on my console aplay -f cd /path/mypipe and I hear the audio. Now I need that audio into asterisk ?Th..
all How could conferance in a 3rd caller without put the second caller on hold i would like to press a feature code mid call and have a 3rd caller enter the call this could be a real person or a automated system to take credit card info mid callthan..
All,I am running pulseaudio on my asterisk server. I setup and extension to connect me with Console/Dsp. I am hearing the audio but its warbly or does not sound right.Any thoughts on what I need to do for that?Another thought is I tried to setup ..