Welcome to the Asterisk Community (8)
Building dahdi on fedora22 and "certs/signing_key.pem: No such file or directory" (5)
Audio goes blank for a few seconds during the conversation, and then comes back on (3)
[SOLVED] Centos 7 compatible init.d or systemd script for Asterisk 13 (5)
Match against multiple IP address in PJSIP? ( 2 ) (28)
Web Request Missing Variable Is Null (3)
Microsoft CRM Integration with AsteriskNow (3)
Asterisk : Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) (8)
Need Ssh access to my Digium / Switch box system (3)
Modern 3G IP Radio PTT server (1)
Digium API - is there "hook up" event? (1)
Increase duration of DTMF tones on Digium G200 gateways (2)
Timeout function of function Background isn't working (8)
User Input and auto Dial (3)
Can I hide the clients number from users when using asterisk? (17)
Is there a way to get the sound level from a caller? (2)
Same line input and output (7)
Asterisk SIP Integration with Avaya SES (16)
Monitor picked call (10)
Handle_response_invite: Failed to authenticate on INVITE to '" (13)
Call file playback executes before the person answer the call (7)
Open web page specfic to call (3)
How to Budge into a call with an audio file every 30 seconds (10)
Level 3 Enterprise Voice (1)
Separate SMS email destination based on ${DONGLENAME} in dial plan (4)
V5.2 signalling (7)
TCP with PJSIP - Contact Header in Outbound Invite Message (4)
Shell script disobeying phpagi call flow (5)
OpenBTS and Asterisk as a Proxy (1)
Switchvox and Digium Phone API Questions (2)