Asterisk   Asterisk SIP


About the Asterisk SIP category (1)
Cant register on at&t IP Flex (3)
Error to load "res_pjsip.so" in Asterisk 13.21.1 with PJSIP (7)
Mitel 400 client - 401 Unauthorized - Failed to authenticate (2)
Alert System for a School (2)
Error after restart with 'sip reload' as fix (7)
Namedcallgroup doesn't work with RealTime Sip (1)
Enable TLSv1.2 only in SIP/TLS transport (6)
Change (and update) callee ID during execution of MP3Player()? (2)
Confusing values when graphing Jitter using RTCP (1)
500 Internal Server Error after REFER (1)
Pjsip: How to identify endpoint by transport (3)
Getting SIP read from UDP:192.168.1.1:49079 instead of Public IP address (2)
Asterisk terminating outbound call when picked up, sends 'BYE' message (7)
Call issue from different IP addresses (17)
P-Asserted-Identity and using TIP/TIR (16)
Incoming call from multiple trunk (5)
Asterisk System Connectivity With Cisco Phones (7)
No RTP stream opens automatically (10)
Possible incorrect payload type sent by Asterisk (5)
NAT is driving me crazy! (7)
Asterisk Configuration trunk and extension (6)
Force subsequent requests to be sent to source port of invite ( 2 ) (27)
[SOLVED]Asterisk Realtime PJSIP show endpoints like Unknow (3)
[SOLVED]Asterisk realtime ARA using PJSIP doesn't authenticate (7)
Asterisk after migrate to PJSIP no ring and unavailable (13)
Information related protocol (8)
Asterisk 15.1.2 TLS configuration using (PJSIP stack) (1)
Realtime over ODBC don't work (1)
How to configure TLS and SRTP in asterisk server and client softphone to make secure calling on the Asterisk (3)