Asterisk   Asterisk SIP


Topic Replies Activity
About the Asterisk SIP category 2 January 30, 2020
Force Asterisk to renew IP address of peer 1 March 31, 2020
Sip peer replication 3 March 31, 2020
Can't load pjsip 6 March 31, 2020
Asterisk 17/PJSIP/BLF not working: SUBSCRIBE not OKed 4 March 31, 2020
PJSIP create_rtp() UDP6 vs UDP4 4 March 31, 2020
Select codecs to avoid transcoding 2 March 30, 2020
Two IP-Phones registered but calls not working - New to Asterisk 3 March 30, 2020
Pjsip tls registration 4 March 30, 2020
TLS call from internet drop after 30 seconds 2 March 29, 2020
Hitting 171060 error, after setting "Reload Transports", how to handle without restart? 4 March 29, 2020
New MWI stasis subscription on SIP reregister 1 March 27, 2020
How to call to all aors of the extentions 3 March 27, 2020
Pjsip endpoint unavailable but contact available? 9 March 27, 2020
Listen to Inband DTMF even if telephone-event media attribute was present in SDP 2 March 27, 2020
Inband DTMF detection through external library 3 March 27, 2020
How to set up multiple media address in pjsip? 3 March 26, 2020
Asterisk 17.3.0/PJSIP 2.10, dialplan does't work on AWS EC2 istance 7 March 26, 2020
No audio - Fresh install 5 March 26, 2020
Asterisk 17.3 - lost packets 2 March 26, 2020
Res_pjproject does not load 9 March 25, 2020
SIP 200OK over wss not recivied to asterisk 3 March 24, 2020
What is the maximum time value between 200 OK and ACK? 1 March 24, 2020
PJSIP outgoing loadbalancing 3 March 23, 2020
Delay ~10 sec in app_dial.c or dial.c 10 March 23, 2020
Connectivity issue with sip 10 March 20, 2020
IAX Call Distortion 1 March 19, 2020
RTP Stream size change 2 March 19, 2020
Asterisk, SIP phone behind VPN 4 March 18, 2020
302 Redirect with chan_pjsip 2 March 18, 2020