Asterisk   Asterisk SIP


Topic Replies Activity
Different username and authuser in sip registration 10 June 5, 2019
Change From Audio to Video Call using Asterisk server and Grandstream Phones 7 June 5, 2019
Asterisk load test 4 June 5, 2019
Codec Negotiation prefers transcoding over re-negotiation 19 June 4, 2019
Please help! Need help with dealing with multiple physical locations and 911 17 June 4, 2019
Unable to bind contact to AOR 7 June 3, 2019
PJSIP and LDAP authentication 5 June 2, 2019
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 18 June 24, 2019
PJSIP TLS Transport 5 June 24, 2019
Enable TLSv1.2 only in SIP/TLS transport 7 May 24, 2019
Asterisk 13 uses closed TCP connection 18 June 23, 2019
Setting up SIP-trunk between asterisk SIP and asterisk PJSIP servers 5 June 23, 2019
PJSIP Provision multiple devices with same extension 7 June 23, 2019
Capture SIP trace per call 6 June 22, 2019
Registration failure over TLS in a LAN (solved) 3 June 21, 2019
Fail2Ban and ChallengeSent Event 4 June 21, 2019
No Video Audio with NAT , ~~once again~~ 7 June 21, 2019
Asterisk call disconnection after a time of 9s 5 June 21, 2019
Chan_sip.c autodestruct on dialog with owner in place 4 June 21, 2019
Random 403 sent from asterisk server 3 June 20, 2019