Asterisk   Asterisk SIP


Topic Replies Activity
Bad event error 6 March 18, 2020
Lagged and unreachable 11 March 17, 2020
Don't hear any sound in any of our Asterisks. What should i do to fix that? 3 March 17, 2020
Unable to Display Name of Called Party During Call 4 March 15, 2020
How to show PJSIP User-Agent 8 March 13, 2020
Asterisk not connecting with zoiper 15 March 13, 2020
Port 5060 not opening 3 March 13, 2020
Search capabilities within pjsip history 7 March 13, 2020
Two asterisks, direct media, strictrtp=yes, after media renegotiation (re-invite), RTP dropped 1 March 11, 2020
Pjsip database of registrations 5 March 11, 2020
One way audio - not all services 4 March 10, 2020
Asterisk call timeout 1 second 9 March 10, 2020
SIP over VPN / shadowsocks / proxy 5 March 9, 2020
External asterisk caller puts me on hold 4 March 9, 2020
Pjsip endpoints register timeouts 9 March 9, 2020
Define pjsip endpoints 3 March 5, 2020
CID match not working in Asterisk 16 33 March 5, 2020
Autodestruct on dialog asterisk 11 2 March 4, 2020
All Endpoints unreachable when registering a new one 7 March 4, 2020
PJSIP loads all endpoints from realtime database 4 March 4, 2020
Config asterisk for SIP mobile network 5 March 4, 2020
Dtmf between 2 asterisk servers pjsip 3 March 4, 2020
Tool to check the quality of SIP trunk provided by the end supplier 3 March 3, 2020
If webrtc is yes in ps_endpoints table, webphone works but softphone not 2 March 3, 2020
SSL Connection, Certificate, SIP Client IP 7 March 3, 2020
PJSIP - TLS transport dont load 4 March 2, 2020
Not able to connect call 4 March 2, 2020
When call was transferred at phone[9001], the call is established but no audio 4 March 31, 2020
PJSIP Trunk issues 7 March 29, 2020
How to configure instant messaging in Asterisk SIP? 14 February 28, 2020