Asterisk   Asterisk SIP

About the Asterisk SIP category (1)
Help with response to "OPTIONS" request issues (7)
UDP Socket Leak (3)
PJSIP, events and AORs (1)
Issue : Lose connection to the voip provider (1)
Register SIP extension and SIP trunk to same host IP (8)
Two extensions on the same LAN but no audio (4)
IP address in Call-ID header (5)
[SOLVED] PJSIP - No matching endpoint found (3)
Troubleshooting advice for dropped calls (3)
What does reinvite do? (2)
Peer Matching in Asterisk using chan_sip (4)
Asterisk LetsEncrypt certificate unable to load (3)
Debug several hosts (3)
Understanding endpoints, aors, auth, (3)
Selecting codec by endpoint subnet (1)
Silence in some incoming calls (1)
Autodestruct on dialog (2)
Question about SIP resist upper limit (2)
What does the dial parameter mean in sip.conf? (2)
Why allowguest=yes by default? (3)
How can I create an IP route with channels limited (7)
Call not being hangup (7)
When is contact considered unreachable in PJSIP? (7)
How to send custom data form asterisk to sipml5 (4)
Kernel Optimization for Asterisk . . . (10)
Test Asterisk Context with Sipp ----- Detect A DTMF send from SIPP (14)
Rejected because extension not found in context 'from-test' (10)
How to enter non alphanumeric characters for the password? (3)
Rtpmap: telephone-event (9)