Asterisk   Asterisk SIP


Topic Replies Activity
Device status vs presence status 1 May 31, 2019
Hangup after call end not working on polycom phones 3 May 31, 2019
Query regarding call disconnection after 32s 5 May 30, 2019
Transfer function 3 May 29, 2019
ImportError: No module named astdicts 5 May 28, 2019
RTP re-invite not working 6 May 28, 2019
The user part of the contact header 13 May 28, 2019
Call not sending when host = dynamic rather than gateway IP 16 May 28, 2019
Asterisk BUG for Sip Trunk 10 May 27, 2019
How to multiple call with voip gateway channel 6 May 26, 2019
PJSIP settings for two German Telekom products 17 May 25, 2019
PJSIP TLS Transport 4 May 25, 2019
Enable TLSv1.2 only in SIP/TLS transport 7 May 24, 2019
Asterisk 13 uses closed TCP connection 17 May 24, 2019
Setting up SIP-trunk between asterisk SIP and asterisk PJSIP servers 4 May 24, 2019
PJSIP Provision multiple devices with same extension 6 May 24, 2019
Capture SIP trace per call 5 May 23, 2019
Registration failure over TLS in a LAN (solved) 2 May 22, 2019
Fail2Ban and ChallengeSent Event 3 May 22, 2019
No Video Audio with NAT , ~~once again~~ 6 May 22, 2019
Asterisk call disconnection after a time of 9s 4 May 22, 2019
Chan_sip.c autodestruct on dialog with owner in place 3 May 22, 2019
Random 403 sent from asterisk server 2 May 21, 2019
SIP port for outbound sip calling 3 May 21, 2019
Stop Invite during call 6 May 20, 2019
When wifi restarts, Zoiper Pro cannot call out 2 June 18, 2019
Issue with Audio 12 June 17, 2019
Pjsip SIP/2.0 401 Unauthorized 7 June 16, 2019
Callpick Group NOTIFY 5 June 16, 2019
Opus Bitrate negotiated in SDP headers? 6 June 15, 2019